Device and method for rerecording an existing audio sample

Abstract

The invention relates to a device for rerecording an existing audio sample having a first number of channels (111, 112), the device comprising: a loudspeaker arrangement for playing back the first number of channels using a first number of loudspeaker systems (201, 202, 203) in a playback chamber (110), wherein the first number is equal to 1 or greater than 1, and wherein the one loudspeaker system (203) or the loudspeaker systems (201, 202, 203) of the first number of loudspeaker systems are designed to generate both a translational sound field and a rotational sound field in the playback chamber (110) for each channel (111, 112, 113) of the first number of channels; a microphone arrangement having a second number of microphone systems (221-231) which are arranged at different microphone positions in the playback chamber (110), wherein each microphone system of the second number of microphone systems (221-231) is designed to detect a pressure signal (OD) and also a directed differential signal (D); and an interface (150) for outputting or recording the pressure signal and the directed differential signal or a differential signal derived from the directed differential signal, as a rerecorded audio sample (240).

Description

[0001] Apparatus and method for re-recording an existing piece of sound

[0002] Description

[0003] The present invention relates to the reproduction and recording of audio signals and in particular to the re-recording of an existing sound piece, which is also referred to as reamping.

[0004] Typically, acoustic scenes are recorded using a set of microphones. Each microphone outputs a microphone signal. For an orchestral audio scene, for example, 25 microphones might be used. A sound engineer then mixes the 25 microphone outputs into, for example, a standard format such as stereo, 5.1, 7.1, 7.2, or another equivalent format. In a stereo format, for example, two stereo channels are created by the sound engineer or an automatic mixing process. In a 5.1 format, the mix results in five channels and one subwoofer channel. Similarly, in a 7.2 format, for example, a mix is ​​created into seven channels and two subwoofer channels. If the audio scene is to be rendered or processed in a playback environment, a mix result is applied to electrodynamic loudspeakers.In a stereo playback scenario, there are two speakers, with the first speaker receiving the first stereo channel and the second speaker receiving the second stereo channel. In a 7.2 playback format, for example, there are seven speakers in predetermined positions and two subwoofers that can be placed relatively freely. The seven channels are routed to the corresponding speakers, and the two subwoofer channels are routed to the corresponding subwoofers.

[0005] The use of a single microphone array for capturing audio signals and a single loudspeaker array for reproducing the audio signals typically neglects the true nature of the sound sources. European patent EP 2692154 B1 describes a set for capturing and reproducing an audio scene that records and reproduces not only translation, but also rotation and, furthermore, vibration. Therefore, a sound scene is not reproduced by a single acquisition signal or a single mixed signal, but by two acquisition signals or two mixed signals, which are simultaneously recorded and simultaneously reproduced. This ensures that different emission characteristics of the audio scene are recorded and reproduced in a playback environment compared to a standard recording.

[0006] For this purpose, as described in the European patent, a set of microphones is placed between the acoustic scene and an (imaginary) listening room in order to capture the “conventional” or translational signal, which is characterized by a high level of directionality or high quality.

[0007] In addition, a second set of microphones is placed above or to the side of the acoustic scene to record a low-quality or low-directivity signal, which is intended to represent the rotation of the sound waves as opposed to the translation.

[0008] On the playback side, corresponding loudspeakers are placed at typical standard positions, each with an omnidirectional arrangement to reproduce the rotational signal and a directional arrangement to reproduce the "conventional" translational sound signal. Furthermore, there is a subwoofer either at each of the standard positions or just a single subwoofer at any one location.

[0009] European patent EP 2692144 B1 discloses a loudspeaker for reproducing, on the one hand, the translational audio signal and, on the other hand, the rotational audio signal. The loudspeaker thus has an omnidirectional emitting arrangement on the one hand and a directional emitting arrangement on the other.

[0010] European patent EP 2692151 B1 discloses an electret microphone that can be used to record the omnidirectional or directional signal.

[0011] European patent EP 3061262 B1 discloses an earphone and a method for manufacturing an earphone that generates both a translational sound field and a rotational sound field.

[0012] The European patent application EP 3061266 AO, which is intended for grant, discloses a headphone and a method for producing a headphone which is designed to generate the “conventional” translational sound signal using a first transducer and to generate the rotational sound field using a second transducer arranged perpendicular to the first transducer.

[0013] The recording and playback of the rotary sound field in addition to the translatory sound field leads to a significantly improved and thus high-quality audio signal perception, which almost conveys the impression of a live concert, even though the audio signal is played back through loudspeakers or headphones or earphones.

[0014] This creates a sound experience that is virtually indistinguishable from the original sound scene, where the sound is emitted not by loudspeakers but by musical instruments or human voices. This is achieved by taking into account that the sound is emitted not only translationally but also rotationally and, if necessary, vibratorically, and therefore needs to be recorded and reproduced accordingly.

[0015] There are various ways to synthetically generate a representation of an existing sound sample, in which only the translational sound field has been recorded, that contains both translational and rotational components. Such methods are disclosed, for example, in WO2022157251 A2, WO2022218822A1, or WO2022253768A1.

[0016] To generate synthetic signals, stereo properties of existing sound samples are used, and the difference between the left and right stereo channels is used to synthetically reconstruct a rotation of the sound field.

[0017] However, this approach is limited to stereo signals and does not allow, for example, to “reconstruct” a rotary sound field for mono signals.

[0018] Furthermore, the synthetic method is limited in that it is no longer suitable for higher-order formats, i.e., formats with more than two channels (left and right) or more than three channels (left, center, right), if high-quality results are to be achieved. Furthermore, it has been shown that even synthetic recovery of rotation information from translationally recorded stereo signals can result in quality issues, for example, from the fact that the difference signal of a stereo signal exhibits strong power fluctuations, which pose difficulties with the synthetically generated rotation signal.Overall, the synthetic generation of "enriched" sound samples—that is, samples that not only contain translational components but also describe the rotational sound field and can ultimately be reproduced using suitable playback systems—represents a good way to enhance existing, purely translational audio content. However, the ultimately achievable qualities for a truly complete sound field with translational sound on the one hand and rotational sound on the other are limited with synthetic generation.

[0019] The object of the present invention is therefore to create an improved concept for producing a piece of clay.

[0020] This object is achieved by a device for re-recording an existing piece of sound according to claim 1, a method for re-recording an existing piece of sound according to claim 33 or a newly recorded piece of sound according to claim 34.

[0021] The present invention is based on the discovery that the reamping process is used to generate a high-quality audio representation. During the reamping process, or when re-recording an existing audio sample with one or more channels, i.e., a first number of channels, the audio sample is played back using loudspeakers arranged at specific loudspeaker positions. If the existing audio sample has only a single mono channel, the loudspeaker for re-recording is arranged in the center relative to a listener's position in a playback room. If the existing audio sample has two channels, i.e., a left channel and a right channel, two different loudspeaker systems are arranged at the two positions on the left and right.Analogously, if the audio sample has three channels—a left channel, a center channel, and a right channel—three loudspeaker systems are arranged at the left, center, and right positions in the playback room. However, the loudspeaker systems are not simple translational loudspeakers, as in conventional reamping, but rather loudspeaker systems that can generate both a translational sound field and a rotational sound field in the playback room. For this purpose, the loudspeaker systems are controlled with two different audio signals, namely an audio signal for the sound transducer system for generating the translational sound field, and another audio signal for the sound transducer system for the rotational sound field. Both audio signals are generated from one and the same channel of the existing audio sample, according to the invention.

[0022] To obtain the newly recorded sound piece, according to the invention, individual microphone systems are arranged at the playback positions desired for a higher-order format. These microphone systems are each designed to detect both an omnidirectional pressure signal and at least one directional difference signal for the position at which the respective microphone systems are arranged. The omnidirectional pressure signal represents the sound pressure and is a typical mono signal or a typical single-channel signal from a multi-channel recording. The directional difference signal also represents a component of the sound velocity at the microphone position.According to the invention, the sound velocity can be detected by various directed difference signals, whereby for high-quality recording it is preferred to record all three components of the sound velocity separately, i.e. both the directed difference signal in the x-direction and the directed difference signal in the y-direction and the directed difference signal in the z-direction.

[0023] The result is output or recorded through an interface in such a way that a newly recorded sound sample is obtained which now has a number of channels equal to the number of microphone systems used, whereby each channel no longer has a single mono signal, as in the existing sound sample, but in addition to an omnidirectional audio signal also also one or more directional difference signals in order to describe the sound velocity at this position either approximately (in the case of only one directional difference signal or a summed difference signal) or particularly precisely in the case of three directional difference signals.

[0024] The method according to the invention and the corresponding device for re-recording are characterized in that any desired formats can be generated for the sound samples, whereby, depending on the embodiment, additional microphone systems can be mounted at corresponding additional positions in the playback room. Particularly preferred is a playback room with specific acoustics, which, in particular if this information is still available, is either the same playback room in which the existing sound sample was recorded, or which has similar acoustic properties or at least comparable acoustic properties to the playback room in which the existing sound sample was recorded.This makes it possible to breathe new life into even old recordings that only exist in mono or stereo, because the loudspeaker systems according to the invention not only generate the translational sound field in the playback room, but also the rotational sound field, which is close to the rotational sound field that the artist also generated when first recording the existing piece of sound in the playback room, but which was not recorded at the time.

[0025] According to the invention, this “omission” in the past can be remedied even if the artist or group of artists is not or no longer available.

[0026] This makes it possible to convert existing recordings that were made without rotary sound field components into a new, particularly high-quality format, whereby the special format can be achieved simply by arranging corresponding microphone systems in corresponding positions required by the special format.

