EP3005737B1 - Mixing desk, method and computer program for providing a sound signal - Google Patents

Description

Embodiments of the present invention are concerned with an apparatus, method and computer program for providing a sound signal based on at least two source signals recorded by microphones located within a room or an acoustic scene.

More complex recordings or acoustic scenes are usually recorded using audio mixing consoles as far as the recording of the audio signals is concerned. The acoustic scene should be understood to mean any sound composition or any sound signal. To account for the fact that the acoustic signal or audio signal received at a listener position is typically from a variety of different sources, the term acoustic scene is used herein, an acoustic scene in which herein Of course, these senses can also be generated by only a single sound source. However, the character of such an acoustic scene is determined not only by the number or distribution of the sound sources generating them in a room, but also by the shape or geometry of the room itself. For example, in closed rooms, as a part of the room acoustics, the one listener will speak directly superimposed on the sound source reaching sound components caused by the boundary walls reflections that can be understood simplistic as, inter alia, delayed and attenuated copy of the direct sound components.

In such environments, an audio mixing console is often used for the production of audio material, having a plurality of channels, each associated with one of many microphones, which in turn are located within the acoustic scene, for example within a concert hall or the like. The individual audio or source signals can be present both analog and digital, for example as a series of digital sample values, the sample values being equidistant in time and corresponding in each case to an amplitude of the sampled audio signal. Depending on the audio signal used, such a mixer can therefore for example be dedicated Hardware or as a software component on a PC or a programmable CPU implemented, provided that the audio signals are digital. Electrical sound signals that can be processed with such audio mixing consoles may, in addition to microphones, also originate from other players, for example from instruments and effect devices or the like. Each individual sound signal or each audio signal to be processed can be assigned to a separate channel strip on the mixer, wherein a channel strip can provide several functionalities for changing the sound of the associated audio signal, for example a change in volume, filtering, mixing with other channel strips, a distribution or a splitting of the relevant channel or the like.

When recording complex audio scenes, such as concert recordings, it is often the task to produce the sound signal or the mixed recording so that a listener when listening to the recording as faithful as possible true sound produced. In this case, this so-called mixing of the originally recorded microphone or source signals for different reproduction configurations may have to be different, for example for different numbers of output channels or loudspeakers. Examples would be a stereo configuration and multi-channel configurations such as 4.0, 5.1 or the like. In order to create such a spatial sound mixing or mixing, so far for each sound source or for each microphone or source signal, the volume on the respective channel strip is set so that for the desired Abhörkonfiguration created by the sound engineer room. This is mainly achieved by so-called panning algorithms, the volume between multiple playback channels or speakers is distributed so that a phantom sound source between the speakers is created to achieve a spatial impression. This means, for example, that the listener will create the impression that the reproduced object is located spatially between the speakers due to the different volume levels for the individual playback channels. In order to make this possible, each channel has to be adjusted manually based on the real position of the recording microphone within the acoustic scene and compared with a partially significant number of other microphones.

Even more complicated and time consuming or costly such sound mixtures, if the listener to create the impression, move the recorded sound source yourself. Then, for each of the time-varying spatial configurations or for each time step within the movement of a sound source, the volume for all channel strips involved must be readjusted manually, which is not only extremely time-consuming, but also prone to error.

In some scenarios, for example when recording a symphony orchestra, a high number of microphone or source signals of, for example, over 100 are recorded simultaneously and possibly processed in real time to a sound mix. In order to achieve such a spatial blending, so far the operator or sound engineer on a conventional mixing console at least in advance of the actual recording, the spatial relationship between the individual microphone or source signals thereby generate, this first the positions of the microphones and their assignment to The individual channel strips are noted by hand in order to regulate the volumes and possibly other parameters such as a distribution of volumes for several channels or reverb (pan and reverb) of the individual channel strips so that the sound mix at the desired listening position or for a desired speaker arrangement achieved the desired spatial effect. In a symphony orchestra with more than 100 instruments, each of which is recorded separately as a direct source signal, this can be an almost impossible task. In order to replicate a realistic spatial arrangement of the recorded source signals of the microphones in the mixer even after recording, the positions of the microphones have been sketched by hand or numbered their positions to then reproduce the spatial sound mixing in a complex procedure by adjusting the volume of all individual channels , However, with a very large number of microphone signals to be recorded, it is not only the subsequent mixing of a successful recording that poses a great challenge.

Rather, with a large number of source signals to be recorded, it is already a difficult task to ensure that all microphone signals are delivered without interference to the mixer or to a software used for audio mixing. So far, this has to be checked by the sound engineer or an operator of a mixer listening to each channel strips separately or checked, which is very time consuming and in the event of the occurrence of an interference signal whose origin can not be located immediately, a time-consuming troubleshooting result. When listening in or switching on and off of individual channels or source signals must also be paid close attention will be that the extra recordings that associated the microphone signal and its position with the mixer's channel during recording are sound. But this control can take several hours for large shots, and moreover, errors that are made in the complex control, in retrospect difficult or impossible to compensate after the recording is completed.

Thus, when recording acoustic scenes by means of at least two microphones, there is a need to provide a concept that can make the making and / or mixing of the recording more efficient and less susceptible to error.

This object is achieved by a mixer, a tone signal generator, a method and a computer program each having the features of the independent claims. Advantageous embodiments and further developments are the subject of the dependent claims.

In particular, some embodiments of the present invention enable this by using a sound signal generator to provide a sound signal for a virtual listening position within a room in which an acoustic scene from at least a first microphone at a first known position within the room as a first source signal and at least one second microphone is recorded at a second known position within the room as a second source signal. To facilitate this, the audio signal generator has an input interface to receive the first and second source signals received by the first microphone and by the second microphone. A geometry processor within the audio signal generator is configured to obtain, based on the first position and the virtual listening position, first geometry information comprising a first distance between the first known position and the virtual listening position (202), and a first geometry information based on the second position and the virtual listening position second distance between the second known position and the virtual listening position (202) to determine second geometry information so that they can be taken into account by a signal generator, which serves to provide the audio signal. For this purpose, the signal generator is designed to at least the first source signal and the second source signal according to a combination rule combine to get the sound signal. In this case, according to the exemplary embodiments of the present invention, the combination is performed using the first geometry information and the second geometry information. That is, according to the embodiments of the present invention, for a virtual listening position at which no real microphone needs to be located in the acoustic scene to be mixed, a sound signal may be generated from two source signals recorded by real microphones spatial perception at the location of the virtual listening position correspond or may resemble. This can be achieved, for example, in particular by using geometry information, which for example indicates the relative position between the positions of the real microphones and the virtual monitoring position, directly in the provision or generation of the sound signal for the virtual listening position. This can therefore be possible without complex calculations, so that the provision of the sound signal can be done in real time or approximately in real time.

The direct use of geometry information to generate a sound signal for a virtual listening position may also make it possible to create a sound mixture by simply shifting the position or coordinates of the virtual listening position, without the possibly large number of source signals individually and manually adjusted would have to be. The creation of an individual sound mixture can, for example, also enable an efficient control of the setup prior to the actual recording, wherein, for example, the recording quality or the arrangement of the real microphones in the scene can be controlled by virtue of the virtual listening position within the acoustic scene or within the acoustic scene is freely moved in the acoustic space, so that a sound engineer can immediately receive an automatic acoustic feedback, whether the individual microphones are wired correctly or whether they work properly. For example, the functionality of each individual microphone can be checked without having to hide all of the other microphones if the virtual listening position is brought close to the position of one of the real microphones, so that its share of the provided sound signal dominates. This in turn allows control of the source or audio signal recorded by the respective microphone.

Further, embodiments of the invention may eventually enable even when an error occurs during live recording by quickly identifying the error intervene so quickly and fix the error can, for example, by replacing a microphone or a cable that at least large parts of the concert can still be recorded correctly.

Further, according to the embodiments of the present invention, it may not be necessary to record the position of a plurality of microphones used for recording an acoustic scene independently of the source signals, to subsequently add the spatial arrangement of the recording microphones imitate the mix of the signal representing the acoustic scene. Rather, according to some embodiments, the previously known positions of the microphones recording the source signals within the acoustic space can be directly taken into account as control parameters or characteristics of individual channel strips in an audio mixing console and preserved or recorded together with the source signal.

Some embodiments of the present invention are a mixer for processing at least a first and a second source signal and providing a mixed audio signal, the mixer comprising a tone signal generator for providing a sound signal for a virtual listening position within a room in which an acoustic scene of at least a first The microphone is recorded at a first known location within the room as the first source signal and from at least one second microphone at a second known location within the room as the second source signal, the audio signal generator comprising: an input interface adapted to receive the one of the second source signal receiving the first microphone recorded first source signal and received by the second microphone second source signal; a geometry processor configured to determine first geometry information based on the first position and the virtual listening position, and determine second geometry information based on the second position and the virtual listening position; and a signal generator for providing the sound signal, wherein the signal generator is configured to combine at least the first source signal and the second source signal according to a combination rule using the first geometry information and the second geometry information. This may allow an operator of a mixing console to easily, efficiently, and without a high probability of error, control the microphone cabling prior to recording, for example.

In accordance with some embodiments, the mixer further includes a user interface configured to display a graphical representation of the locations of a plurality of microphones as well as one or more virtual listening positions. That is, some embodiments of mixing consoles also allow a graphical representation of the geometric relationships in the recording of the acoustic scene, which can enable a sound engineer in a simple and intuitive way to create a spatial mix or a microphone setup to control or build up or adjust for recording a complex acoustic scene.

According to some further embodiments, a mixing console additionally comprises an input device which is designed to input or change at least the virtual listening position, in particular by direct interaction or influencing of the graphical representation of the virtual listening position. This makes it possible in a particularly intuitive manner to carry out a control of individual listening positions or of microphones associated with these positions, for example by using the virtual monitoring position within the acoustic scene or the acoustic space with the mouse or by means of the finger and a touch-sensitive screen (FIG. Touchscreen) can be moved to the place of interest.

Some other mixer embodiments also allow to characterize each of the microphones as belonging to a particular one of several different microphone types via the input interface. In particular, a microphone type can correspond to microphones which predominantly record a direct sound component due to their geometric relative position with respect to the objects or sound sources of the acoustic scene to be recorded. A second type of microphone may for the same reason primarily identify a diffuse sound-absorbing microphone. The possibility of assigning the individual microphones to different types can serve, for example, to combine the source signals, which are recorded by the different types, with respectively different combination rules in order to obtain the sound signal for the virtual listening position.

This may, according to some embodiments, be used in particular for different combination or superposition rules for microphones, which are predominantly To record diffuse sound and use for those microphones that record predominantly direct sound, in order to arrive at a natural sound impression or a signal that has advantageous properties for the given requirement. For example, in some embodiments where the audio signal is generated to form a weighted sum of at least a first and a second source signal, the weights for the different microphone types are determined differently. For example, in microphones that predominantly record direct sound, a drop in volume corresponding to the reality can be implemented with increasing distance from the microphone via a suitably chosen weighting factor. In some embodiments, the weight is proportional to the inverse of a power of the distance of the microphone to the virtual listening position. According to some embodiments, the weight is proportional to the inverse of the distance, which corresponds to the sound propagation of an idealized point sound source. According to some embodiments, for microphones that are associated with the first type of microphone, that is the recording of direct sound, the weighting factors are proportional to the near field radius multiplied inverse of the distance of the microphone to the virtual listening position. This can lead to an improved perception of the sound signal by taking into account the assumed influence of a near field radius, within which a constant volume of the source signal is assumed.

According to some embodiments of the invention, for microphones which are assigned to a second microphone type and by means of which predominantly diffuse sound components are recorded, the sound signal is generated from the recorded source signals x ₁ and x ₂ by forming a weighted sum, the weights g ₁ and g ₂ depend on the relative positions of the microphones and at the same time fulfill an additional boundary condition. In particular, according to some embodiments of the present invention, the sum of the weights G = g ₁ + g ₂ or a quadratic sum of the weights G2 = g ₁ ² + g ₂ ^{2 is} constant and in particular one. This can lead to a combination of the source signals in which a volume of the generated sound signal for different relative positions between the microphones at least approximately corresponds to a volume of each of the source signals, which in turn can lead to a good perceptual quality of the generated audio signal, since the diffuse signal components within a acoustic volume have approximately identical volume.

According to some embodiments of the present invention, first of all a first intermediate signal and a second intermediate signal are formed from the source signals by means of two weighted sums of different weights. From the first and second intermediate signal, the sound signal is then determined by means of a further weighted sum, wherein the weights are dependent on a correlation coefficient between the first and the second source signal. This can make it possible, depending on the similarity of the two recorded source signals, to combine combination rules or panning methods in such a weighted manner that volume increases, as may occur in principle depending on the selected method and the signals to be combined, are further reduced. This may possibly lead to an overall volume of the generated sound signal remaining approximately constant independently of the combined signal forms, so that the mediated spatial impression also largely corresponds to what is desired without a priori knowledge of the source signal.

According to some further embodiments, in areas in which the virtual listening position is surrounded by three microphones each recording a source signal, the sound signals, in particular as regards their diffuse sound components, are formed using the three source signals. The provision of the audio signal comprises generating a weighted sum of the three recorded source signals. The microphones associated with the source signals form a triangle, the weights for a source signal being determined based on a perpendicular projection of the virtual listening position to the height of the triangle passing through the position of the respective microphone. Different methods of determining the weights may be used. Nevertheless, the volume can remain approximately unchanged, even if three instead of only two source signals are combined, which can contribute to a more sonically realistic reproduction of the sound field at the virtual listening position.

According to some embodiments of the present invention, either the first or the second source signal is delayed by a delay time prior to combining the two source signals if a comparison of the first geometry information and the second geometry information satisfies a predetermined criterion, especially if the two distances are less than a minimum allowable distance differ from each other. This can make it possible to produce the sound signals without causing color discoloration, possibly due to the superposition of a signal in a small spatial Distance from each other was recorded, could be generated. More particularly, in some embodiments, each of the source signals used is efficiently decelerated such that its propagation time or latency corresponds to the maximum signal propagation time from the location of all participating microphones to the virtual listening position such that destructive interference of similar or identical signals is avoided by a forced identical signal propagation time can.

According to some further embodiments, directional dependencies are also taken into account in the superposition or weighted summation of the source signals, that is, the virtual listening position can be assigned a preferred direction and a directional characteristic specified with respect to the preferred direction. This can make it possible to achieve a realistic effect when generating the sound signal by additionally taking into account a known directional characteristic, for example of a real microphone or of the human ear.

Figur 1:

Ein Ausführungsbeispiel eines Tonsignalerzeugers;

Figur 2:

Eine Illustration einer akustischen Szene, deren Quellsignale mit Ausführungsbeispielen von Tonsignalerzeugern verarbeitet werden;

Figur 3:

Ein Beispiel für eine Kombinationsregel zum Erzeugen eines Tonsignals gemäß einigen Ausführungsbeispielen der Erfindung;

Figur 4:

Eine Illustration zur Verdeutlichung eines weiteren Beispiels einer möglichen Kombinationsregel;

Figur 5:

Eine grafische Illustration einer Kombinationsregel zur Verwendung mit drei Quellsignalen;

Figur 6:

Eine Illustration einer weiteren Kombinationsregel;

Figur 7:

Eine Illustration einer richtungsabhängigen Kombinationsregel;

Figur 8:

Eine schematische Darstellung eines Ausführungsbeispiels eines Mischpults;

Figur 9:

Eine schematische Darstellung eines Ausführungsbeispiels eines Verfahrens zum Erzeugen eines Tonsignals; und

Figur 10:

Eine schematische Darstellung eines Ausführungsbeispiels einer Benutzerschnittstelle.

Embodiments of the present invention will be explained below with reference to the accompanying figures. Show it:

FIG. 1:

An embodiment of a tone signal generator;

FIG. 2:

An illustration of an acoustic scene whose source signals are processed with embodiments of audio signal generators;

FIG. 3:

An example of a combination rule for generating a sound signal according to some embodiments of the invention;

FIG. 4:

An illustration to illustrate another example of a possible combination rule;

FIG. 5:

A graphical illustration of a combination rule for use with three source signals;

FIG. 6:

An illustration of another combination rule;

FIG. 7:

An illustration of a directional combination rule;

FIG. 8:

A schematic representation of an embodiment of a mixing console;

FIG. 9:

A schematic representation of an embodiment of a method for generating a sound signal; and

FIG. 10:

A schematic representation of an embodiment of a user interface.

Various embodiments will now be described in more detail with reference to the accompanying drawings, in which some embodiments are illustrated. In the figures, the thickness dimensions of lines, layers and / or regions may be exaggerated for the sake of clarity.

In the following description of the attached figures, which show only some exemplary embodiments, like reference characters may designate the same or similar components. Further, summary reference numerals may be used for components and objects that occur multiple times in one embodiment or in a drawing but are described together in terms of one or more features. Components or objects which are described by the same or by the same reference numerals may be the same, but possibly also different, in terms of individual, several or all features, for example their dimensions, unless otherwise explicitly or implicitly stated in the description.

Although embodiments may be modified and changed in various ways, exemplary embodiments are illustrated in the figures as examples and will be described in detail herein. It should be understood, however, that it is not intended to limit embodiments to the particular forms disclosed, but that embodiments are intended to cover all functional and / or structural modifications, equivalents and alternatives that are within the scope of the invention. Like reference numerals designate like or similar elements throughout the description of the figures.

Note that an element referred to as being "connected" or "coupled" to another element may be directly connected or coupled to the other element, or intervening elements may be present. Conversely, when an element is referred to as being "directly connected" or "directly coupled" to another element, there are no intervening elements. Other terms used to describe the relationship between elements should be similar Be interpreted (eg, "between" versus "directly in between", "adjacent" to "directly adjacent", etc.).

The terminology used herein is for the purpose of describing particular embodiments only and is not intended to limit the embodiments. As used herein, the singular forms "a," "a," "an," and "the" are also meant to include the plural forms unless the context clearly indicates otherwise. Furthermore, it should be understood that the terms such as e.g. "including," "including," "having," and / or "having," as used herein, indicates the presence of said features, integers, steps, operations, elements, and / or components, but the presence or addition of a resp one or more features, integers, steps, operations, elements, components, and / or groups thereof.

Unless defined otherwise, all terms (including technical and scientific terms) used herein have the same meaning as commonly assigned to one of ordinary skill in the art to which the embodiments pertain. Further, it should be understood that terms, e.g. those that are defined in commonly used dictionaries are to be interpreted as having the meaning consistent with their meaning in the context of the relevant art, and not to be interpreted in an idealized or overly formal sense, unless this is so is explicitly defined.

Figur1 FIG. 3 shows a schematic representation of an exemplary embodiment of a tone signal generator 100 which comprises an input interface 102, a geometry processor 104 and a signal generator 106. The sound signal generator 100 serves to provide a sound signal for a virtual listening position 202 within a space 200 included in FIG. 1 is indicated only schematically. In the space 200, an acoustic scene is recorded by means of at least a first microphone 204 and a second microphone 206. The source 208 of the acoustic scene is here illustrated only schematically as an area within the space 200 within which a plurality of sound sources may be located, leading to a sound field referred to as an acoustic scene within the space 200, which in turn may be accessed by means of the Microphones 204 and 206 is recorded.

The input interface 102 is configured to receive a first source signal 210 received by the first microphone 204 and a second source signal 212 received by the second microphone 206. In this case, the first and second source signals 210 and 212 can be both analog and digital signals, which can be transmitted both encoded and uncoded by the microphones. That is, according to some embodiments, the source signals 210 and 212 may already be encoded or compressed according to a compression method such as the Advanced Audio Codec (AAC), MPEG 1, Layer 3 (MP3), or the like.

The first and second mics 204 and 206 are located at known locations within the space 200, which are also known to the geometry processor 104. The geometry processor 104 also knows the position or the coordinates of the virtual listening position 202 and is designed to determine first geometry information 110 from the first position of the first microphone 204 and the virtual listening position 202. The geometry processor 104 is further configured to determine second geometry information 112 from the second position and the virtual listening position 202.

An example of such geometry information is, without being exhaustive, a distance between the first position and the virtual listening position 202 or a relative orientation between a preferred direction associated with the virtual listening position 202 and a position of one of the microphones 204 or 206 Of course, the geometry may be described in any manner, for example by means of Cartesian coordinates, spherical coordinates or cylindrical coordinates in a one-, two- or three-dimensional space. In other words, the first geometry information may include a first distance between the first known position and the virtual listening position and the second geometry information comprises a second distance between the second known position and the virtual listening position.

The signal generator is configured to provide the audio signal combining the first source signal 210 and the second source signal 212, the combination following a combination rule according to which both the first geometry information 110 and the second geometry information 112 are considered.

The sound signal 120 is thus obtained from the first and the second source signals 210 and 212, in which case the first and the second geometry information 110 and 112 are used. That is, information about the geometric characteristics or relationships between the virtual listening position 12 and the positions of the microphones 204 and 206 are used directly to determine the sound signal 120.

By varying the virtual listening position 202, a sound signal may thus possibly be obtained in a simple and intuitive manner, which makes it possible to control a functionality of the microphones arranged in the vicinity of the virtual monitoring position 202 without, for example, individually transferring the multiplicity of microphones within an orchestra must be listened to these each assigned channels of a mixer.

According to the embodiments in which the first geometry information and the second geometry information comprise as at least one information the first distance d ₁ between the virtual listening position 202 and the first position and d ₂ between the virtual listening position 202 and the second position, the sound signal is generated 120 generates, among other things, a weighted sum of the first source signal 210 and the second source signal 212.

Although in FIG. 1 For the sake of simplicity and for better understanding only two microphones 204 and 206 are shown, it goes without saying that according to further embodiments of the present invention of a Tonsignalalerer 100 any number of microphones of in FIG. 1 schematically illustrated type can be used to generate a sound signal for a virtual listening position, as will be explained here and with reference to the following embodiments.

That is, according to some embodiments, the audio signal x is generated from a linear combination of the first source signal 210 (x ₁ ) and the second source signal 212 (x ₂ ), the first source signal x ₁ having a first weight g ₁ and the second source signal x ₂ weighted with a second weight g ₂ , so that: $$\mathrm{x}={\mathrm{<figure-callout\; id="1"\; label="G"\; filenames="imgf0003.png,imgf0007.png"\; state="\{\{state\}\}">G</figure-callout>}}_{\mathrm{1}}*{\mathrm{x}}_{\mathrm{1}}+{\mathrm{<figure-callout\; id="2"\; label="G"\; filenames="imgf0007.png"\; state="\{\{state\}\}">G</figure-callout>}}_{\mathrm{2}}*{\mathrm{x}}_{\mathrm{2}}\mathrm{,}$$

According to further embodiments, as already mentioned, further source signals x ₃ ,..., X _n with associated weights g ₃ ,..., G _{n may} additionally be taken into account. Of course, sound signals are time-dependent, in the present case for reasons of clarity In some cases, the explicit reference to the time dependence is dispensed with and indications of sound or source signals x are to be understood synonymously with the specification x (t).