[0027] When re-recording a mono piece, two microphone systems can be used for stereo reproduction, or, for example, five microphone systems for surround reproduction, or even much more sophisticated formats can be achieved, such as a multi-channel arrangement with 22 channels (or any subset thereof with fewer than 22 channels) according to ITU-R BS.2159-9 in an upper layer, a middle layer, and a lower layer, or an arbitrarily selected arrangement of the various 25 loudspeakers according to ISO / IEC 23003:1 . Another format is the Dolby Atmos format with seven loudspeakers in the middle layer and four or six loudspeakers in the upper layer.All of this can be achieved simply by placing microphone systems, not playback speakers, at the corresponding playback positions of these formats. These microphone systems record signals at these locations. When the newly recorded audio piece is to be played back at some point, these signals represent the corresponding signals at the corresponding playback positions and can be reproduced there by suitable speakers. However, this reproduction will not take place with simple translational speaker systems, but rather with speaker systems capable of generating both a translational sound field and a rotational sound field—that is, systems capable of generating not only common-mode signals but also differential-mode signals.

[0028] Preferred embodiments of the present invention are explained in detail below with reference to the accompanying drawings. They show:

[0029] Fig. 1a shows a preferred embodiment of the device for re-recording an existing piece of sound;

[0030] Fig. 1 b is a schematic representation of the existing piece of clay;

[0031] Fig. 1c is a schematic representation of the newly recorded piece of sound;

[0032] Fig. 2 shows an illustration of the control of the individual loudspeakers of the various loudspeaker systems according to an embodiment;

[0033] Fig. 3 shows the loudspeaker positions and the microphone system positions for a Dolby Atmos format with seven channels in the middle layer and six channels in the upper layer;

[0034] Fig. 4 is a schematic overview of different loudspeaker positions and arrangements according to ISO / IEC 23003:1;

[0035] Fig. 5 is a schematic representation of the playback positions of a 22.2 multi-channel arrangement according to ITU-R BS.2159-9;

[0036] Fig. 6a is a schematic representation of the vertical positions in Dolby Atmos;

[0037] Fig. 6b is a plan view of playback positions of a Dolby Atmos 7.4 format;

[0038] Fig. 7 is a schematic representation of a preferred loudspeaker system; Fig. 8 is a further embodiment of a preferred loudspeaker system;

[0039] Fig. 9 shows a preferred embodiment of the sound transducer system for emitting rotational sound;

[0040] Fig. 10 shows another embodiment of a loudspeaker system;

[0041] Fig. 11 a microphone system with two sub-microphones;

[0042] Fig. 12 a microphone system with three sub-microphones;

[0043] Fig. 13a shows a combiner for generating the directed difference signals;

[0044] Fig. 13b shows a single combiner for differential signal generation;

[0045] Fig. 13c shows a combiner according to an embodiment of the microphone system;

[0046] Fig. 14 shows a microphone system or a microphone according to an embodiment;

[0047] Fig. 15 shows a preferred embodiment for generating a control signal for the left side or the right side of Fig. 2; and

[0048] Fig. 16 shows a preferred embodiment for generating a control signal with two variably amplified push-pull signals for the first or left and the second or right push-pull signal feed.

[0049] Fig. 1a shows a device for re-recording an existing sound piece with a first number of channels, wherein a first channel 111 and a second channel 112 are shown by way of example in Fig. 1a. The device comprises a loudspeaker arrangement for reproducing the first number of channels with a first number of loudspeaker systems in a playback room 110. In the embodiment shown in Fig. 1a, the loudspeaker arrangement comprises a first loudspeaker system 201 at a first loudspeaker position, which in the embodiment shown in Fig. 1a is the left position. A second loudspeaker system is shown at 202 and arranged at a second loudspeaker position. Since the existing sound piece has two channels 111, 112, wherein the first channel is intended for the left loudspeaker system 201 and the second channel 112 is intended for the right loudspeaker system 202, only two loudspeaker systems are used.

[0050] Each loudspeaker system is configured to generate both a translational sound field and a rotational sound field for each channel 111, 112 of the first number of channels. For this purpose, each loudspeaker system preferably comprises two different sound transducer systems, namely a sound transducer system for generating the rotational sound field and a separate sound transducer system for generating the translational sound field. For this purpose, a signal processing element 100 is preferably provided to generate two audio signals from each channel. From the first channel 111, the signal processing element 100 generates the first audio signal 111a for the translational sound transducer system and the second audio signal 111b for the rotational sound transducer system.Accordingly, the signal processing element 100 also generates a first audio signal 112a and a second audio signal 112b from the second channel 112, wherein these audio signals are fed to the respective different sound transducer systems of the second loudspeaker system.

[0051] Although in preferred embodiments for the loudspeaker systems, which will be explained in detail with reference to Figures 7 to 10, completely separate sound transducer systems are used to generate the rotational sound field on the one hand and the translational sound field on the other hand, sound transducer systems with at least two loudspeakers can alternatively be used, which are both controlled with the common-mode signal and the differential-mode signal, so that these two loudspeakers together generate both the translational sound field and the rotational sound field, as is shown, for example, in the international application WO2022218824A2, which is hereby incorporated by reference in its entirety.

[0052] The use of separate systems, however, allows the rotating sound field to be radiated almost omnidirectionally, allowing particularly good omnidirectional excitation of the rotating sound field in the playback room, even if the audio sample has only a single mono channel or only a stereo signal with two stereo channels in a 3-channel format. The first number of channels can therefore be one if the existing audio sample has only a single mono channel, or if only a single channel is to be used for reamping. Alternatively, the existing audio sample can also have two or more channels, so the first number of channels can also be greater than 1.

[0053] The re-recording device further comprises a microphone arrangement with a second number of microphone systems 221, 222, 223, 224, 225 arranged at different positions in the playback space. In particular, each microphone system of the second number of microphone systems is configured to detect both an omnidirectional pressure signal and a directional difference signal. The omnidirectional signal is denoted by OD in Fig. 1a, while the directional difference signal is denoted by D. The subsequent number refers to the number of the microphone system by which the signals are detected.

[0054] Furthermore, the loudspeaker arrangement comprises an interface 150 for outputting or recording the omnidirectional pressure signal and at least one directional difference signal for each of the microphone systems of the second number of microphone systems to obtain the newly recorded sound sample 240 either in recorded form or in output form. In the embodiment shown in Fig. 1a, the sound sample after reamping comprises nine or ten different audio signals generated from an existing sound sample with two channels. In the embodiment shown in Fig. 1a, the second microphone system 221 is shown in the center position.Depending on the implementation, it has been shown that this playback position is well described solely by the omnidirectional pressure signal, because typically, for example, a speaker or singer is positioned at this location, which is essentially characterized by direct sound generation and output. Therefore, at this position, the directional difference signal may be secondary and can be omitted, as indicated in Fig. 1a with the bracket around the directional difference signal D2 at the input of the interface 150 or at the output of the second microphone 221.

[0055] Alternatively, the omnidirectional signal can also be generated at this point, for example, by adding the first and second channels, and can thus be inserted separately, as shown at 151. If neither the omnidirectional signal OD2 nor the directional difference signal D2 from the microphone system 221 is required at the center position, the installation of a microphone system at this point can be completely omitted in order to reduce the effort required for reamping or re-recording. Then, the center channel from the existing audio sample, if present, is added to the newly recorded audio sample, or, if not present, from the existing left and right channels by downmixing, and the result of the downmixing is added to the newly recorded audio sample.

[0056] As will be shown with reference to Fig. 3, the use of the second microphone system 221 at the center position is also preferably omitted if a three-channel signal having a center channel is to be re-recorded. Then, in the newly recorded sound sample, channel or track B is generated from the original center channel 113, as shown in Fig. 2, and inserted into the newly recorded sound sample via the insertion element 151 and the interface 150, without actually recording with a microphone system having taken place. Of course, however, depending on the embodiment, a microphone system 221 can also be set up at the center position, as shown in Fig. 1a, and both the omnidirectional signal and the directional difference signal from this microphone system can be introduced into the newly recorded sound sample via the interface 150.

[0057] In principle, playback room 110 can be any playback room, which doesn't necessarily have to be a closed room, but simply has a defined spatial extent in which the speaker systems and microphone systems can be set up at specific positions. Thus, the room can also be an "open space" if, for example, the sound piece was a live concert, i.e., not recorded in a specific studio.

[0058] However, if the piece of sound has been recorded in a studio, it is preferable to use this same studio as the playback room for the re-recording, since the rotating sound field that was generated but not recorded by the artist during the very first recording of the piece of sound is then, in a sense, subsequently recorded by the re-recording procedure according to the invention and made available for high-quality audio playback.

[0059] Fig. 1b shows a representation of an existing audio sample, which will typically have certain metadata, and which, in addition to the metadata, will have a left channel or track A, a right track or channel C, and a middle or mono channel B. The left channel is labeled 111 or 1001. The right

[0060] Channel is labeled 112 or 1002 and the middle channel is labeled 1013.

[0061] For comparison, Fig. 1c shows the newly recorded sound sample, which again has metadata that can now be more extensive than the original metadata because the data assigned to the metadata has now become considerably richer. In the exemplary scenario in Fig. 1c, in which five playback positions are shown: left wide, right wide, left, right, and left rear, the original three channels have already become 20 audio tracks, with one “channel,” e.g., the LW channel, consisting of three individual channel signals, namely the omnidirectional pressure signal OD1, the directional difference signal D1a in the x-direction, the directional difference signal D1b in the y-direction, and the directional difference signal D1c in the z-direction. The generation or recording of these difference signals is described in detail below with reference to the more detailed explanation of the microphone system in Figs. 11 to 14.