FIG. 2 schematically shows the space 200, wherein the in FIG. 2 is assumed to be limited by rectangular walls, which are responsible for the emergence of a diffuse sound field. Furthermore, it is assumed for the sake of simplification that, although in the FIG. 2 source 208 may be located within the confined area one or more sources of sound, they can be considered in terms of their effect for the individual microphones initially simplified as a single source. The direct sound emitted by these sound sources is multiply reflected by the walls delimiting the space 200, so that a diffuse sound field generated by the multiple reflections of the already attenuated signals results from uncorrelated superimposed signals having a constant volume at least approximately within the entire space , This superimposed is a direct sound component, ie the one sound directly from the sound sources located within the source 208 reaches the possible listening positions, in particular so the microphones 220 to 232, without having been previously reflected. That is, within the space 200, the sound field can be conceptually idealized into two components, namely a direct sound component, which directly reaches the corresponding listening position from the place of generation of the sound and a diffuse sound component, which consists of an approximately uncorrelated superimposition of a plurality of directly radiated and reflected signals.

At the in FIG. 2 Because of the proximity of the microphones 220 to 224 to the source 208, the illustration shown may be predominantly direct sound, that is, the volume or sound pressure of the signal picked up by these microphones is predominantly due to a direct portion of sound within the source 208 arranged sound sources ago. By contrast, it may be assumed, for example, that the microphones 226-232 recorded a signal predominantly due to the diffuse sound component, since the spatial distance between the source 208 and the microphones 226-232 is large, so that the volume of the direct sound is at least comparable at these positions or less than the volume of the diffuse sound field.

In order to take into account the reduction of the volume as the distance increases in generating the sound signal for the virtual listening position 202, according to some embodiments of the invention, a weight g _n for the individual source signals depends on the distance between the virtual listening position 202 and the microphones 220 to 232 selected for recording the source signals. FIG. 3 shows an example of a way to determine such a weight or such a factor for multiplication with the source signal, in which case microphone 222 was chosen as an example. As FIG. 3 schematically illustrates, according to some embodiments, the weight g _n is chosen to be proportional to the inverse of a power of the first distance d ₁ , ie: $${\mathrm{<figure-callout\; id="1"\; label="G"\; filenames="imgf0003.png,imgf0007.png"\; state="\{\{state\}\}">G</figure-callout>}}_{\mathrm{1}}\alpha \frac{\mathrm{1}}{{{\mathrm{d}}_{\mathrm{1}}}^{\mathrm{n}}}\mathrm{,}$$

According to some embodiments, n = 1 is chosen as the power, that is, the weight or the weight factor is inversely proportional to the distance d ₁ , a dependency which corresponds approximately to the free-field propagation of a punctiform uniformly emitting sound source. That is, in some embodiments, it is assumed that the volume is inversely proportional to the distance 240. According to some further embodiments, a so-called near field radius 242 (r ₁ ) is additionally considered for some or all of the microphones 220 to 232. The near field radius 242 corresponds to an area immediately around a sound source, in particular to the area within which the sound wave or the sound front is formed. Within the near field radius, the sound pressure level or the volume of the audio signal is assumed to be constant. In a simple model presentation, it may be assumed that no significant attenuation occurs in the medium within a single wavelength of an audio or audio signal, so that the sound pressure is constant at least within a single wavelength (corresponding to the near field radius). It follows that the near field radius can also be frequency-dependent.

By analogous use of the near field radius according to some embodiments of the invention, a sound signal at the virtual listening position 202 can be generated by weighting the variables relevant for the control of the acoustic scene or the configuration and wiring of the individual microphones particularly clearly when the virtual Listening position 202 one of the real positions of the microphones 220 to 232 approaches. Although, according to some embodiments of the present invention, a frequency independent quantity is assumed for the near field radius r, according to some Further embodiments, a frequency dependence of the near field radius be implemented. According to some embodiments, it is thus assumed for generation of the sound signal that within a near field radius r around one of the microphones 220 to 232 the volume is constant. According to some further embodiments, to simplify the calculation of the signal and possibly still take into account the influence of a near field radius, it is assumed as general calculation rule that the weight g ₁ is proportional to a quotient of the near field radius r _{1 of} the considered microphone 222 and the distance d ₁ of virtual listening position 202 and microphone 222, so that: $${\mathrm{<figure-callout\; id="1"\; label="G"\; filenames="imgf0003.png,imgf0007.png"\; state="\{\{state\}\}">G</figure-callout>}}_{\mathrm{1}}\alpha \frac{{\mathrm{r}}_{\mathrm{1}}}{{{\mathrm{d}}_{\mathrm{1}}}^{\mathrm{n}}}\mathrm{,}$$

Such a parameterization or distance dependence can take into account both the considerations for the near field and the considerations for the far field. As already mentioned above, the far field of a point-shaped sound source is followed by a far field, in which the sound pressure is halved with each doubling of the distance from the sound source when the field propagates freely, ie the level decreases by 6 dB in each case. This property is also known as the law of distance or 1 / r law. Although, according to some embodiments of the invention, sources 208 may be recorded whose sound sources radiate directionally, it may possibly be assumed that point-shaped sound sources, if not a realistic reproduction of the sound field at the location of the virtual listening position 202 in the foreground, but rather the possibility Microphones or the recording quality of a complex acoustic scene quickly and efficiently to control or listen to.

As already in FIG. 2 indicated, the Nahfeldradii for different microphones can be chosen differently according to some embodiments. In particular, the different microphone types can be taken into account. As a type of microphone is to be understood herein information that, detached from the actual structure of the individual microphone describes a property of the microphone or its use, which differs from an identical property or use of another microphone, which is also for receiving the source 208 is used. An example of such a distinction is the distinction between microphones of a first type (type "D" in FIG FIG. 2 ), which due to their geometric positioning predominantly record direct sound components and such microphones, due to the larger Distance or another relative position with respect to the source 208 predominantly record or record the diffuse sound field (microphones of the type "A" in FIG FIG. 2 ). In particular, in such a division of the microphones in different types of microphone, the use of different Nahfeldradii may be useful. In this case, according to some embodiments, the near-field radius of type A microphones is chosen to be greater than that for type D microphones, which may result in a simple way of controlling the individual microphones when the virtual listening position 202 is placed near them, without the physical Roughly distort conditions or the sound impression, especially since the diffuse sound field, as shown above, is approximately the same over large areas.

Generally speaking, tone signal generators 100, in accordance with some embodiments of the present invention, for combining the source signals use different combination rules when the microphones that record the respective source signals are associated with different microphone types. That is, a first combination rule is used when the two microphones to be combined are associated with a first microphone type and a second combination rule is used when the two microphones to be combined or the source signals recorded by these microphones are assigned to a second, different microphone type ,

In particular, according to some embodiments, the microphones of each different type can first be processed completely separated and combined into a sub-signal X _virt , whereupon in a final step by the sound signal generator or a mixer used the final signal is generated by combining the previously generated sub-signals. Applied to the in FIG. 2 For example, this means that first a partial signal x _A can be determined for the virtual listening position 202, which only takes into account type A microphones 226 to 232. At the same time or before or after, a second partial signal x _{D could be determined} for the virtual listening position 202, which takes into account only the microphones of the type D, ie the microphones 220 to 224, but combines them according to another combination rule. In a final step, the final sound signal x for the virtual listening position 202 could then be generated by combining these two sub-signals, in particular by a linear combination of the first sub-signal x _D , which was obtained by means of the microphones of the first type (D) and a second partial signal x _A , which was obtained by means of the microphones of the second type (A), so that: $$\mathrm{x}={\mathrm{x}}_{\mathrm{A}}+{\mathrm{x}}_{\mathrm{D}\mathrm{,}}$$

FIG. 4 shows one of the FIG. 2 similar schematic view of an acoustic scene together with positions of microphones 220 to 224, the direct sound record and a series of microphones of the type A, of which in particular the microphones 250 to 256 are to be considered below. On the basis of this some possibilities are discussed with which combination rules a sound signal for the virtual listening position 202, which in the in the FIGS. 4 and 5 represented configuration within a triangular area spanned by the microphones 250 to 254 can be generated.

Generally speaking, the interpolation of the volume or the generation of the sound signal for the virtual listening position 202 can take place taking into account the positions of the nearest microphones or taking into account the positions of all microphones. For example, it can be advantageous, inter alia for reducing the computational load, to use only the closest microphones for generating the audio signal at the virtual listening position 202. These can be found, for example, by means of a Delaunay triangulation or determined by any other algorithms for searching the nearest neighbors (nearest neighbor). Some specific possibilities for determining the volume adjustment or, generally speaking, for combining the source signals associated with the microphones 250 to 254 will be discussed in more detail below with reference to FIG FIG. 5 , described.

If the virtual listening position 202 were not within one of the triangulation triangles, but outside, for example, at the in FIG. 4 dashed drawn further virtual listening position 260 would be available for interpolation of the signal or for combining a sound signal from the source signals of the microphones only two source signals of the nearest neighbors. For the sake of simplicity, the possibility of combining two source signals will also be described below with reference to FIG FIG. 5 discussed, wherein the source signal of the microphone 250 is initially neglected in the interpolation of two source signals.

According to some embodiments of the invention, the audio signal for the virtual listening position 202 is generated according to a first fading rule, the so-called linear panning law. According to this method, the tone signal x _virt1 is determined using the following calculation rule: $${\mathrm{<figure-callout\; id="1"\; label="x\; virt"\; filenames="imgf0003.png,imgf0007.png"\; state="\{\{state\}\}">x\; </figure-callout>}}_{\mathrm{virt}}={\mathrm{<figure-callout\; id="1"\; label="G"\; filenames="imgf0003.png,imgf0007.png"\; state="\{\{state\}\}">G</figure-callout>}}_{\mathrm{1}}*{\mathrm{x}}_{\mathrm{1}}+\left(\mathrm{1}-{\mathrm{<figure-callout\; id="1"\; label="G"\; filenames="imgf0003.png,imgf0007.png"\; state="\{\{state\}\}">G</figure-callout>}}_{\mathrm{1}}\right)*{\mathrm{x}}_{\mathrm{2}}.{\mathrm{where\; <figure-callout\; id="2"\; label="g"\; filenames="imgf0007.png"\; state="\{\{state\}\}">g</figure-callout>}}_{\mathrm{2}}=\left(\mathrm{1}-{\mathrm{<figure-callout\; id="1"\; label="G"\; filenames="imgf0003.png,imgf0007.png"\; state="\{\{state\}\}">G</figure-callout>}}_{\mathrm{1}}\right)\mathrm{,}$$

That is to say, the weights of the individual source signals x ₁ and x _{2 to be} added add up to 1 linearly and the tone signal x _virt1 is formed either by one of the two signals x ₁ or x ₂ alone or a linear combination of the two. Because of this linear relationship, the sound signals thus generated for any values of g ₁ at identical source signals at a constant volume, whereas completely different (decorrelated) source signals x ₁ and x ₂ lead to a sound signal for the value g ₁ = 0.5 a drop in volume of minus 3dB, ie by a factor of 0.5.