[0062] Depending on the design, a maximum, i.e., maximally accurate, representation of the sound velocity can be explicitly recorded using all three components x, y, and z. However, certain "intermediate" representations can also be used to reduce the transmission effort or the size of the newly recorded sound sample. For example, only a single difference signal can be recorded and transmitted, for example, the largest of the three difference signals. Alternatively, the three difference signals can be added together to generate and transmit a total difference signal, for example, by adding Di a , D and Tue cAlternatively, two difference signals, i.e. x, y or x, z or y, z, can be added together. Any combinations are also possible, for example only the transmission of the largest difference signal in terms of its power, e.g. in a frame or in a certain time range, and optionally additionally the second largest signal or a sum of the two remaining directional components, etc. In any case, the newly recorded sound sample will preferably have more “channels” than the original sound sample, and each channel will, in addition to the preferably omnidirectional pressure signal, also comprise at least one directional difference signal, which is an actually measured difference signal or a signal resulting from the combination, such as addition, of measured difference signals.In any case, this difference signal will describe the sound velocity more or less accurately and will allow a corresponding differential mode transducer, i.e. a corresponding transducer for generating a rotating sound field, to be controlled when the newly recorded sound sample is to be played back. The pressure signal is, as described, preferably omnidirectional, but depending on the microphone, it can also contain remnants of a directivity. However, it is a pressure signal and, in contrast to the directed difference signals that represent the sound velocity, does not represent the direction of a vector field, but merely the undirected pressure at the microphone position. It can be obtained in various ways, e.g. by adding signals from microphone diaphragms, by using a dedicated pressure transducer, or, as in the preferred case, by adding the two components of the diaphragms of a sub-microphone, such asof the sub-microphone for the z-direction.

[0063] The inventive concept is preferably useful for "upscaling," meaning that the second number of microphone systems is greater than the first number of channels. In principle, however, it is also possible, for example, to use only one channel of an existing audio sample and to use this channel to drive an inventive loudspeaker system in order to then obtain the newly recorded mono signal, which now includes a directional difference signal in addition to the omnidirectional pressure signal, using an inventive microphone arranged at the center position next to the loudspeaker system or between listener position 114 and the loudspeaker system. The second number can therefore also be equal to the first number.

[0064] Other formats include creating a newly recorded sample with stereo characteristics from an original sample with mono characteristics, or from an original stereo sample that lacked rotation information. Furthermore, it is also possible to create an enriched, i.e. newly recorded sample from three original channels using three microphone systems. This sample again has three "channels," but each channel now contains one or more directional difference signals in addition to the (omnidirectional) pressure signal.

[0065] Fig. 3 shows a specific embodiment of the arrangement of the loudspeaker systems and microphone systems, using the example of a re-recording for the Dolby Atmos format with the inventive loudspeaker systems from Molecular Acoustics (MA) and with the microphone systems from Molecular Acoustics (MA). In particular, a format is shown that has six playback positions and thus positions for the inventive microphone systems, namely positions 222, 223, 224, 225, 226, and 227 in the middle layer. In addition, six further playback positions and thus microphone system positions 226, 227, 228, 229, 230, and 231 are shown in the upper layer. The resulting newly recorded signal then has the enriched “channels” for the microphone positions 222 to 231 and additionally the center signal, which, however, is directly extracted from the existing sample and added to the signal as shown and illustrated at 151 in Fig. 1a.Furthermore, if in the embodiment shown in Fig. 3 the omnidirectional signal and three directional difference signals are recorded for each microphone position, 12 “enriched” channels and one center channel are created from the original three-channel signal with the loudspeaker positions 201, 202, 203, wherein each enriched channel has an omnidirectional signal and three directional difference signals, so that the newly recorded sound sample for the implementation of Fig. 3 has a total of 48 + 1, i.e. 49 channels as a third number of channels, if the first number of channels is equal to three and the second number of microphone systems is equal to 12.

[0066] The English expressions in Fig. 3 correspond to the following German terms: left = links, right = rechts, front = vorne, mid = Mitte, rear = hinten, top = oben, wide = breit, side = Seite, MA triple microphone units = Molecular Acoustics triple microphone units, loudspeaker = Lautsprecher.

[0067] Fig. 4 shows channel abbreviations and loudspeaker positions in the top view as shown in ISO / IEC 23003:1.

[0068] The English terms correspond to the following German terms: center = middle, low frequency enhancement = low frequency enhancement, surround = surroundings, direct = direct, side = side, vertical = vertical, height = height.

[0069] Another playback format, the 22.2 multi-channel format, is shown in Fig. 5, as described in ITU / R BS.2159-9. Here, there are three layers: the bottom layer 501, the middle layer 502, and the top layer 503. The bottom layer has three channels, the middle layer has ten channels, and the top layer has nine channels, with two additional subwoofer or LFE channels. The individual abbreviations for the speaker positions are shown in Fig. 5 in the lower view and in the upper view with respect to a TV screen 504.

[0070] Fig. 6a and Fig. 6b show the vertical positions and a top view of the playback positions for Dolby Atmos, respectively. The format shown in Fig. 6b differs from the format shown in Fig. 3 in that the two middle microphone positions or speaker positions 228, 229 TML (top center left) and TMR (top center right) are still used in Fig. 3, whereas these are not present in Fig. 6b.

[0071] According to the invention, microphone systems can be positioned at all playback positions as shown in Fig. 3, Fig. 4, Fig. 5, Fig. 6a, 6b. Any subsets of microphone systems can also be placed, as can be seen from the comparison of Fig. 6b and Fig. 3, whereby in Fig. 6b the two positions TML and TMR are not occupied by microphone systems, so that a smaller format is created. Alternatively, user-specific formats that do not correspond to the standardized formats can also be used, although it is preferred to map corresponding formats or parts of correspondingly standardized formats with microphone systems and then add the corresponding designations to the metadata of the newly recorded sound sample.

[0072] Fig. 2 shows a preferred embodiment of the control of the three loudspeaker systems 201, 202, 203. Each individual loudspeaker system, which is arranged at a specific loudspeaker position, such as the left position, the middle position and the right position, comprises a first sound transducer system 201a, 203a, 202a for reproducing the conventional (translational) sound signal by means of a sound transducer system referred to as a CM or common mode transducer. The sound transducer system can comprise a single loudspeaker or multiple loudspeakers, i.e. multiple loudspeaker diaphragms, which are typically housed and can be supported by a subwoofer, as will be explained with reference to Figs. 7 to 10. In addition, each loudspeaker system comprises a further sound transducer system 201b, 203b, 202b, which is controlled with a separate audio signal and is designed to generate a rotating sound field.This transducer system is designated DM (Differential Mode) in Fig. 2 and is symbolically indicated by two speakers directed away from each other, radiating in opposite directions. However, a sound transducer system is preferably used that is nearly omnidirectional, i.e., radiates essentially the same in all directions, and is preferably designed as a dodecahedron, as shown in Fig. 9.

[0073] Each audio signal is amplified by a respective amplifier 201c, 201d, 203c, 203d, 202c, 202d. Typically, the loudspeaker systems are active loudspeaker systems, so the amplifiers are housed within the loudspeaker systems, as shown in Fig. 2. Alternatively, however, the amplifiers can also be arranged separately from the loudspeakers, for example, in the signal processing element 100, which, however, is shown in Fig. 2 as not including the power amplifiers.

[0074] To generate the two audio signals 113a, 113b, preferably only attenuators 113c and 113d are used. The channel signal of the existing audio sample for the center, i.e., track B or mono channel 113, is fed to a branching point 113e and from there, via a first attenuator 113c, to the sound transducer system for generating the rotary sound 203b, while the second audio signal 113a is generated by attenuating the original signal 113 by the second attenuator 113d. Preferably, the attenuators 113c, 113d are set differently, such that the attenuator 113d attenuates less than the attenuator 113c. By adjusting these adjustable attenuators, it is possible to adjust the proportion of the rotary sound for the center channel.Even if only a single mono signal is present, the radiation of the audio signal 113b, although this signal is in phase with the audio signal 113a, results in a rotational sound field being excited in the playback space due to the second sound transducer system 203b. This effect, also referred to as the "generator effect," results in a rotational sound field being excited in the playback space, even if the originally existing sound sample does not even have a stereo signal, and therefore no synthetic generation of a differential mode signal is possible, which requires the presence of stereo signals. The rotational sound field generated by the transducer 203b results in a rotational sound field being propagated or excited in the playback space, which is recorded at the microphone positions in the x, y, and z directions, if necessary, for a complete representation of the sound velocity.Thus, a significantly improved reproduction can be achieved simply by using a preferably closed reproduction chamber in the novel reamping of a mono signal according to the invention. The generation of the control signals 111a and 112a for the sound transducer system for reproducing the translational sound field takes place for both the left and right channels by feeding the left stereo channel and the right stereo channel, respectively, directly or after attenuation by a preferably adjustable attenuator 1105 and 1205, respectively, to the corresponding power amplifiers 201d, 202d.

[0075] To generate the difference signal, i.e., the control signal 111b, 112b, the input-side left stereo channel 111 or 1001 and the right stereo channel 112 or 1002 are now used. Two polarity reversals are performed in blocks 1015, 1016, and the results of the polarity reversals are then added in the respective blocks 1013, 1014 to obtain the push-pull signals 1011 or 1012, which, depending on the embodiment, are then fed directly or after amplification by a voltage-controlled amplifier (VGA) 1031, 1032 to the second sound transducer system for generating the rotating sound field. On the way to the loudspeaker, attenuation can also be performed in an attenuator 1101 or 1201. In addition, depending on the implementation, the difference signal at the output of the amplifier 1031 or 1032 or at the output of the adder 1013, 1014 can be supplemented by an attenuator 1103 or1203 attenuated common mode signal can be added, wherein the signal at the output of the adder 1102 or 1202, which carries out this addition, can be further attenuated by an optional attenuator 1104, 1204 in order to finally obtain the respective second audio signal for the DM converter system, which is marked with 111 b or 112 b.

[0076] It should be noted that all elements 1101, 1102, 1103, 1104, 1105 and 1201, 1202, 1203, 1204, 1205 are optional elements that, depending on the implementation, can be fixed to a specific damping value, which can vary from element to element. Alternatively, some or all elements may be absent or set to a damping value of 0. Only subgroups of these elements may be present.

[0077] Furthermore, the use of amplifier 1301 or 1302, which is controlled by control signal 1035 via branching point 1033, is optional, as is also represented by the likewise optional switch, which then feeds either signal 1011 or signal 1036 or signal 1012 or signal 1037, after optional processing, to the second sound transducer system or DM converter system 201b, 202b. It is merely preferred that the second audio signal, i.e., the control signal for the sound transducer system for generating the rotating sound field, be derived from the differences between the two stereo channels present in the originally existing sound sample. Furthermore, the use of a polarity inverter 1015, 1016 and subsequent addition at 1013, 1014 is also merely optional.A subtraction can also take place, or an approximate subtraction in the sense that the signal at the output of the operation, i.e. the signal labelled ‘AC’ or ‘CA’, is due to the difference between the left stereo channel and the right stereo channel, without this difference being the exact result of an exact subtraction.

[0078] The optional use of the voltage-controlled amplifier 1031 or 1032, or generally the control of such an amplifier based on a correlation of the left and right channels, is discussed with reference to Figs. 15 and 16. This procedure is also described in WO2022253768 A1, which is hereby incorporated by reference in its entirety.

[0079] Fig. 15 shows a device for generating a control signal for a sound generator, comprising a push-pull signal generator 1010, a controllable amplifier 1030, and a controller 1020. The push-pull signal generator 1010 is configured to generate a push-pull signal 1011 from a first channel signal and a second channel signal. The first channel signal 1001 and the second channel signal 1002 originate from a multi-channel audio signal and can, for example, be the left channel signal and the right channel signal.

[0080] The controllable amplifier 1030 is configured to amplify or attenuate the push-pull signal 1011 with an adjustable amplification or attenuation according to a setting value 1035 that the controllable amplifier 1030 receives from the controller 1020. In particular, the device in Fig. 15 is configured to use the amplified push-pull signal 1036 as the basis for the control signal for one or more sound generators.

[0081] The controller 1020 is configured to determine the setting value 1035 such that, given a first similarity between the first channel signal and the second channel signal, a first setting value is determined, and such that, given a second similarity between the first channel signal and the second channel signal, a second setting value is determined. In particular, the first similarity represents a lower similarity than the second similarity, and the first setting value represents a smaller gain than the second setting value or a greater attenuation than the second setting value. This relationship is schematically illustrated in the mapping function 1000, which represents a setting value for a gain (setting value greater than 1) and / or for an attenuation (setting value less than 1), depending on a similarity scale.In particular, the gain becomes increasingly larger for larger similarity values, i.e., for stronger similarities between the first channel signal and the second channel signal. This is advantageous in that it balances out or partially compensates for the level loss of the push-pull signal, which is preferably generated as a difference signal or an approximate difference signal. On the other hand, the gain becomes increasingly smaller the more dissimilar the two channel signals are, because the level of the push-pull signal then continues to increase. A special situation arises when the first channel signal and the second channel signal are particularly dissimilar, i.e., fully correlated but in antiphase. In this case, the calculation of the push-pull signal leads to an exaggeration of the level of the push-pull signal, which, according to the mapping function, maps similarity values ​​to setting values, as shown schematically at 1000 in Fig.15, is approached according to the invention in such a way that the push-pull signal is then amplified less or even attenuated, i.e. with an amplification factor of less than 1 in a linear scale or with a negative amplification factor in a logarithmic scale, such as a dB scale.

[0082] Amplification can be an increase in level, i.e., an amplification with a gain factor greater than 1, or a positive gain factor on a dB scale. However, amplification can also be an amplification with a gain factor less than 1, i.e., an attenuation. In this case, the gain factor lies between 0.1, or in the negative range on a dB scale.

[0083] Depending on the embodiment, a direct analysis of the signals to determine the setting value takes place in the device of Fig. 15. Alternatively, the multi-channel audio signal, which comprises the first channel signal 1001, 111 and the second channel signal 1002, 112, comprises metadata 1050. The controller 1020 is configured to extract the setting value 1035, 1051 from the metadata 1050. The controllable amplifier is configured to apply the adjustable amplification or attenuation to the push-pull signal 1011 according to the extracted setting value. This is represented by the arrow pointing into block 1020 for the metadata at 1051. In this case, a direct signal analysis does not necessarily take place in the device of Fig. 15. In a mixed implementation, a starting value for the setting value is read from the metadata 1051, which can then be refined by a device designed for actual signal analysis.In contrast, a device that cannot perform signal analysis but can only read the metadata 1051 will use the same starting value for an entire piece, which already represents an improvement, or at certain times within a piece, at which a new setting value is again available in the metadata, will use this new setting value to adjust the controllable amplifier(s).

[0084] Preferably, the controller 1020 is designed to determine a correlation value between the first channel signal 1001 and the second channel signal 1002, wherein the correlation value is a measure of the similarity. Particularly preferably, the controller 1020 is designed to calculate a normalized cross-correlation function from the first channel signal and the second channel signal, wherein a value of the normalized cross-correlation function is a measure of the similarity. In particular, the controller 1020 is designed to calculate a correlation value using a correlation function that has a value range of negative and positive values, wherein the controller is designed to determine a setting value that represents an attenuation or amplification for a negative value of the correlation function, and to determine the setting value that represents an amplification or attenuation for a positive value of the correlation function, i.e. the other one.A typical normalized cross-correlation function has a range of values ​​between -1 and +1 , where the value -1 means that the two signals are fully correlated but in antiphase, and thus maximally dissimilar.

[0085] On the other hand, a value of +1 is obtained when the two channel signals are fully correlated and in phase, i.e., maximally similar. The push-pull signal becomes increasingly larger as the value of a normalized cross-correlation function decreases from -1 to 0, which is why the gain factor is continually reduced in this range. In contrast, with a value of the normalized cross-correlation function between 0 and -1, the similarity becomes increasingly smaller, which is why the push-pull signal is increasingly attenuated or amplified less and less to counteract the exaggeration of the push-pull signal. A similarity between the channel signals is therefore only concurrent with the cross-correlation function when the two channel signals are in phase, i.e., when the sign of the cross-correlation function is +1. In contrast, the similarity is opposite to the value of the cross-correlation function when the sign of the cross-correlation function is negative.

[0086] A preferred mapping function, which could be used, for example, as a mapping function in 1000 of Fig. 1, has the gain av in dB as the y-axis. The similarity value along the x-axis has a value range between -1 and +1, whereby maximum similarity would be obtained at a cross-correlation value of +1. On the other hand, maximum dissimilarity, which corresponds to a value of the cross-correlation function with respect to the absolute value of 1 and a negative sign, will result in the gain becoming an attenuation, i.e. a gain less than 1 or a gain in the negative dB range. In such an embodiment, the relationship between the similarity value on the one hand and the setting value on the other hand is linear, up to values ​​of approximately 0.8 in one embodiment. Above values ​​greater than, for example,At 0.8, the gain curve to the set value is shown in dotted lines and, in certain embodiments, will no longer be linear. This is because the push-pull signal becomes increasingly smaller and, in extreme cases, when 100% in-phase signals are present at the controller input, can even reach 0. In such a case, a very high gain would result in only a very small signal being amplified, which would then consist largely of noise. Therefore, in this case, the gain is either left at a maximum level or reduced to 0 to essentially "switch off" the push-pull signal.Yet another possibility is to reduce the gain factor beforehand, as shown by the curved dashed line, which then flows into the vertical dotted line, in order to already go into a certain compression at values ​​of perhaps 0.6 or 0.7, and then go to a gain of 0 dB at a value of 0.8 or at values ​​greater than 0.8 but less than 1 .

[0087] In an alternative embodiment of the present invention, the setting value is not determined based on the channel signals 1001, 1002, but rather based on the push-pull signal 1011, as shown by the dotted lines from the push-pull signal 1011 to the controller 1020 in Fig. 15. In this case, a level, an amplitude or another amplitude-related quantity, such as the magnitude of the amplitude, the square of the amplitude or even the cube of the amplitude as an approximation for the loudness, etc. of the push-pull signal 1011, is detected. Depending on this level, the gain is adjusted in order to select a large gain when the level or amplitude of the push-pull signal is low and to use an attenuation or amplification with a gain value less than 1 when the level of the push-pull signal is high. Depending on the implementation, the gain similarity mapping function may be stored in a look-up table in the controller 1020 of Fig.1, or can be calculated using a function with a quantitative similarity value as the input and the setting value 1035 as the output. Alternative approaches other than a table or a function can also be used.