A second fade _rule according to which the tone signal x _virt2 can be generated is the so-called sine and cosine law: $${\mathrm{<figure-callout\; id="2"\; label="x\; virt"\; filenames="imgf0007.png"\; state="\{\{state\}\}">x\; </figure-callout>}}_{\mathrm{virt}}=\cos \left(\mathrm{\delta }\right)*{\mathrm{x}}_{\mathrm{1}}+\sin \left(\mathrm{\delta }\right)*{\mathrm{x}}_{\mathrm{2}}.\mathrm{where\; \delta }\in \left[\mathrm{0}\mathrm{°}.\mathrm{90}\mathrm{°}\right]\mathrm{,}$$

Eine vierte Überblendregel, die verwendet werden kann, um das Tonsignal x_virt4 zu erzeugen, ist das sogenannte Sinus- Gesetz: $${\mathrm{x}}_{\mathrm{virt}4}={\mathrm{g}}_{\mathrm{1}}*{\mathrm{x}}_{\mathrm{1}}+{\mathrm{g}}_{\mathrm{2}}*{\mathrm{x}}_{\mathrm{2}},\mathrm{wobei}\frac{\sin \mathit{\theta }}{{\sin \mathit{\theta }}_{\mathrm{0}}}=\frac{{\mathrm{g}}_{\mathrm{1}}-{\mathrm{g}}_{\mathrm{2}}}{{\mathrm{g}}_{\mathrm{1}}+{\mathrm{g}}_{\mathrm{2}}}\mathrm{und}\mathit{\theta }\in \left[\mathrm{0}\mathrm{°};\mathrm{90}\mathrm{°}\right]\mathrm{.}$$

The parameter δ which determines the individual weights g ₁ and g ₂ ranges from 0 ° to 90 ° and is calculated from the distance between the virtual listening position 202 and the microphones 252 and 254. Here, the squares of the weights for arbitrary values of δ to 1, a constant-tone sound signal can be generated by means of the sine-cosine law for any parameter δ if the source signals are decorrelated. However, with identical source signals for the parameter δ = 45 °, a volume increase of 3dB results.
A third fade rule, which leads to the results similar to the second fade _rule , and according to which the tone signal x _virt3 can be generated, is the so-called tangent law: $${\mathrm{<figure-callout\; id="3"\; label="x\; virt"\; filenames="imgf0007.png"\; state="\{\{state\}\}">x\; </figure-callout>}}_{\mathrm{virt}}={\mathrm{<figure-callout\; id="1"\; label="G"\; filenames="imgf0003.png,imgf0007.png"\; state="\{\{state\}\}">G</figure-callout>}}_{\mathrm{1}}*{\mathrm{x}}_{\mathrm{1}}+{\mathrm{<figure-callout\; id="2"\; label="G"\; filenames="imgf0007.png"\; state="\{\{state\}\}">G</figure-callout>}}_{\mathrm{2}}*{\mathrm{x}}_{\mathrm{2}}.\mathrm{in\; which}\frac{\tan \mathit{\theta }}{{\mathrm{<figure-callout\; id="0"\; label="tan\; \theta "\; filenames="imgf0005.png"\; state="\{\{state\}\}">tan\; </figure-callout>}\mathit{\theta }\mathit{}}_{\mathrm{0}}}=\frac{{\mathrm{G}}_{\mathrm{1}}-{\mathrm{G}}_{\mathrm{2}}}{{\mathrm{<figure-callout\; id="1"\; label="G"\; filenames="imgf0003.png,imgf0007.png"\; state="\{\{state\}\}">G</figure-callout>}}_{\mathrm{1}}+{\mathrm{<figure-callout\; id="2"\; label="G"\; filenames="imgf0007.png"\; state="\{\{state\}\}">G</figure-callout>}}_{\mathrm{2}}}\mathrm{and}\mathit{\theta }\in \left[\mathrm{0}\mathrm{°};\mathrm{90}\mathrm{°}\right]\mathrm{,}$$

A fourth fading _rule that can be used to generate the tone signal x _virt4 is the so-called sine law: $${\mathrm{<figure-callout\; id="4"\; label="x\; virt"\; filenames="imgf0007.png"\; state="\{\{state\}\}">x\; </figure-callout>}}_{\mathrm{virt}}={\mathrm{<figure-callout\; id="1"\; label="G"\; filenames="imgf0003.png,imgf0007.png"\; state="\{\{state\}\}">G</figure-callout>}}_{\mathrm{1}}*{\mathrm{x}}_{\mathrm{1}}+{\mathrm{<figure-callout\; id="2"\; label="G"\; filenames="imgf0007.png"\; state="\{\{state\}\}">G</figure-callout>}}_{\mathrm{2}}*{\mathrm{x}}_{\mathrm{2}}.\mathrm{in\; which}\frac{\sin \mathit{\theta }}{{\mathrm{<figure-callout\; id="0"\; label="sin\; \theta "\; filenames="imgf0005.png"\; state="\{\{state\}\}">sin\; </figure-callout>}\mathit{\theta }\mathit{}}_{\mathrm{0}}}=\frac{{\mathrm{G}}_{\mathrm{1}}-{\mathrm{G}}_{\mathrm{2}}}{{\mathrm{<figure-callout\; id="1"\; label="G"\; filenames="imgf0003.png,imgf0007.png"\; state="\{\{state\}\}">G</figure-callout>}}_{\mathrm{1}}+{\mathrm{<figure-callout\; id="2"\; label="G"\; filenames="imgf0007.png"\; state="\{\{state\}\}">G</figure-callout>}}_{\mathrm{2}}}\mathrm{and}\mathit{\theta }\in \left[\mathrm{0}\mathrm{°};\mathrm{90}\mathrm{°}\right]\mathrm{,}$$

Again, the squares of the weights add up to any one of the possible values of the parameter θ . The parameter θ is again determined by the distances between the virtual listening position 202 and the microphones, and can be from minus 45 degrees to 45 degrees.

In particular, for the combination of two source signals, over which there is limited a-priori knowledge, as may be the case, for example, in a spatially slightly varying diffuse sound field, a fourth combination rule may be used, according to which the first previously described fade rule and combining the second previously described fading rule depending on the source signals to be combined. In particular, according to the fourth combination rule, there is a linear combination of two intermediate _signals x _virt1 and x _virt2 , which were initially generated separately for the source signals x ₁ and x ₂ according to the first and the second crossfading rule. In particular, according to some embodiments of the present invention, the weighting _factor for the linear combination is the correlation coefficient σ _x1x2 between the source signals x ₁ and x ₂ , which is defined as follows and represents a measure of the similarity of the two signals: $${\sigma }_{{\mathrm{x}}_1{\mathrm{x}}_2}=\frac{\mathbf{e}\left\{\left(x_1-\mathbf{<figure-callout\; id="1"\; label="e\; x"\; filenames="imgf0003.png,imgf0007.png"\; state="\{\{state\}\}">e\; </figure-callout>}\left\{x_{}\right\}\left\{{}_1\right\}\right)*\left(x_2-\mathbf{<figure-callout\; id="2"\; label="e\; x"\; filenames="imgf0007.png"\; state="\{\{state\}\}">e\; </figure-callout>}\left\{x_{}\right\}\left\{{}_2\right\}\right)\right\}}{{\sigma }_{{\mathrm{x}}_1}{\sigma }_{{\mathrm{x}}_2}}\approx \frac{\mathbf{e}\left({\mathrm{x}}_1*{\mathrm{x}}_2\right)}{{\sigma }_{{\mathrm{x}}_1}{\sigma }_{{\mathrm{x}}_2}}\mathrm{,}$$

Where E denotes the expected value or the linear mean value and σ indicates the standard deviation of the relevant variable or the respective source signal, whereby for acoustic signals to a good approximation the linear mean value E {x} is zero. $${\mathrm{x}}_{\mathrm{virt}}={\sigma }_{x1x2}*{\mathrm{<figure-callout\; id="1"\; label="x\; virt"\; filenames="imgf0003.png,imgf0007.png"\; state="\{\{state\}\}">x\; </figure-callout>}}_{\mathrm{virt}}+\left(1-{\sigma }_{x1x2}\right)*{\mathrm{<figure-callout\; id="2"\; label="x\; virt"\; filenames="imgf0007.png"\; state="\{\{state\}\}">x\; </figure-callout>}}_{\mathrm{virt}}\mathrm{,}$$

That is, according to some embodiments of the present invention, the combination _rule further comprises forming a weighted sum x _virt from the intermediate signals x _virt1 and x _virt2 weighted with a correlation coefficient σ _x1x2 for a correlation between the first source signal x ₁ and the second source signal x ₂ .

By using the fourth combination rule, a combination with approximately constant volume can thus be achieved over the entire parameter range according to some embodiments of the present invention. This can also be overwhelming regardless of whether the signals to be combined are dissimilar or similar.

Inasmuch as according to some embodiments of the present invention, a sound signal is to be acquired at a virtual listening position 202 that is within a triangle bounded by three microphones 250 to 254, in accordance with some embodiments of the present invention, the three source signals of the microphones 250 to 254 may be linearly combined in which the individual signal components of the source signals associated with the microphones 250 to 254 are based on a vertical projection of the virtual listening position 202 on the height of the triangle which is assigned to the position of the microphone assigned to the respective source signal.

If, for example, the signal component of the microphone 250 or the weight associated with this source signal are to be determined, a vertical projection of the virtual listening position 202 to the height 262 is initially made, which is assigned to the microphone 250 or the corner of the triangle at which the microphone 250 is located. This results in the FIG. 5 This in turn divides the height 262 into a first height portion 266 facing the microphone 250 and a height portion 268 facing away from the microphone. The ratio of these height portions 266 and 268 is used to weight according to one of the above fade rules for the source signal of the microphone 250, assuming that at the opposite end of the height 262 from the microphone 250 is a sound source or microphone which constantly records a signal of zero amplitude.

That is, according to the embodiments of the invention, the height of each triangle side is determined and the distance of the virtual microphone to each triangle side is determined. Along the corresponding height, the microphone signal is blinded linearly or, depending on the selected fade rule, from the vertex of the triangle to the opposite triangle side to zero. For the in FIG. 5 As shown, this means that the source signal of the microphone 250 is used with the weight 1 when the projection 264 is at the position of the microphone 250 and zero when it is on the connecting line between the position of the microphones 252 and 254, ie located on the opposite side of the triangle. Between these two extremes becomes the source signal of the microphone 250 faded in or out. In general terms, this means that when combining the signal from three signals, three source signals x ₁ to x _{3 are} taken into account whose associated microphones 250 to 254 span a triangular area within which the virtual listening position 202 is located. In this case, the weights g ₁ to g ₃ for the linear combination of the source signals x ₁ to x ₃ are determined based on a vertical projection of the virtual listening position 202 on the height of the triangle which is assigned to the position of the microphone assigned to the respective source signal or through the this height runs.

In so far for the determination of the signal, the above-discussed fourth fade control is used, a common correlation coefficient for the three source signals x ₁ can be determined to x _3, characterized in that first a correlation between the respective adjacent source signals is determined, resulting in a total of three correlation coefficient values. From the three correlation coefficients thus obtained, a common correlation coefficient is formed by averaging, which in turn determines the weighting for the sum of sub-signals formed by the first fading rule (linear panning) and the second fading rule (sine-cosine law). That is, it is first determined a first sub-signal with the sine-cosine law, then a second sub-signal is determined with the linear panning and the two sub-signals are linearly combined by weighting with the correlation coefficient.

FIG. 6 FIG. 11 is an illustration of another possible configuration of positions of microphones 270 through 278 within which a virtual listening position 202 is located. In particular, based on FIG. 6 another possible combination rule is illustrated, the properties of which can be arbitrarily combined with the above-described possible combinations or which, taken alone, can also be a combination rule in the sense described herein.

According to some embodiments of the invention, as in FIG. 6 schematically considered, a source signal only in combination with the sound signal for a virtual listening position 202 taken into account when the microphone associated with the source signal is within a predetermined configurable distance R from the virtual listening position 202. As a result, computation time can possibly be saved, for example, by taking into account only those microphones, in accordance with some embodiments whose signal contributions lie above the human hearing threshold according to the selected combination rules.