[0088] Fig. 15 shows a general implementation that can be used when only a single control signal for a single sound transducer is to be calculated. Fig. 15 also shows a basic implementation in which additional push-pull signals for additional loudspeakers can also be calculated. Such a special implementation for calculating two push-pull signals, for example a push-pull signal for a left side and a push-pull signal for the right side, is shown in Fig. 16. In Fig. 16, with a common-mode signal feed, a left channel signal is fed in as an example for the first channel signal 1001, which also has the reference numeral 1011 in subsequent illustrations. Furthermore, a right signal 1002 or 1012 can be fed in in a further common-mode signal feed.

[0089] The push-pull signal generator 1010 is shown in dashed lines and comprises an adder 1013, a further adder 1014, and two polarity reversal stages 1015 and 1016, respectively. This ensures that a first push-pull signal 1011 is calculated as a difference signal from the left channel signal and the right channel signal, and that a further push-pull signal 1012 is generated from the difference between the right channel and the left channel. Both of these signals are input to the controllable amplifier 1030, which comprises a first individual amplifier 1031 for the left or first push-pull signal, and a second individual amplifier 1032 for the second or right push-pull signal 1012. The amplifier 1030 has an input for the setting value g(t), which here can be a voltage value derived from a signed value c(t) of the normalized cross-correlation function, which provides a value range between -1 and +1.The amplifiers 1031, 1032 receive the same setting value via the branching point 1033 and are preferably designed as voltage-controlled amplifiers. They provide a gain between -10 dB and +10 dB for the given similarity value range between -1 and +1, which is preferably converted into driver voltages, i.e., values ​​with the dimension volts (V). In the embodiment shown in Fig. 16, the controller 1020 internally calculates the value c(t) of the preferably used normalized cross-correlation function and converts this value via a mapping 1000 into the corresponding gain value g(t), which is fed to the amplifier 1030 via the terminal 1035. This value is preferably provided at a branching point 1033 and passed on in exactly the same way to the two individual amplifiers 1031, 1032.However, alternative different gain values ​​can also be used for the different signals, although it is preferred to use the same setting value for both differences.

[0090] Depending on the implementation, individual amplifiers 1031, 1032 can be configured to receive a specific voltage value, a specific current value, or a specific digital value as a setting value. In such a case, the controller 1020 is configured to convert a corresponding similarity value using the table 1000 into the voltage value, current value, or digital value required by the amplifiers 1031, 1032. In alternative embodiments, the controllable amplifier 1030 can also be configured to already include a conversion using a table 1000. Then, with regard to the terminology of the present invention, this conversion is to be considered part of the controller. It is therefore pointed out that the controller 1020 and the amplifier 1030 do not necessarily have to be separate physical elements or semiconductor components or separate entities, but that the definitions of these elements are functional definitions.The controllable amplifier 1030 finally provides two push-pull signals, ie a first signal 1036, 72 for left and a second signal 1037 for right.

[0091] A preferred embodiment of a loudspeaker system is discussed below with reference to Figs. 7 to 10. This technology is also described in WO2012130986, which is hereby incorporated by reference in its entirety.

[0092] A preferred loudspeaker is described below with reference to Fig. 7 and Fig. 8. The loudspeaker comprises an elongated housing 300 having at least one subwoofer speaker 310 for emitting lower sound frequencies. Furthermore, a support portion 312 is provided on an upper end 310a of the longitudinal housing. Furthermore, the longitudinal housing has a lower end 310b, and the longitudinal housing is preferably closed in its entirety and is closed in particular by a bottom plate 310b and the upper plate 310a in which the support portion 312 is provided.Furthermore, an omnidirectionally emitting loudspeaker assembly 314 is provided, which has individual loudspeakers for emitting higher sound frequencies, arranged in different directions with respect to this longitudinal housing 300, wherein the loudspeaker assembly is fixed to the support section 312 as shown and is not surrounded by the longitudinal housing 300. Preferably, the longitudinal housing is a cylindrical housing with a circle as a diameter over the entire length of the cylindrical housing 300. Preferably, the longitudinal housing has a length greater than 50 cm or 100 cm and a lateral dimension greater than 20 cm. As shown in Fig.As illustrated in Figure 8, a preferred dimension of the longitudinal enclosure is 175 cm, the diameter is 30 cm, and the dimension of the beam in the direction of the longitudinal enclosure is 15 cm, and the speaker assembly 314 is formed in a wall shape and has a diameter of 30 cm, which is the same as the diameter of the longitudinal enclosure. The beam portion 312 preferably has a base portion that has matching dimensions with the longitudinal enclosure 300. Thus, if the longitudinal enclosure is a round cylinder, the base portion of the beam is a circle that matches the diameter of the longitudinal enclosure. However, if the longitudinal enclosure is square, then the lower portion of the beam 312 is also square and matches the dimensions of the longitudinal enclosure 300.

[0093] Furthermore, the support 312 has a tip portion with a cross-sectional area that is less than 20% of the cross-sectional area of ​​the base portion, with the speaker assembly 314 being fixed to the tip portion. As illustrated in Fig. 8, the support 312 is preferably conical, so that the entire speaker illustrated in Fig. 8 looks like a pencil with a ball at the tip. This is advantageous due to the fact that the connection between the omnidirectional speaker assembly 314 and the subwoofer-provided enclosure is as small as possible, since only the tip portion 312b of the support is in contact with the speaker assembly 314. This results in good sound decoupling between the speaker assembly and the longitudinal enclosure.Furthermore, it is preferred that the longitudinal enclosure be placed below the loudspeaker array, since the omnidirectional emission is even better when it comes from above rather than from below the longitudinal enclosure.

[0094] The loudspeaker arrangement 314 has a sphere-like support structure 316, which is also shown in Fig. 9 as a further example. The individual loudspeakers are mounted so that each individual loudspeaker emits in a different direction. To illustrate the support structure 316, Fig. 8 shows several levels, each level pointing in a different direction and each level representing an individual loudspeaker with a diaphragm, for example a simple piston-like loudspeaker, but without a rear wall enclosure for this loudspeaker. The support structure can be realized in particular as shown in Fig. 9, wherein the loudspeaker spaces or levels 318 are again shown. Furthermore, it is preferred that the structure as shown in Fig.9, further comprises many holes 320, so that the support structure 360 ​​merely fulfills its function as a support structure, but does not influence the sound emission and, in particular, does not prevent the membranes of the individual loudspeakers from being freely suspended in the loudspeaker arrangement 314. Since freely suspended membranes generate a good rotational component, a useful and high-quality processing of rotational sound can be generated. The support structure is therefore preferably as unobtrusive as possible, so that it merely fulfills its function of structurally supporting the individual piston-like loudspeakers without influencing the possibility of deflections of the individual membranes.

[0095] Preferably, the loudspeaker arrangement comprises at least six individual loudspeakers, and in particular even twelve individual loudspeakers arranged in twelve different directions. In this example, the loudspeaker arrangement 314 comprises a pentagonal dodecahedron (e.g., a body with 12 evenly distributed surfaces) with twelve individual faces, each individual face being provided with an individual loudspeaker diaphragm. Importantly, the loudspeaker arrangement 314 does not comprise a loudspeaker cabinet, and the individual loudspeakers are held by the supporting structure 316, so that the diaphragms of the individual speakers are freely suspended.

[0096] Furthermore, as illustrated in Fig. 10 in another example, the longitudinal enclosure 300 not only includes the subwoofer, but also includes electronic components required to power the subwoofer speaker and the speakers of the speaker assembly 314. Furthermore, the longitudinal enclosure 300 does not only include a single subwoofer. Instead, one or more subwoofer speakers may be provided in the front of the enclosure, where the enclosure has openings indicated at 310 in Fig. 10, which may be covered by any type of covering material such as foam-like film or the like. The entire volume of the closed enclosure serves as a resonator for the subwoofer speakers. The enclosure also includes one or more directional speakers for mid and / or high frequencies, indicated in Fig.10 with 602, which are preferably aligned with the one or more subwoofers indicated in Fig. 10 with 310. These directional loudspeakers are arranged in the longitudinal housing 300, and if more than one such loudspeaker is present, then these loudspeakers are preferably arranged in a row, as illustrated in Fig. 10, and the entire loudspeaker is arranged with respect to the listener such that the loudspeakers 602 face the listeners. Subsequently, the individual loudspeakers in the loudspeaker arrangement 314 are supplied with the second detection signal or the second audio signal, and the directional loudspeakers are supplied with the respective first audio signal. Thus, each individual loudspeaker system has an omnidirectional arrangement 316, a directional arrangement 602, and a subwoofer 310. Preferably, the three loudspeakers 602 are arranged in a d'Appolito arrangement, i.e., the upper and lower speakers are midrange speakers and the speaker in the middle is a tweeter.

[0097] Alternatively, however, the loudspeaker in Fig. 10 may be used without the directional loudspeaker 602 to implement the omnidirectional arrangement in Fig. 1b for each loudspeaker location, and an additional directional loudspeaker may be placed, for example, only near the center position or near each loudspeaker position to reproduce the high directivity sound separately from the low directivity sound.

[0098] In addition, the housing includes another speaker 604 suspended from an upper portion of the housing and having a freely suspended diaphragm. This speaker is a low / mid-frequency speaker for a low / mid frequency range between 80 and 300 Hz, and preferably between 100 and 300 Hz. This additional speaker is advantageous because, due to the freely suspended diaphragm, the speaker generates rotational stimulation energy in the low / mid frequency range. This rotation amplifies the rotation generated at low / mid frequencies by speakers 314. Speaker 604 receives the low / mid frequency portion of the signal provided to the speaker at 314, e.g., the second detection signal or the second mixed signal.In a preferred embodiment with a single subwoofer, the subwoofer is a twelve-inch subwoofer in the closed longitudinal enclosure 300, and the loudspeaker assembly 314 is a pentagonal dodecahedral mid-high frequency loudspeaker assembly with freely vibrating midrange diaphragms.

[0099] With reference to Figs. 11 to 14, a preferred embodiment of a microphone arrangement is presented below. If only a single differential signal is to be captured, it can be configured with a single sub-microphone, but is preferably configured with two or even three sub-microphones to capture the entire sound waveform in the x, y, and z directions. This technique is also described in WO2022157252A1, which is hereby incorporated by reference in its entirety.

[0100] Fig. 11 shows a first sub-microphone 1 with a diaphragm pair comprising a first diaphragm 11 and a second diaphragm 12 arranged opposite one another. In addition, Fig. 11 shows a second sub-microphone 2 with a second diaphragm pair comprising a third diaphragm 13 and a fourth diaphragm arranged opposite one another. The first diaphragm pair is arranged such that the first diaphragm 11 and the second diaphragm can be deflected along a first spatial axis, such as the x-axis, for example, and the second diaphragm pair is arranged such that the third diaphragm 13 and the fourth diaphragm 14 can be deflected along a second spatial axis, such as the y-axis of Fig. 11. The second spatial axis differs from the first spatial axis, i.e. the two spatial axes are not parallel. Preferably, the two spatial axes x, y are orthogonal to one another or have an angle lying between 60 and 120°.

[0101] Fig. 12 further shows a third sub-microphone 13 with a third diaphragm pair, which has a fifth diaphragm 15 and a sixth diaphragm 16, which are arranged opposite one another, wherein the third diaphragm pair is arranged such that the fifth diaphragm 15 and the sixth diaphragm 16 can be deflected along a third spatial axis, such as the z-axis. The third spatial axis differs from the first spatial axis and the second spatial axis, wherein preferably all three spatial axes are orthogonal to one another. Different angles between the third spatial axis and the first or the second spatial axis, such as in a range between 60 and 120°, are preferred. Fig. 12 further shows a very schematic sensitivity characteristic for each diaphragm 11 to 16, which additionally has either the letter F or the letter R. F stands for front and R stands for rear.The different sensitivity characteristics of the individual membranes, each of which typically has a counter electrode, are also arranged against each other.

[0102] As further shown, for example, in Fig. 11 or Fig. 12, it is preferred that the diaphragms of the various diaphragm pairs are arranged directly opposite one another, parallel to one another, and aligned with one another, wherein furthermore a distance between the two diaphragm pairs is small and preferably less than 2 cm. Furthermore, it is preferred that the distance for each diaphragm pair is substantially the same within a tolerance. Fig. 11 further shows output lines for each diaphragm. In particular, the first sub-microphone 1 is designed such that in response to a deflection of the first diaphragm 11, a first diaphragm signal is delivered, and in response to a deflection of the second diaphragm, a second diaphragm signal is delivered, which has a specific phase relationship to the first diaphragm signal, which results from the arrangement of the diaphragms or the wiring or the recorded sound field.

[0103] Furthermore, the second sub-microphone 2, which has the two diaphragms 13, 14, also has output lines to deliver a third diaphragm signal from the third diaphragm 13 and a fourth diaphragm signal from the fourth diaphragm 14. Furthermore, depending on the implementation, the third sub-microphone is also configured to deliver a fifth diaphragm signal in response to a deflection of the fifth diaphragm 15 and a sixth diaphragm signal in response to a deflection of the diaphragm 16 in the third spatial axis, for example, in the z-direction.

[0104] The first sub-microphone, the second sub-microphone, and, if present, the third sub-microphone are configured to combine the corresponding diaphragm signals of the diaphragms of the diaphragm pair. This is illustrated in Fig. 13a by a schematic combiner shown at 30 as a block for all two or three sub-microphones, respectively. However, a corresponding individual combiner, as shown, for example, at 31 in Fig. 13b, can be provided for each individual sub-microphone, so that the diaphragm signals of one sub-microphone are always combined, but that diaphragm signals from other sub-microphones are not combined with one another, at least for the generation of a first differential output signal 21 for the first sub-microphone, a second differential output signal 22 for the second sub-microphone, and a third differential output signal 23 for the third sub-microphone.However, in preferred embodiments, the combiner 30 is further configured to form not only the difference signals 21, 22, 23, but also common-mode or common-mode signals or CM signals 24. These CM signals 24 can, for example, be merely individual component signals X, Y, Z, as are known from Ambisonics technology, or an omnidirectional signal, which is obtained, for example, when the membrane signals of all individual membranes are added together without phase shifting of individual membrane signals.

[0105] To generate a differential signal, such as the differential output signal Diffx 21, the combiner 30 is configured to combine the first membrane signal 11 and the second membrane signal 12 with a modified first phase relationship. The first differential output signal Diffx 21 is thus assigned to the first spatial axis, for example, the x-axis.

[0106] Furthermore, the second sub-microphone is configured to combine the second diaphragm signal 13 and the third diaphragm signal 14 with a modified second phase relationship to deliver a second differential output signal Diffy, which is shown at 22 in Fig. 13a and is associated with the second spatial axis y. Furthermore, the third sub-microphone is configured to combine the fifth diaphragm signal 15 and the sixth diaphragm signal 16 with a phase relationship modified from the third phase relationship to deliver a third differential output signal, which is shown at 23 in Fig. 13a and is associated with the spatial axis z.

[0107] Preferably, the combination is performed as schematically illustrated in Fig. 13c. To change the first phase relationship between the first membrane signal 11 and the second membrane signal 12, Fig. 13c schematically shows a phase change element 40, which preferably has a phase value of 180°, wherein the phase angle of the phase element can be in the range between 90 and 270°. However, the preferred range is 170° to 190°, or 180° in the most preferred embodiment.

[0108] The phase-changing device 41 is provided to change the second phase relationship for the second sub-microphone, so that an addition, as shown schematically in Fig. 13c, takes place with a changed second phase relationship. Furthermore, a phase-changing element 42 is also provided for the third sub-microphone, which changes the third phase relationship between the diaphragm signals 15, 16 and adds the signals with the changed third phase relationship to obtain the third differential output signal Diffz 23 from Fig. 13c.

[0109] As already illustrated by reference numeral 24 in Fig. 13a, the combiner is also designed to form conventional common-mode signals. To form a CM-z signal, the fifth membrane signal 15 and the sixth membrane signal 16 are added together with the original third phase relationship, i.e., without the effect of a phase element 42, for example.

[0110] A corresponding procedure is used to obtain a conventional y-directional component of a directional microphone by adding the membrane signals of the second membrane pair, 13, 14, with the original phase relation, i.e., without the effect of a phase element 41. Analogously, an x-component of a directional microphone is also obtained by adding the two directional characteristics, i.e., for the front membrane 11 and the rear membrane 12, again without the effect of a phase element 40.

[0111] A total omnidirectional signal can be obtained when all six membrane signals are added together in their original first, second and third phase relation, this omnidirectional signal being referred to, for example, as a W signal or P signal, as is known from Ambisonics technology, or for a B-format signal which has an omnidirectional component and a directional component in the X direction, a directional component in the Y direction and a Z component in the Z direction.

[0112] In contrast to such a B-format signal, the microphone according to the invention provides, in addition to these signals or alternatively to these signals, difference signals for the individual directions, i.e. signals that result when a difference is formed between the front and rear directional characteristics in order to capture the sound field that prevails, so to speak, laterally with respect to the oppositely arranged membranes, i.e., for example, above and below the two membranes 11, 12 from Fig. 11.

[0113] The change between the first phase relation on the left in Fig. 13c and the second phase relation on the right in Fig. 13c before the corresponding addition can be achieved by an actually provided phase shifter, a delay line, a phase reversal, or even a phase reversal. The latter case of phase reversal is used for a preferred embodiment in which the membrane signals are transmitted as symmetrical signals between a positive line 11a and a negative line 11b. Such a schematic representation of the membrane signal 11 is shown in Fig. 13b, where the "line" 11 in Fig. 13c corresponds to the positive individual line 11a, the negative individual line 11b, and a ground (GND) 11c. The same applies to the second membrane signal 12, which in turn consists of a positive line 12a, a negative line 12b, and a common ground 12c.The actual membrane signal is transmitted as the difference between the positive and negative lines, as is known for symmetrical line transmission.

[0114] In order to combine such a signal, the combiner 30 is designed as shown in Fig. 13b for a single combiner 31. In its corresponding implementation, the single combiner 31 would be provided for each of the three sub-microphones 1, 2, 3 of Fig. 11. The single combiner 31 has two inputs 32, 34 for the positive potential and two inputs 33, 35 for the negative potential as well as one (or two) ground inputs 38 for the ground potential GND. In order to achieve the phase reversal, as shown in Fig. 13c by the element 40, 41, or 42, respectively, in the embodiment shown in Fig. 13b with symmetrical signal transmission, the positive and negative lines are reversed, as shown on the left in Fig. 13b for the membrane signal 12. The positive line 12b is connected to the negative input 35 and the negative line 12b is connected to the positive input 34.At the output, the single combiner then supplies the difference signal 21, designated Diffx, which in turn is transmitted as a difference signal between the positive line 36 and the negative line 37, wherein an output ground 39 (GND) is also provided.

[0115] Although in Fig. 13b such a single combiner is shown only for the first sub-microphone, it is preferred to use such a single combiner also for the second sub-microphone and for the third sub-microphone.