According to some embodiments of the invention, as shown in FIG FIG. 7 schematically illustrated, the combination rule further take into account a directional characteristic for the virtual listening position 202. That is to say, for example, the first weight g ₁ for the first source signal x _{1 of} the first microphone 220 may additionally be proportional to a directional factor rf ₁ , which results from a sensitivity function or a directional characteristic for the virtual listening position 202 and from the relative position between virtual Listening position 202 and microphone 220. D. h. According to these embodiments, the first geometry information further comprises first directional information about a direction between the microphone 220 and a preferred direction 280 associated with the virtual listening position 202, in which the directional characteristic 282 has its maximum sensitivity.

Generally speaking, the weighting factors g ₁ and g _{2 of} the linear combination of the source signals x ₁ and x _{2 are} additionally dependent on a first directional factor rf ₁ and a second directional factor rf ₂ , which take into account the directional characteristic 280 at the virtual listening position 202 according to some embodiments.

In other words, the combination rules discussed in the preceding paragraphs can be summarized as follows. The individual implementations are described in more detail in the next sections. All variants have in common that comb filter effects could occur when adding the signals. If this is potentially the case, the signals can be delayed accordingly beforehand. Therefore, the algorithm usable for delay is first shown.

For microphones that are more than two meters apart, signals can be summed up without producing noticeable comb filter effects. It is also safe to sum up signals from microphones whose position intervals comply with the so-called 3: 1 rule. The rule states that when recording a sound source with two microphones, the distance between the sound source and the second microphone should be at least three times the distance from the sound source to the first microphone in order to obtain any noticeable comb filter effects. requirement are microphones of the same sensitivity and the drop of the sound pressure level with the distance, for example according to the 1 / r-law.

Gemäß einigen weiteren Ausführungsbeispielen werden alle Quellsignale mit der maximalen bestimmten Latenz beaufschlagt.The system or a tone signal generator or its geometry processor initially determines whether both conditions are met. If this is not the case, the signals can be delayed prior to the calculation of the virtual microphone signal according to the current position of the virtual microphone. If necessary, the distances of all microphones to the virtual microphone are determined and the signals with respect to the microphone, which is furthest from the virtual one, are shifted in time. For this purpose the largest distance is determined and the difference to the remaining distances is formed. The latency Δt _i in samples now results from the ratio of the respective distance d _i to the sound velocity c multiplied by the sampling rate Fs. The calculated value can be rounded in digital implementations, for example, if the signal is only to be delayed by whole samples. N here are the number of recording microphones: $${\mathrm{\Delta }\mathit{t}}_i=\mathit{round}\left(\frac{d_i}{c}*\mathit{fs}\right)\mathrm{with\; i}=1.....\mathrm{N}\mathrm{,}$$

According to some other embodiments, all the source signals are applied with the maximum specific latency.

To calculate the virtual microphone signal, the following variants can be implemented. In this case, close microphones or microphones for recording direct sound are referred to below as microphones of a first microphone type and Ambientmikrofone or microphones for recording a diffuse sound component as microphones of a second microphone type. Furthermore, the virtual listening position is also referred to as a position of a virtual microphone.

Auf diese Weise erhält man das aufgrund der räumlichen Entfernung d_i gedämpfte Mikrofonsignal x_i,gedämpft. Alle so berechneten Signale werden aufaddiert und bilden gemeinsam das Signal für das virtuelle Mikrofon: $$x_{\mathit{virtMic}}\left(t\right)=\sum _{i=1}^N{x}_{i,\mathit{ged}\ddot{a}\mathit{mpft}}\left(t\right)\mathrm{.}$$

Gemäß einer zweiten Variante erfolgt eine Trennung des Direkt- und Diffusschalls. Das Diffusschallfeld soll dabei im gesamten Raum annähernd gleich laut sein. Hierfür wird der Raum durch die Anordnung der Ambientmikrofone in bestimmte Bereiche gegliedert. Je nach Bereich berechnet sich der diffuse Schallanteil aus einem, zwei oder drei Mikrofonsignalen. Die Signale der Nahmikrofone fallen mit der Entfernung nach dem Abstandsgesetz ab.According to a first variant, both the signals of the close microphones or of the microphones of a first microphone type and the signals of the ambient microphones fall off according to the law of distance. As a result, each microphone can be heard in his position particularly dominant. For the calculation of the virtual microphone signal, the near field radii around the near and ambient microphones can first be determined by the user. Within this radius, the volume of the signals remains constant. If you now place the virtual microphone in the recording scene, the distances from the virtual microphone to each individual real microphone are calculated. For this, the sample values of the microphone signals x _i [t] is divided by the instantaneous distance d _i and multiplied by the near field radius r _nah . N indicates the number of recording microphones: $$x_{i.\mathit{ged}\ddot{a}\mathit{fumes}}\left(t\right)=\frac{x_i\left[t\right]}{d_i}*r_{\mathit{close}}\mathit{with\; i}=1.....N\mathrm{,}$$

In this manner, the attenuated microphone signal x _{i, attenuated} due to the spatial distance d _i. All signals calculated in this way are added together and together form the signal for the virtual microphone: $$x_{\mathit{virtMic}}\left(t\right)=\underset{i=1}{\overset{N}{\Sigma }}{x}_{i.\mathit{ged}\ddot{a}\mathit{fumes}}\left(t\right)\mathrm{,}$$

According to a second variant, a separation of the direct and diffuse sound. The diffuse sound field should be approximately equal in volume throughout the room. For this purpose, the room is arranged by the arrangement of Ambientmikrofone in certain areas. Depending on the area, the diffuse sound component is calculated from one, two or three microphone signals. The signals of the Nahmikrofone fall off with the distance according to the distance law.

Figure 4 shows an example of a room layout. The dots symbolize the ambient microphones. The outer ambient microphones form a polygon. The area within this polygon is divided into triangles. For this the Delaunay triangulation is used. With this method, a triangle mesh can be formed from a set of points. Above all, it is distinguished by the fact that the circumference of a triangle does not include any further points of the set. By fulfilling this so-called perimeter condition, triangles are created with the largest possible interior angles. In Figure 4 this triangulation is represented by four points.

Delaunay triangulation groups closely spaced microphones and maps each microphone to the surrounding space. The signal for the virtual microphone is calculated within the polygon of three microphone signals. Outside the polygon, two vertical lines are defined for each connecting line of two vertices, which run through the corner points. This limits certain areas outside the polygon as well. The virtual microphone can thus be located either between two microphones or with a microphone at a corner point. To calculate the diffuse sound component, it should first be determined whether the virtual microphone is inside or outside the polygon forming the edge. ever by position, the diffused portion of the virtual microphone signal is calculated from one, two or three microphone signals.

Im Bereich zwischen zwei Mikrofonen setzt sich das virtuelle Mikrofonsignal aus den beiden entsprechenden Mikrofonsignalen x₁ und x₂ zusammen. Je nach Position wird zwischen den beiden Signalen mit Hilfe verschiedener Überblendregeln bzw. Panning-Verfahren überblendet. Diese werden nachfolgend auch wie folgt bezeichnet: lineares Panning-Gesetz (erste Überblendregel), Sinus-Cosinus-Gesetz (zweite Überblendregel), Tangens-Gesetz (dritte Überblendregel) und Kombination aus linearem Panning-Gesetz und Sinus-Cosinus-Gesetz (vierte Überblendregel).If the virtual microphone is outside the polygon, a distinction is made between the areas at one vertex and between two microphones. If the virtual microphone in the area near a microphone is at a vertex of the polygon, only the x _i signal from that microphone will be used to calculate the diffused sound portion: $$x_{\mathit{diffuse}}\left[t\right]=x_i\left[t\right]\mathrm{,}$$

In the area between two microphones the virtual microphone signal is composed of the two respective microphone signals x ₁ and _x2. Depending on the position, the two signals are superimposed using different fade rules or panning methods. These are also referred to as follows: linear panning law (first fade rule), sine cosine law (second fade rule), tangent law (third fade rule) and combination of linear panning law and sine cosine law (fourth fade rule ).

For the combination of the two panning methods linear (x _virt1 ) and sine-cosine law (x _virt2 ) the correlation coefficient σ _{x1x2 of} the two signals x ₁ and x _{2 is} determined: $${\mathrm{\sigma }}_{{\mathrm{x}}_{\mathrm{1}}{\mathrm{x}}_{\mathrm{2}}}=\frac{\mathbf{e}\left\{\left({\mathrm{x}}_{\mathrm{1}}-\mathbf{<figure-callout\; id="1"\; label="e\; x"\; filenames="imgf0003.png,imgf0007.png"\; state="\{\{state\}\}">e\; </figure-callout>}\left\{{\mathrm{x}}_{}\right\}\left\{{\mathrm{}}_{\mathrm{1}}\right\}\right)*\left({\mathrm{x}}_{\mathrm{2}}-\mathbf{<figure-callout\; id="2"\; label="e\; x"\; filenames="imgf0007.png"\; state="\{\{state\}\}">e\; </figure-callout>}\left\{{\mathrm{x}}_{}\right\}\left\{{\mathrm{}}_{\mathrm{2}}\right\}\right)\right\}}{{\mathrm{\sigma }}_{{\mathrm{x}}_{\mathrm{1}}}{\mathrm{\sigma }}_{{\mathrm{x}}_{\mathrm{2}}}}\approx \frac{\mathbf{e}\left\{{\mathrm{x}}_{\mathrm{1}}*{\mathrm{x}}_{\mathrm{2}}\right\}}{{\mathrm{\sigma }}_{{\mathrm{x}}_{\mathrm{1}}}{\mathrm{\sigma }}_{{\mathrm{x}}_{\mathrm{2}}}}$$

wobei $${\mathrm{x}}_{\mathrm{virt}\mathrm{1}}={\mathrm{g}}_{\mathrm{1}}*{\mathrm{x}}_{\mathrm{1}}+\left(\mathrm{1}-{\mathrm{g}}_{\mathrm{1}}\right)*{\mathrm{x}}_{\mathrm{2}},$$

wobei g₂ = (1- g₁); "lineares panning" $${\mathrm{x}}_{\mathrm{virt}\mathrm{2}}=\cos \left(\mathrm{\delta }\right)*{\mathrm{x}}_{\mathrm{1}}+\sin \left(\mathrm{\delta }\right)*{\mathrm{x}}_{\mathrm{2}}$$

wobei δ E [0°; 90°]; "Sinus-Cosinus-Gesetz".Depending on the size of the coefficient σ _x1x2 , the respective law flows into the calculation of the weighted sum x _virt : $${\mathrm{x}}_{\mathrm{virt}}={\sigma }_{x1x2}*{\mathrm{<figure-callout\; id="1"\; label="x\; virt"\; filenames="imgf0003.png,imgf0007.png"\; state="\{\{state\}\}">x\; </figure-callout>}}_{\mathrm{virt}}+\left(1-{\sigma }_{x1x2}\right)*{\mathrm{<figure-callout\; id="2"\; label="x\; virt"\; filenames="imgf0007.png"\; state="\{\{state\}\}">x\; </figure-callout>}}_{\mathrm{virt}}.$$

in which $${\mathrm{<figure-callout\; id="1"\; label="x\; virt"\; filenames="imgf0003.png,imgf0007.png"\; state="\{\{state\}\}">x\; </figure-callout>}}_{\mathrm{virt}}={\mathrm{<figure-callout\; id="1"\; label="G"\; filenames="imgf0003.png,imgf0007.png"\; state="\{\{state\}\}">G</figure-callout>}}_{\mathrm{1}}*{\mathrm{x}}_{\mathrm{1}}+\left(\mathrm{1}-{\mathrm{<figure-callout\; id="1"\; label="G"\; filenames="imgf0003.png,imgf0007.png"\; state="\{\{state\}\}">G</figure-callout>}}_{\mathrm{1}}\right)*{\mathrm{x}}_{\mathrm{2}}.$$

where g ₂ = (1- g ₁ ); "linear panning" $${\mathrm{<figure-callout\; id="2"\; label="x\; virt"\; filenames="imgf0007.png"\; state="\{\{state\}\}">x\; </figure-callout>}}_{\mathrm{virt}}=\cos \left(\mathrm{\delta }\right)*{\mathrm{x}}_{\mathrm{1}}+\sin \left(\mathrm{\delta }\right)*{\mathrm{x}}_{\mathrm{2}}$$

where δ E [0 °; 90 °]; "Sine-cosine law."

If the correlation coefficient σ _x1x2 equals 1, these are identical signals and only crossfades linearly. With a correlation coefficient of 0, only the sine-cosine law is used.

In some implementations, the correlation coefficient may not only describe an instantaneous value, but may be integrated over a period of time. At the correlation meter For example, this period can be 0.5 s. Since the exemplary embodiments of the invention or the virtual microphones do not always have to be real-time capable systems, the correlation coefficient can also be determined over a longer period of time, for example 30 s

In the area within the polygon or the virtual listening position is located within triangles whose vertices were defined by means of Delaunay triangulation, such as FIG. 5 was clarified. In each triangle, the diffused sound component of the virtual microphone signal is composed of the three source signals of the microphones located at the corners. For this purpose, the height h of each triangle side is determined and the distance d _v i _{rtMic of} the virtual microphone to each triangle side is determined. Depending on the set panning method or depending on the fade rule used, the microphone signal is blended to zero from one vertex to the opposite side of the triangle along the corresponding height.

For this purpose, in principle, the panning methods described above can be used, which are also used for the calculation of the signal outside the polygon. The division of the distance d _virtMic by the value of the height h normalizes the distance to a length of 1 and delivers the corresponding position on the panning curve. Hereby, the value can be read on the y-axis, with which each of the three signals is multiplied according to the set panning method.

For the combination of linear panning law and the sine-cosine law, the correlation coefficient is first determined from two respective source signals. This gives three correlation coefficients, from which the mean value is subsequently formed.

This average determines the weighting of the sum of linear and sine-cosine panning law. Again, if the value equals 1, it will only be superimposed using a linear panning law. For a value equal to 0, only the sine-cosine law is used. In conclusion, all three signals add up the diffused portion of the sound.

The proportion of direct sound is superimposed on the diffuse, with the direct sound content of microphones of the type "D" and the indirect sound content of microphones of the type "A" recorded in the previously introduced sense. Finally, the diffuse and direct sound components are added together to give the signal for the virtual microphone: $$x_{\mathit{virtMic}}\left[t\right]=x_{\mathit{diffuse}}\left[t\right]+x_{\mathit{directly}}\left[t\right]\mathrm{,}$$

It is also possible to extend this variant. If desired, an arbitrarily large radius can be set around a microphone. Within this range, only the microphone located there can be heard. All other microphones will be zeroed or weighted to 0 so that the virtual microphone signal matches the signal from the selected microphone: $$x_{\mathit{virtMic}}\left[t\right]=x_{i.\mathit{sel}}\left[t\right]\mathrm{,}$$

According to the third variant, only the microphones, which are located in a certain radius around the virtual microphone, flow into the calculation of the virtual microphone signal. For this purpose, the distances of all microphones to the virtual microphone are first determined and determined from which microphones are within the circle. The signals of the microphones which are outside the circle are set to zero or get the weight 0.

Um plötzlich auftretende Lautstärkesprünge beim Übergang eines Mikrofons in oder aus dem Kreis heraus zu vermeiden, können die Signale am Rand des Kreises zusätzlich linear ein- bzw. ausgeblendet werden. Bei dieser Variante muss keine Unterscheidung in Nah- und Ambientmikrofone stattfinden.The signal values of the microphones x _i (t) within the circle are added up in equal parts, resulting in the signal for the virtual microphone. If N specifies the number of recording microphones within the circle, the following applies: $$x_{\mathit{virtMic}}\left(t\right)=\frac{1}{N}{\displaystyle \underset{i=1}{\overset{N}{\Sigma }}}{x}_i\left(t\right)\mathrm{,}$$

In order to avoid suddenly occurring volume jumps when a microphone enters or leaves the circle, the signals at the edge of the circle can additionally be shown or hidden in a linear manner. In this variant, there is no need to distinguish between near and ambient microphones.

In all variants, it may also be useful to assign an additional directional characteristic to the virtual microphone. For this purpose, the virtual microphone can be provided with a directional vector r, which initially points in the main direction of the directional characteristic (in the polar diagram). Since the directional characteristic of a microphone can be effective only for direct sound for some embodiments, then the directional characteristic affects only the signals of the Nahmikrofone. The signals of the ambient microphones are included unchanged in the calculation according to the combination rule. From the virtual microphone Vectors are formed to all close-up microphones. For each of the close-up microphones, the angle φ _{i, close} between this vector and the directional vector of the virtual microphone is calculated. In FIG. 7 this is exemplified for a microphone 220. By inserting the angle in the general microphone equation s (φ) = a + b * cos (φ), one obtains a factor s for each source signal, which corresponds to an additional sound attenuation due to the directional characteristic. Before adding all the source signals, each signal is multiplied by the corresponding factor. For example, there is the possibility between the directional characteristics sphere (a = 1, b = 0), width kidney (a = 0.71, b = 29), kidney (a = 0.5, b = 0.5), Supercardioid (a = 0.37, b = 0.63), hypercardioid (a = 0.25, b = 0.75), and eight (a = 0, b = 1). For example, the virtual microphone can be rotated with an accuracy of 1 ° or less.

FIG. 8 12 schematically shows a mixer 300 comprising a tone signal generator 100 and by means of which signals can be received by microphones 290 to 295 which are used to record an acoustic scene 208. The mixer serves to process the source signals from at least two microphones 290 to 295 and to provide a mixed-tone signal 302, which is included in the FIG. 8 selected representation is indicated only schematically.

In accordance with some embodiments of the present invention, the mixer further includes a user interface 306 configured to display a graphical representation of the positions of the plurality of microphones 290 to 295, and additionally the position of a virtual listening position 202 within the acoustic space the microphones 290 to 295 are located is arranged.

According to some embodiments, the user interface further allows each of the microphones 290 to 295 to be assigned a type of microphone, for example, a first type (1) identifying microphones for direct sound recording and a second type (2) designating microphones for recording diffuse sound components.

According to some further embodiments, the user interface is further adapted to provide a user of the mixer in a simple manner, for example by movement of a in FIG. 8 schematically illustrated cursor 310 or a computer mouse, the virtual listening position 202 intuitively and easily to move Thus, in a simple manner to allow control of the entire acoustic scene or the recording equipment.

FIG. 9 schematically shows an embodiment of a method for providing a sound signal comprising in a signal receiving step 500 receiving a first source signal x ₁ picked up by the first microphone and a second source signal x ₂ picked up by the second microphone.

During an analysis step 502, first geometry information is determined based on the first position and the virtual listening position and second geometry information based on the second position and the virtual listening position. In a combining step 505, at least the first source signal x ₁ and the second source signal x _{2 are combined} according to a combination rule using the first geometry information and the second geometry information.

FIG. 10 again schematically shows a user interface 306 for an embodiment of the invention, which differs from the in FIG. 8 slightly different. In this or in a so-called "interaction canvas" the positions of the microphones can be specified, in particular as sound sources or microphones of different types or microphones (1,2,3,4). For this purpose, the position of at least one receiver or a virtual listening position 202 can be specified (circle with cross). Each sound source may be associated with one of the mixer channels 310-316.

Although the generation of a single audio signal at a virtual listening position 202 has been discussed predominantly with reference to the preceding embodiments, it should be understood that according to other embodiments of the present invention, multiple, for example, 2, 3, 4 up to an arbitrary number of audio signals for more virtual Abhörpositionen can be generated, in each case the combination rules described above are used.

In this case, in other embodiments, for example, by using a plurality of spatially adjacent virtual Abhörpositionen different interception models, such as human hearing, are generated. By defining two virtual listening positions, the distance of the human hearing or the Pinna, for example, in conjunction with a frequency-dependent directional characteristic for each of the virtual Abhörpositionen a signal can be generated that simulates the hearing experience that a human listener would have in place between the two virtual Abhörpositionen when listening directly by means of a headphone or the like. That is, at the location of the left ear canal or the left earpiece, the first virtual Abhörposition would be generated, which also has a frequency-dependent directional characteristic, so that the signal propagation along the ear canal in the sense of a Head-Related-Transfer-Function (HRTF) on the frequency-dependent directional characteristic could be simulated. If one were to do the same for the second virtual listening position with respect to the right ear, according to some embodiments of the present invention, two mono signals would be obtained which correspond directly to the sound impression that a real listener would have at the location of the virtual listening position when listening directly to, for example, a headphone.

In a similar manner, for example, a conventional stereo microphone can also be simulated.

• Monitoring der räumlichen Klangszene, die gerade aufgezeichnet wird.
• Erstellung von teilautomatisierten Tonmischungen durch Steuerung von virtuellen Empfängern.
• Eine visuelle Darstellung der räumlichen Anordnung.

In summary, according to some embodiments of the invention, the position of a sound source (eg, a microphone) in the mixer / recording software can be specified or automatically detected. At least three new tools are available to the sound engineer based on the position of the sound source:

• Monitoring the spatial sound scene being recorded.
• Creation of semi-automated sound mixes by controlling virtual receivers.
• A visual representation of the spatial arrangement.

1. 1. Abstandsabhängige Lautstärke
2. 2. Lautstärkeinterpolation zwischen zwei oder mehreren Schallquellen
3. 3. Ein kleiner Bereich um die jeweilige Schallquelle, in der nur diese zu hören ist (der Abstandswert kann konfiguriert werden)

Solche Berechnungsvorschriften der Empfängersignale können verändert werden, beispielsweise indem:

1. 1. Ein Empfängerbereich um die Schallquelle oder den Empfänger angegeben wird,
2. 2. Eine Richtcharakteristik für den Empfänger angegeben wird.

Für jede Schallquelle kann ein Typ (z.B.: Direktschallmikrofon, Ambient- oder Diffusschallmikrofon) gewählt werden. Durch die Wahl des Typs wird die Berechnungsvorschrift des Signals am Empfänger gesteuert. FIG. 10 schematically shows a potential user interface with the positions of the sound sources and one or more "virtual receivers". Via the user interface or via an interaction canvas, each microphone (numbered 1 to 4) can be assigned a position. Each microphone is connected to a channel strip of the mixer / recording software. The positioning of one or more receivers (circle with cross) calculates sound signals from the sound sources that can be used for monitoring or finding signal errors or creating mixes. The microphones or sound sources for this purpose are different types of functions assigned, eg Nahmikrofon (type "D") or Ambientmikrofon (type "A") or even part of a microphone array, which should be evaluated only together with the others. Depending on the function, the calculation rules used are adjusted. Furthermore, the user has the opportunity to configure the calculation of the output signal. In addition, other parameters can be set, such as the type of crossfading between adjacent microphones. Variable components or procedures of the calculation can be:

1. 1. Distance-dependent volume
2. 2. Volume interpolation between two or more sound sources
3. 3. A small area around the respective sound source, in which only these can be heard (the distance value can be configured)

Such calculation rules of the receiver signals can be changed, for example by:

1. 1. a receiver area is specified around the sound source or the receiver,
2. 2. A directional characteristic is given for the receiver.