[0116] Fig. 14 shows a preferred embodiment of the microphone in which the three sub-microphones are all held by a diaphragm holder 50, each sub-microphone having an elongated housing, with the diaphragm pairs arranged in the corresponding tip of the sub-microphone, preferably protected from the outside world by a permeable grid. In particular, the two diaphragms of the first sub-microphone 1 are arranged in the yz-plane so that a deflection in the x-direction is achieved. Furthermore, the two diaphragms of the second sub-microphone 2 are arranged in the x-z-plane so as to achieve a deflection in the y-direction, i.e. in the second spatial axis. Furthermore, the two diaphragms of the third sub-microphone 3 are arranged in the xy-plane so as to be deflected by sound in the z-direction.The individual sub-microphones also have an output line that either carries the individual membrane signals to the outside, or that already carries the differential output signal 21, 22, or 23 (not shown in Fig. 14). Depending on which electronics are already installed in the elongated housing of the corresponding sub-microphone, the individual lines can also carry the conventional common-mode components to the outside in the individual directions, as shown at 24b, 24c for x and y. The signal Z, which will be explained with reference to Fig. 7, is not shown in Fig. 14, but can already be generated by the third sub-microphone 3, preferably within the elongated housing.

[0117] The three sub-microphones are designed such that each diaphragm has a counter electrode, so that in total, the microphone according to the invention shown in Fig. 14 has six individual diaphragms and six corresponding counter electrodes. These counter electrodes together form a separate condenser microphone for each diaphragm, whereby, depending on the implementation, a condenser or electret foil can also be applied to the corresponding counter electrode in order to have six individual condenser or electret microphones in the arrangement shown in Fig. 14. The "tips" of the three sub-microphones 1, 2, 3 are directed towards a common area or a common axis in order to position the three diaphragm pairs as close to one another as possible in order to be able to detect a rotational oscillation, represented by its three individual components that indicate the direction of rotation. To achieve this, a microphone holder is preferably provided, which is indicated at 50 in Fig.14. The microphone holder is triangular or can also be kite-shaped or have another shape. However, it comprises two sides that are at a 90° angle to each other in order to align the sub-microphone 1 and the sub-microphone 2 at a 90° angle to each other. For this purpose, a first holder 51 is provided, which is located on the first side of the two sides arranged at right angles to each other, and a second holder 52 is provided, which is located on the other side of the two sides arranged at right angles to each other. To hold the third microphone, a third holder 53 is provided, which is formed at the bisector of the 90° angle of the two sides on which the first holder 51 and the second holder 52 are provided and protrudes from the plane of the drawing in order to bring the third sub-microphone as close as possible, with regard to its sensitive microphone tip, to the two microphone tips of the first and second sub-microphones.The holders 51, 52 and 53 are preferably designed as clips in order to be able to mount the individual sub-microphones without tools. However, other holding devices can also be provided to hold the elongated sub-microphones in the appropriate angular shape so that the membrane pairs are aligned as shown in Fig. 11 or Fig. 12. For other arrangements in which the exact right-angled arrangement between the individual microphones is not critical, the microphones can also be arranged at an angle between 70 and 110° or the third holder 53 or the third sub-microphone can be arranged at an angle between 30 and 60° with respect to the first holder or the second holder. The microphone holder 50 is also attached to a stand 54, which is shown schematically in Fig. 14.Instead of the tripod 54, the microphone can also be suspended from the ceiling using a cable structure to keep the lower area free, for example, when recording a stage. Instead of the elastic clips shown for the individual holders in Fig. 14, magnetic holders, snap-in elements, or other holders can also be used.

[0118] A newly recorded sound sample according to the invention comprises, at least for a first channel, a first pressure signal, a first directional difference signal, and a second directional difference signal; and for a second channel, a second pressure signal, a third directional difference signal, and a fourth directional difference signal. As described and illustrated in Fig. 1c, the sound sample may comprise a large number of additional channels or pressure signals and / or directional difference signals. It may further be recorded on a physical storage medium, such as a non-volatile or digital storage medium.

[0119] Although some aspects have been described in connection with a device, it is understood that these aspects also represent a description of the corresponding method, so that a block or component of a device can also be understood as a corresponding method step or as a feature of a method step. Similarly, aspects described in connection with or as a method step also represent a description of a corresponding block, detail, or feature of a corresponding device.

Claims

Claims: Device for re-recording an existing sound piece with a first number of channels (111, 112), comprising the following features: a loudspeaker arrangement for reproducing the first number of channels with a first number of loudspeaker systems (201, 202, 203) in a playback space (110), wherein the first number is equal to 1 or greater than 1, and wherein the one loudspeaker system (203) or the loudspeaker systems (201, 202, 203) of the first number of loudspeaker systems are configured to generate for each channel (111, 112, 113) of the first number of channels both a translational sound field and a rotational sound field in the playback space (110);a microphone arrangement with a second number of microphone systems (221-231) arranged at different microphone positions in the playback space (110), each microphone system of the second number of microphone systems (221-231) being configured to detect a pressure signal (OD) and furthermore a directional differential signal (D);and an interface (150) for outputting or recording the pressure signal and the directed differential signal or a differential signal derived from the directed differential signal as a newly recorded audio clip (240). Device according to claim 1, wherein the existing audio clip has a mono channel (113) and the first number is equal to 1, wherein a single loudspeaker system (203) is arranged in a center loudspeaker position relative to a listener position (114) in the playback space (110), the second number being greater than the first number, and wherein the microphone arrangement comprises at least a first microphone system (201) at a first left microphone position relative to the listener position (114) in the playback space and a second microphone system (202) at a second right microphone position relative to the listener position (114) in the playback space (110).

3. Device according to claim 1 or 2, wherein the existing sound piece has a first left channel (111) and a second right channel (112) and the first number is equal to 2 or 3, wherein a first loudspeaker system of the first number of loudspeaker systems is arranged at a left loudspeaker position with respect to a listener position (114) in the playback space, wherein a second loudspeaker system of the second number of loudspeaker systems is arranged at a right loudspeaker position with respect to the listener position (114) in the playback space (110), and wherein the microphone arrangement has at least a first microphone system (201) at a first left microphone position with respect to the listener position (114) in the playback space (110) and a second microphone system at a second right microphone position with respect to the listener position (114) in the playback space.

4. Device according to claim 3, wherein the existing sound piece further comprises a center channel (113) and the first number is equal to 3, wherein a further loudspeaker system (203) is arranged at a center loudspeaker position with respect to the listener position (114) in the playback space (110) in order to generate both a translational sound field and a rotational sound field for the center channel.

5. Device according to claim 4, wherein the interface (150) is configured to copy the center channel (113) into the newly recorded sound piece as the center channel of the newly recorded sound piece (151).