For each sound source, one type (eg direct sound microphone, ambient or diffuse sound microphone) can be selected. By selecting the type, the calculation rule of the signal at the receiver is controlled.

This leads to a particularly simple operation in the specific application. Preparing a recording with a lot of microphones is so much easier. In this case, each microphone can already be allocated a position in the mixing console in the setup process before the actual recording. The sound mixing no longer has to be done by adjusting the volume for each sound source on the channel strip, but can be done by specifying a position of the receiver in the sound source scene (eg: simple click with the mouse in the scene). Based on a selectable model for calculating the volume at the receiver site, a new signal is calculated for each repositioning of the receiver. By "shutting down" the single microphones, a noise signal can be identified very quickly. Likewise, by positioning a spatial sound mixture can be created if the receiver signal is used as the output speaker signal on. It is no longer necessary to set a volume for each individual channel; the setting is made by selecting the position of the receiver for all sound sources simultaneously. The algorithms also offer a novel creative tool.

berechnet. Die Variable x kann verschiedene Werte annehmen, in Abhängigkeit vom Typ der Schallquelle z.B. x=1; x=1/2. Befindet sich der Empfänger im Kreis mit dem Radius r₁ gilt ein festgesetzter (konstanter) Lautstärkewert. Je größer die Entfernung der Schallquelle zum Empfänger ist, desto leiser ist das Tonsignal.The scheme for the distance-dependent calculation of sound signals shows FIG. 3 , Here, depending on the radius R _L a volume value g after $$\mathrm{G}=\frac{1}{{R_L}^x}$$

calculated. The variable x can assume different values, depending on the type of sound source eg x = 1; x = 1/2. If the receiver is in a circle with the radius r ₁ , a fixed (constant) volume value applies. The greater the distance of the sound source to the receiver, the quieter the sound signal.

A scheme for volume interpolation shows FIG. 5 , In this case, the calculation of the volume arriving at the receiver is based on the position of the receiver between two or more microphones. The selection of the active sound sources can be determined by so-called "nearest-neighbor" algorithms. The calculation of an audible signal at the receiver location or at the virtual listening position is effected by an interpolation rule between two or more sound source signals. The respective volumes are adjusted dynamically to allow the listener a constantly pleasant volume.

In addition to the activation of all sound sources at the same time, using the distance-dependent volume calculation, sound sources can be activated by another algorithm. Here, an area around the receiver with the radius R is defined. The value of R can be varied by the user. If the sound source is in this range, it is audible to the receiver. This algorithm, shown in FIG. 6, can also be combined with the distance-dependent volume calculation. So there is an area with the radius R around the receiver. If sound sources are within the radius, they are audible to the receiver. If the sound sources are outside, their signal is not included in the calculation of the output signal.

To calculate the volume of the sound sources at the receiver or at the virtual listening position, it is possible to define a directional characteristic for the receiver. This indicates how strongly the sound signal of a sound source is effective direction-dependent on the receiver. The directional characteristic may be a frequency-dependent filter or a pure volume value. FIG. 7 shows this schematically. The virtual receiver is provided with a direction vector which can be rotated by the user. A selection of simple ones Geometries are presented to the user as well as a selection of directional characteristics of popular microphone types and also some examples of human ears to create a virtual listener. The receiver or the virtual microphone at the virtual listening position has, for example, a cardioid characteristic. Depending on this directional characteristic, the signals of the sound sources have a different influence on the receiver. Depending on the direction of arrival signals are attenuated differently.
In summary, some embodiments consist of a mixer for processing at least a first and a second source signal and providing a mixed audio signal, the mixer comprising a tone signal generator for providing a sound signal for a virtual listening position within a room in which an acoustic scene of at least a first microphone at a first known location within the room as the first source signal and from at least one second microphone at a second known location within the room as the second source signal, the audio signal generator comprising: an input interface adapted to be from the first source signal Microphone received first source signal and received by the second microphone second source signal to receive; a geometry processor configured to determine first geometry information based on the first position and the virtual listening position, and determine second geometry information based on the second position and the virtual listening position; and one A signal generator for providing the sound signal, wherein the signal generator is adapted to combine at least the first source signal and the second source signal according to a combination rule using the first geometry information and the second geometry information.

In some examples, the signal generator is configured to use a first combination rule when the first microphone and the second microphone are associated with a first microphone type and to use a second, different combination rule when the first microphone and the second microphone are a second microphone type assigned.

In some examples, a first near field radius r ₁ is used according to the first combination rule, and a second, different near field radius r _{2 is} used according to the second combination rule.

In some examples, the first type of microphone is associated with a microphone for receiving a direct portion of sound of the acoustic scene, the second type of microphone being associated with a microphone for receiving a diffuse sound portion of the acoustic scene.

In some examples, the first combination rule comprises forming a weighted sum of the first source signal and the second source signal, having a first weight g ₁ for the first source signal and a second weight g ₂ for the second source signal, the first weight g ₁ for the first source signal Source signal proportional to the inverse of a power of the first distance d ₁ and the second weight g ₂ for the second source signal is proportional to the inverse of a power of the second distance d ₂ .

In some examples, the second combination rule comprises forming a weighted sum of the first source signal x ₁ and the second source signal x ₂ , wherein the weights g ₁ and g ₂ are dependent on the first geometry information and the second geometry information, the weights g ₁ and g ₂ satisfy the boundary condition for all possible geometry information that a sum of the weights G = g ₁ + g ₂ or a quadratic sum G2 = g ₁ ² + g ₂ ^{2 is} constant, in particular 1.

• Überblendregel 1: x_virt1 = g₁*x₁ + * x₂, wobei g₂ =;
• Überblendregel 2: x_virt2 = cos*x₁ + sin*x₂, wobei δ ∈ [0°;90°];
• Uberblendregel 3: x_virt3 = g₁*x1 + g₂*x₂, wobei $\frac{\tan \theta }{\mathrm{<figure-callout\; id="0"\; label="tan\; \theta "\; filenames="imgf0005.png"\; state="\{\{state\}\}">tan</figure-callout>}{\theta }_{}{}_0}=\frac{{\mathrm{<figure-callout\; id="1"\; label="g"\; filenames="imgf0003.png,imgf0007.png"\; state="\{\{state\}\}">g</figure-callout>}}_1-g_2}{{\mathrm{<figure-callout\; id="1"\; label="g"\; filenames="imgf0003.png,imgf0007.png"\; state="\{\{state\}\}">g</figure-callout>}}_1+{\mathrm{<figure-callout\; id="2"\; label="g"\; filenames="imgf0007.png"\; state="\{\{state\}\}">g</figure-callout>}}_2}$
  
  und θ ∈ [0°; 90°];
• Uberblendregel 4: x_virt4=g₁*x₁+g₂*x₂, wobei $\frac{\sin \theta }{\mathrm{<figure-callout\; id="0"\; label="sin\; \theta "\; filenames="imgf0005.png"\; state="\{\{state\}\}">sin</figure-callout>}{\theta }_{}{}_0}=\frac{{\mathrm{<figure-callout\; id="1"\; label="g"\; filenames="imgf0003.png,imgf0007.png"\; state="\{\{state\}\}">g</figure-callout>}}_1-g_2}{{\mathrm{<figure-callout\; id="1"\; label="g"\; filenames="imgf0003.png,imgf0007.png"\; state="\{\{state\}\}">g</figure-callout>}}_1+{\mathrm{<figure-callout\; id="2"\; label="g"\; filenames="imgf0007.png"\; state="\{\{state\}\}">g</figure-callout>}}_2}$
  
  und θ ∈ [0°; 90°].

In some examples, the second combination rule comprises forming a weighted sum x _{virt of} the source signals x ₁ and x ₂ according to at least one of the following blending rules:

• _{Blend rule} 1: x _virt1 = g ₁ * x ₁ + * x ₂ , where g ₂ =;
• Blending _rule 2: x _virt2 = cos * x ₁ + sin * x ₂ , where δ ∈ [0 °, 90 °];
• Blend rule 3: x _virt3 = g ₁ * x1 + g ₂ * x ₂ , where $\frac{\tan \theta }{\mathrm{<figure-callout\; id="0"\; label="tan\; \theta "\; filenames="imgf0005.png"\; state="\{\{state\}\}">tan\; </figure-callout>}{\theta }_{}{}_0}=\frac{G_1-G_2}{{\mathrm{<figure-callout\; id="1"\; label="G"\; filenames="imgf0003.png,imgf0007.png"\; state="\{\{state\}\}">G</figure-callout>}}_1+{\mathrm{<figure-callout\; id="2"\; label="G"\; filenames="imgf0007.png"\; state="\{\{state\}\}">G</figure-callout>}}_2}$
  
  and θ ∈ [0 °; 90 °];
• Blend rule 4: x _virt4 = g ₁ * x ₁ + g ₂ * x ₂ , where $\frac{\sin \theta }{\mathrm{<figure-callout\; id="0"\; label="sin\; \theta "\; filenames="imgf0005.png"\; state="\{\{state\}\}">sin\; </figure-callout>}{\theta }_{}{}_0}=\frac{G_1-G_2}{{\mathrm{<figure-callout\; id="1"\; label="G"\; filenames="imgf0003.png,imgf0007.png"\; state="\{\{state\}\}">G</figure-callout>}}_1+{\mathrm{<figure-callout\; id="2"\; label="G"\; filenames="imgf0007.png"\; state="\{\{state\}\}">G</figure-callout>}}_2}$
  
  and θ ∈ [0 °; 90 °].

wobei x_virt23 entweder x_virt2 oder x_virt3 ist.In some examples, the combination _rule further comprises forming a weighted sum x _virt from the signals x _virt1 and x _virt23 weighted with a correlation coefficient σ _x1x2 for a correlation between the first source signal x ₁ and the second source signal x ₂ according to the following rule: $${\mathrm{x}}_{\mathrm{virt}}={\sigma }_{x1x2}*{\mathrm{<figure-callout\; id="1"\; label="x\; virt"\; filenames="imgf0003.png,imgf0007.png"\; state="\{\{state\}\}">x\; </figure-callout>}}_{\mathrm{virt}}+\left(1-{\sigma }_{x1x2}\right)*{\mathrm{x}}_{\mathrm{virt}23}$$

where x _{virt23 is} either x _virt2 or x _virt3 .

In some examples, according to the second combination rule, in forming the weighted sum, a third signal x ₃ having a third weight g _{3 is also} taken into account, wherein the positions of the microphones associated with the source signals x ₁ , x _2, and x ₃ span a triangular area within of which the virtual interception position is located and wherein the weights g ₁ , g ₂ and g ₃ for each of the source signals x ₁ , x ₂ and x _{3 are} respectively determined based on a perpendicular projection of the virtual interception position to the height of the triangle corresponding to the position associated with the respective source signal associated microphone.

Some examples are a tone signal generator for providing a virtual intercept position tone signal based on a first source signal and a second source signal, comprising: a geometry processor configured to obtain first geometry information based on a first position associated with the first source signal; to determine second geometry information associated with the second source signal; and a signal generator for providing the audio signal, wherein the signal generator is formed, at least the first source signal and to combine the second source signal according to a combination _rule using the first geometry _information and the second geometry _information , wherein according to the combination _rule, a first sub-signal x _{virt1 is formed} according to a first cross-fade _rule and a second sub-signal x _virt2 according to a second cross-fade _rule , and wherein the providing of the audio signal further _comprising forming a weighted sum x _virt from the signals x _virt1 and x _virt2 weighted with a correlation coefficient σ _x1x2 for a correlation between the first source signal x ₁ and the second source signal x ₂ .

wobei g₂ =; und das zweite Teilsignal x_virt2 unter Verwendung folgender Überblendregel: $${\mathrm{x}}_{\mathrm{virt}\mathrm{2}}=\cos *{\mathrm{x}}_{\mathrm{1}}+\sin *{\mathrm{x}}_{\mathrm{2}},$$

wobei δ ∈ [0°;90°] bereitgestellt, wobei das Bereitstellen der gewichteten Summe ferner folgende Berechnung umfasst: $${\mathrm{x}}_{\mathrm{virt}}={\sigma }_{x1x2}\ast {\mathrm{x}}_{\mathrm{virt}1}+\left(1-{\sigma }_{x1x2}\right)\ast {\mathrm{x}}_{\mathrm{virt}2}$$

In some examples of a tone signal _generator , the first sub-signal x _{virt1 is} made using the following first cross-fade _rule : $${\mathrm{<figure-callout\; id="1"\; label="x\; virt"\; filenames="imgf0003.png,imgf0007.png"\; state="\{\{state\}\}">x\; </figure-callout>}}_{\mathrm{virt}}={\mathrm{<figure-callout\; id="1"\; label="G"\; filenames="imgf0003.png,imgf0007.png"\; state="\{\{state\}\}">G</figure-callout>}}_{\mathrm{1}}*{\mathrm{x}}_{\mathrm{1}}+*{\mathrm{x}}_{\mathrm{2}}.\mathrm{in\; which}{\mathrm{<figure-callout\; id="2"\; label="G"\; filenames="imgf0007.png"\; state="\{\{state\}\}">G</figure-callout>}}_{\mathrm{2}}=;$$

where g ₂ =; and the second sub-signal x _virt2 using the following blending _rule : $${\mathrm{<figure-callout\; id="2"\; label="x\; virt"\; filenames="imgf0007.png"\; state="\{\{state\}\}">x\; </figure-callout>}}_{\mathrm{virt}}=\cos *{\mathrm{x}}_{\mathrm{1}}+\sin *{\mathrm{x}}_{\mathrm{2}}.$$

wherein δ ∈ [0 °; 90 °], wherein providing the weighted sum further comprises the following calculation: $${\mathrm{x}}_{\mathrm{virt}}={\sigma }_{x1x2}*{\mathrm{<figure-callout\; id="1"\; label="x\; virt"\; filenames="imgf0003.png,imgf0007.png"\; state="\{\{state\}\}">x\; </figure-callout>}}_{\mathrm{virt}}+\left(1-{\sigma }_{x1x2}\right)*{\mathrm{<figure-callout\; id="2"\; label="x\; virt"\; filenames="imgf0007.png"\; state="\{\{state\}\}">x\; </figure-callout>}}_{\mathrm{virt}}$$

Some examples are a tone signal generator for providing a sound signal for a virtual listening position within a room, in which an acoustic scene of at least a first microphone at a first known position within the room as a first source signal and at least a second microphone at a second known position is recorded within the room as a second source signal, comprising: an input interface configured to receive the first source signal received by the first microphone and the second source signal received by the second microphone; a geometry processor configured to determine, based on the first known position and the virtual listening position, first geometry information comprising a first distance between the first known position and the virtual listening position and a second distance based on the second known position and the virtual listening position determine second geometry information comprising between the second known position and the virtual listening position; and a signal generator for providing the sound signal, wherein the signal generator is configured to combine at least the first source signal and the second source signal according to a combination rule using the first geometry information and the second geometry information.

The features disclosed in the foregoing description, the appended claims and the appended figures may be taken to be and effect both individually and in any combination for the realization of an embodiment in its various forms.

Although some aspects have been described in the context of a tone signal generator, it will be understood that these aspects also constitute a description of the corresponding method such that a block or device of a tone signal generator is also to be understood as a corresponding method step or feature of a method step. Similarly, aspects described in connection with or as a method step also represent a description of a corresponding block or detail or feature of a corresponding tone signal generator.

Depending on particular implementation requirements, embodiments of the invention may be implemented in hardware or in software. The implementation may be performed using a digital storage medium, such as a floppy disk, a DVD, a Blu-Ray Disc, a CD, a ROM, a PROM, an EPROM, an EEPROM or FLASH memory, a hard disk, or other magnetic disk or optical memory are stored on the electronically readable control signals, which can cooperate with a programmable hardware component or cooperate such that the respective method is performed.

A programmable hardware component may be integrated by a processor, a central processing unit (CPU), a graphics processing unit (GPU), a computer, a computer system, an application-specific integrated circuit (ASIC) Circuit (IC = Integrated Circuit), a system on chip (SOC) system, a programmable logic element or a field programmable gate array with a microprocessor (FPGA = Field Programmable Gate Array) may be formed.

The digital storage medium may therefore be machine or computer readable. Thus, some embodiments include a data carrier having electronically readable control signals capable of interacting with a programmable computer system or programmable hardware component such that one of the methods described herein is performed. One embodiment is thus a data carrier (or a digital storage medium or a computer readable medium) on which the program is recorded for performing any of the methods described herein.

In general, embodiments of the present invention may be implemented as a program, firmware, computer program, or computer program product having program code or data, the program code or data operative to perform one of the methods when the program resides on a processor or a processor programmable hardware component expires. The program code or the data can also be stored, for example, on a machine-readable carrier or data carrier. The program code or the data may be present, inter alia, as source code, machine code or bytecode as well as other intermediate code.

Another embodiment is further a data stream, a signal sequence, or a sequence of signals that represents the program for performing any of the methods described herein. The data stream, the signal sequence or the sequence of signals can be configured, for example, to be transferred via a data communication connection, for example via the Internet or another network. Embodiments are also data representing signal sequences that are suitable for transmission over a network or a data communication connection, the data representing the program.

For example, a program according to one embodiment may implement one of the methods during its execution by, for example, reading or writing one or more data into memory locations, thereby optionally switching operations or other operations in transistor structures, amplifier structures, or others electrical, optical, magnetic or operating according to another operating principle components are caused. Accordingly, by reading a memory location, data, values, sensor values or other information can be detected, determined or measured by a program. A program can therefore acquire, determine or measure quantities, values, measured variables and other information by reading from one or more storage locations, as well as effect, initiate or execute an action by writing to one or more storage locations and control other devices, machines and components ,

The embodiments described above are merely illustrative of the principles of the present invention. It will be understood that modifications and variations of the arrangements and details described herein will be apparent to others of ordinary skill in the art. Therefore, it is intended that the invention be limited only by the scope of the appended claims and not by the specific details presented in the description and explanation of the embodiments herein.

Claims (15)

1. A mixing console (300) for processing at least a first and a second source signal and for providing a mixed audio signal, wherein the mixing console comprises an audio signal generator (100) for providing an audio signal (120) for a virtual listening position (202) within a space (200), in which an acoustic scene is recorded by at least a first microphone (204) at a first known position within the space (200) as the first source signal (210) and by at least a second microphone (206) at a second known position within the space (200) as the second source signal (212), wherein the audio signal generator (100) comprises:

   an input interface (102) configured to receive the first source signal (210) recorded by the first microphone (204) and the second source signal (212) recorded by the second microphone (206);

   a geometry processor (104) configured to determine a first piece of geometry information (110) based on the first position and the virtual listening position (202), and a second piece of geometry information (112) based on the second position and the virtual listening position (202); and

   a signal generator (106) for providing the audio signal (120), wherein the signal generator (106) is configured to combine at least the first source signal (210) and the second source signal (212) according to a combination rule using the first piece of geometry information (110) and the second piece of geometry information (112).
2. The mixing console (300) of claim 1, wherein the mixing console further comprises a user interface (306) configured to display a graphic representation of the positions of a plurality of microphones comprising at least the first and the second microphones and of the virtual listening position.
3. The mixing console (300) of claim 2, wherein the user interface (306) further comprises an input device configured to associate a microphone type from a group comprising of at least a first microphone type and a second microphone type with each of the microphones, wherein a microphone type corresponds to one kind of the sound field recorded using the microphone.
4. The mixing console (300) of one of claims 2 to 3, wherein the user interface (306) further comprises an input device configured to input or change at least the virtual listening position (202), particularly by influencing the graphic representation of the virtual listening position.
5. The mixing console (300) of one of the preceding claims, wherein the first piece of geometry information (110) comprises a first distance between the first position and the virtual listening position, and the second piece of geometry information (112) comprises a second distance between the second position and the virtual listening position (202).
6. The mixing console (300) of claim 5, wherein the combination rule comprises forming a weighted sum of the first source signal (210) and the second source signal (212), wherein the first source signal (210) is weighted by a first weight g₁ and the second source signal (212) is weighted by a second weight g₂.
7. The mixing console (300) of claim 6, wherein the first weight g₁ for the first source signal (210) is proportional to the inverse of a power of the first distance d₁ and the second weight g₂ for the second source signal (212) is proportional to the inverse of a power of the second distance d₂.
8. The mixing console (300) of claim 7, wherein the first weight g₁ for the first source signal (210) is proportional to the inverse of the first distance d₁ which is multiplied by a near-field radius r₁ of the first microphone, and the second weight g₂ for the second source signal (212) is proportional to the inverse of the second distance d₂ which is multiplied by a near-field radius r₂ of the second microphone.
9. The mixing console (300) of claim 6, wherein the first weight g₁ for the first source signal (210) is zero if the first distance d₁ is greater than a predetermined listening radius R, and the second weight g₂ for the second source signal (212) is zero if the second distance d₂ is greater than the predetermined listening radius R, wherein the first weight g₁ and the second weight g₂ are otherwise 1.
10. The mixing console (300) of any one of the preceding claims, wherein, according to the combination rule, either the first or the second source signal (212) is delayed by a delay time if a comparison of the first piece of geometry information and the second piece of geometry information meets a predetermined criterion.
11. The mixing console (300) of claim 10, wherein the predetermined criterion is met if a difference between the first distance and the second distance is greater than an operable minimum distance.
12. The mixing console (300) of any one of the preceding claims, wherein, according to the combination rule, the signal from the group of the first and the second source signals, which comprises a shorter signal propagation time from the microphone associated with the signal to the virtual listening position, is delayed such that the delayed signal propagation time corresponds to the signal propagation time from the microphone associated with the other signal of the group to the virtual listening position.
13. The mixing console (300) of any one of claims 6 to 12, wherein the first piece of geometry information (110) further comprises a first piece of directional information on a direction between a preferred direction (280) associated with the virtual listening position and the first position, and a second piece of directional information on a direction between the preferred direction and the second position, wherein the first weight g₁ is proportional to a first directional factor and wherein the second weight g₂ is proportional to a second directional factor, wherein the first direction factor depends on the first piece of directional information and on a directivity associated with the virtual listening position, and the second directional factor depends on the second piece of directional information and the directivity.
14. A method for providing an audio signal for a virtual listening position (202) within a space in which an acoustic scene is recorded by at least a first microphone at a first known position within the space as a first source signal and by at least a second microphone at a second known position within the space as a second source signal, comprising:

    Receiving (500) the first source signal recorded by the first microphone and the second source signal recorded by the second microphone;

    Determining (502) a first piece of geometry information comprising a first distance between the first known position and the virtual listening position (202) based on the first position and the virtual listening position, and a second piece of geometry information (112) comprising a second distance between the second known position and the virtual listening position (202) based on the second position and the virtual listening position; and

    Combining (504) at least the first source signal and the second source signal according to a combination rule using the first piece of geometry information (110) and the second piece of geometry information (112).
15. A computer program for executing the method of claim 14 if the computer program runs on a programmable hardware component.

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