6. Device according to one of the preceding claims, wherein the second number is greater than the first number, and wherein a third number of channels in the newly recorded sound piece (240) is at least twice as large as the second number, wherein the respective pressure signal represents a separate channel in the third number and the respective directed differential signal or the differential signal derived from the directed differential signal also represents a separate channel in the third number of channels. Device according to one of the preceding claims, wherein each channel of the first number of channels (111, 112, 113) is assigned its own distinct playback position in a first playback format, and wherein one or more loudspeaker systems of the first number of loudspeaker systems are arranged at the playback position(s) of the first playback format in the playback space (110), and wherein each microphone system (221-231) is assigned its own playback position of a second playback format, wherein the second playback format differs from the first playback format.Device according to claim 7, wherein the first playback format is a mono format, a stereo format or a 3-channel format with a left channel (111), a center channel (113) and a right channel (112), or wherein the second playback format is an ITU-recommended playback format comprising at least one layer (501, 502, 503) from a group of layers (501, 502, 503) comprising a lower layer (501), a middle layer (502) and an upper layer, wherein at least one loudspeaker position of a first loudspeaker system belongs to the upper layer (503) and at least one loudspeaker position of a second loudspeaker system belongs to the middle layer (502), or is a Dolby Atmos playback format or is an MPEG playback format, or comprises a part of one of the aforementioned playback formats.Device according to one of the preceding claims, wherein the existing sound piece has been recorded in an original playback room with a specific original room acoustics, and wherein the playback room (110) is the same playback room as the original playback room or is another playback room whose room acoustics are the same or similar to those of the original room acoustics. Device according to one of the preceding claims, wherein each loudspeaker system (201, 202, 203) of the first number of loudspeaker systems has the following features: a first sound transducer system (201a, 203a, 202a); a second sound transducer system (201b, 203b, 202b); a first input for a first audio signal (111a, 113a, 112a) which can be supplied to the first sound transducer system (201a, 203a, 202a) for generating the translation sound field; and a second input for a second audio signal (111b, 113b, 112b) which is different from the first audio signal and can be supplied to the second sound transducer system (201b, 203b, 202b) for generating the rotational sound field, wherein the second sound transducer system is configured to emit sound omnidirectionally or with a lower directivity or quality than the first sound transducer system.The device according to claim 10, wherein the first sound transducer system (201a, 203a, 202a) comprises one or more loudspeakers (602) oriented in one direction and housed in an enclosure (300), or wherein the second sound transducer system comprises a plurality of individual loudspeakers (318), each oriented in a different direction, wherein a direction in which a loudspeaker of the plurality of individual loudspeakers (318) is oriented differs from the direction in which the loudspeakers (602) of the first sound transducer system are oriented. The device according to claim 10 or 11, wherein the second sound transducer system comprises a circular support structure (316) on which the individual loudspeakers (318) are arranged as unenclosed loudspeakers with loudspeaker diaphragms, such that the loudspeaker diaphragms emit in different directions during operation. A device according to any one of claims 10 to 12, wherein the second sound transducer system (201b, 203b, 202b) comprises at least six individual loudspeakers, or a dodecahedron with 12 individual faces, wherein a single loudspeaker of the plurality of loudspeakers of the second sound transducer system is arranged on each of the 12 individual faces of the dodecahedron. A device according to any one of claims 10 to 13, further comprising a signal processing element (100) for generating from each channel of the first number of channels of the existing sound piece the first audio signal and the second audio signal for the loudspeaker system assigned to the channel.Device according to claim 14, wherein the first set of channels includes a center channel, and wherein the signal processing element (100) is configured to derive both the first audio signal and the second audio signal for the loudspeaker system associated with the center channel from the center channel by attenuation (113c, 113d) or amplification. Device according to claim 15, wherein the signal processing element is configured to derive the first audio signal (113a) and the second audio signal (113b) such that the second audio signal (113b) has a lower power than the first audio signal (113a).Device according to one of claims 14 to 16, wherein the first number of channels comprises a left channel and a right channel, and wherein the signal processing element (100) is configured to generate the first audio signal for the loudspeaker system (201) associated with the left channel from the left channel without the right channel or with the right channel, wherein a power component of the right channel in the first audio signal (111a) is less than 1 / 10 of the power component of the left channel in the first audio signal (111a), and wherein the signal processing element (100) is configured to generate the second audio signal (112a) for the loudspeaker system (202) associated with the right channel from the right channel without the left channel or with the left channel, wherein a power component of the left channel in the first audio signal (112a) is less than 1 / 10 of the power component of the right channel in the first audio signal (112a). Device according to one of claims 14 to 16, wherein the first number of channels has a left channel and a right channel, and wherein the signal processing element (100) is configured to determine the second audio signal (111b) for the loudspeaker system (201) associated with the left channel (111), or the second audio signal (112b) for the loudspeaker system (202) associated with the right channel (112), from a difference (1011, 1012) between the left channel (111) and the right channel (112), or from an addition (1013, 1014) of the left and right channels, wherein one channel of the left or right channel has been reversed in its polarity (1015, 1016) before the addition.Device according to claim 18, wherein the signal processing element (100) is configured to use a result (1011, 1012) of the difference or addition independently of a correlation between the left channel and the right channel for generating the respective second audio signal (111b, 112b), or to amplify or attenuate (1031, 1032) the result (1011, 1012) of the difference or addition depending on the signal, e.g., depending on a correlation between the first channel and the second channel, and to use a result of the amplification or attenuation for generating the respective second audio signal (111b, 112b).Device according to claim 19, wherein the signal processing element (100) is configured to attenuate the left or right channel (1103, 1203) and to add the result of the attenuation during the generation of the respective second audio signal to the result of the difference or addition, or to the result of the amplification or attenuation (1102, 1202). Device according to any one of claims 10 to 20, wherein the loudspeaker system (201, 202, 203) comprises a power amplifier (201c, 201c, 203c, 203d, 202c, 202d) for the first audio signal and another for the second audio signal. Device according to any one of the preceding claims, wherein one or more of the microphone systems (221-231) of the microphone arrangement have the following features: a first partial microphone (1) with a first diaphragm pair comprising a first diaphragm (11) and a second diaphragm (12) arranged opposite each other; and a second partial microphone (2) with a second diaphragm pair comprising a third diaphragm (13) and a fourth diaphragm (14) arranged opposite each other, wherein the first diaphragm pair is arranged such that the first diaphragm (11) and the second diaphragm (12) are deflectable along a first spatial axis, wherein the second diaphragm pair is arranged such that the third diaphragm (13) and the fourth (14) are deflectable along a second spatial axis, and wherein the second spatial axis is different from the first spatial axis.The device according to claim 22, comprising the following features: a third partial microphone (3) with a third diaphragm pair comprising a fifth diaphragm (15) and a sixth diaphragm (16) arranged opposite each other, wherein the third diaphragm pair is arranged such that the fifth diaphragm (15) and the sixth diaphragm (16) are deflectable along a third spatial axis, the third spatial axis being different from the first and second spatial axes. The device according to claim 22 or 23, wherein the spatial axes are orthogonal to each other, or wherein there is an angle between two spatial axes that is between 60° and 120°. The device according to any one of claims 22 to 24. in which the diaphragms of the first diaphragm pair, the second diaphragm pair, and the third diaphragm pair are directly opposite each other, parallel to each other, aligned with each other, or arranged at a distance of less than 2 cm from each other. Device according to one of claims 22 to 25, in which the first partial microphone (1) is configured to provide a first diaphragm signal in response to a displacement of the first diaphragm (11), and to provide a second diaphragm signal in response to a displacement of the second diaphragm (12), wherein the first diaphragm signal and the second diaphragm signal have a first phase relation, wherein the first partial microphone (1) is configured to combine the first diaphragm signal and the second diaphragm signal with a modified first phase relation to provide a first differential output signal (21) that is associated with the first spatial axis, or in which the second partial microphone (2) is configuredto provide a third membrane signal in response to a displacement of the third membrane (13), and to provide a fourth membrane signal in response to a displacement of the fourth membrane (14), wherein the third membrane signal and the fourth membrane signal have a second phase relation to each other, and wherein the second partial microphone (2) is configured to combine the third membrane signal and the fourth membrane signal with a modified second phase relation (30) to provide a second differential output signal (22) associated with the second spatial axis, or wherein a third partial microphone (3) is configured to provide a fifth membrane signal in response to a displacement of a fifth membrane (15), and to provide a sixth membrane signal in response to a displacement of a sixth membrane, wherein the fifth membrane signal and the sixth membrane signal have a third phase relation, and wherein the third partial microphone (3) is configuredto combine the fifth membrane signal and the sixth membrane signal with a modified third phase relation (30) to provide a third differential output signal (23) that is associated with the third spatial axis. Device according to claim 26, in which the modified first phase relation differs by 180° from the first phase relation or differs by a phase between 150° and 210° from the first phase relation, or in which the modified second or third phase relation differs by 180° from the second or third phase relation or differs by a phase between 150° and 210° from the second or third phase relation, respectively. Device according to one of claims 25 or 26, in which the first membrane signal is transmitted as a symmetrical signal on a first positive line (11a) and a first negative line (11b), in which the second membrane signal is transmitted as a symmetrical signal on a second positive line (12a) and a second negative line (12b),wherein the first partial microphone (1) has a combiner (31) with a first positive input (32) and a first negative input (33) for the first diaphragm signal and with a second positive input (34) and a second negative input (35) for the second diaphragm signal, wherein the second negative line (12b) of the second diaphragm signal is connected to the second positive input (34) of the combiner (31), and wherein the second positive line (12a) of the second diaphragm signal is connected to the second negative input (35) of the combiner, and wherein the first positive line (11a) of the first diaphragm signal is connected to the first positive input (32) of the combiner (31), and wherein the first negative line (11b) of the first diaphragm signal is connected to the first negative input (33) of the combiner (31). Device according to any one of claims 25 to 28, wherein the first partial microphone (1) is configuredto add the first membrane signal and the second membrane signal in the first phase relation to provide a first common-mode output signal (24b), or, in which the second sub-microphone (1) is configured to add the third diaphragm signal and the fourth diaphragm signal in the second phase relation to provide a second common-mode output signal (24c), or in which a third sub-microphone is configured to add a fifth diaphragm signal and a sixth diaphragm signal in a third phase relation to provide a third common-mode output signal (24d), or which is configured to combine the first diaphragm signal, the second diaphragm signal, the third diaphragm signal, the fourth diaphragm signal and optionally the fifth diaphragm signal and the sixth diaphragm signal in the first, the second and optionally the third phase relation to provide an at least partially omnidirectional or omnidirectional common-mode output signal (24a).

30. Device according to one of claims 26 to 28, wherein the interface (150) is configured to output or record, for each microphone system or for each microphone system except one microphone system at a center microphone position, the pressure signal and the first difference signal, the second difference signal and the third difference signal as a newly recorded sound piece (240).

31. Device according to one of claims 26 to 28, wherein the interface (150) is configured to output or record the pressure signal for each microphone system and to combine at least two directed differential signals and to output a result of the combination as the differential signal derived from the directed differential signal or to perform two different combinations to output two differential signals derived from the directed differential signals.

32. Device according to one of claims 26 to 28, in which a microphone position of a microphone system (222, 223) is arranged next to a sound transducer system of a loudspeaker system (201, 202) for generating the rotational sound field, wherein the first partial microphone or the second partial microphone or the third partial microphone are configured to attenuate a diaphragm signal of a diaphragm directed towards a transducer of the transducer system before it is combined with another diaphragm signal. Method for re-recording an existing sound piece with a first number of channels (111, 112), comprising the following steps: Reproducing the first number of channels with a first number of loudspeaker systems (201, 202, 203) in a playback space (110), wherein the first number is equal to 1 or greater than 1, and wherein the one loudspeaker system (203) or the loudspeaker systems (201, 202, 203) of the first number of loudspeaker systems are configured to generate for each channel (111, 112, 113) of the first number of channels both a translational sound field and a rotational sound field in the playback space (110); Recording with a second number of microphone systems (221-231) arranged at different microphone positions in the playback space (110), each microphone system of the second number of microphone systems (221-231) being configured to detect a pressure signal (OD) and furthermore a directional differential signal (D); and Outputting or recording the pressure signal and the directed differential signal, or a differential signal derived from the directed differential signal, as a newly recorded tone piece (240). Newly recorded tone piece, comprising: for a first channel, a first pressure signal, a first directed differential signal, and a second directed differential signal; and for a second channel, a second pressure signal, a third directed differential signal, and a fourth directed differential signal.

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