Sound Acquisition via the Extraction of Geometrical Information from Direction of Arrival Estimates Description The present invention relates to audio processing and, in particular, to an apparatus and method for sound acquisition via the extraction of geometrical information from direction of arrival estimates. Traditional spatial sound recording aims at capturing a sound field with multiple microphones such that at the reproduction side, a listener perceives the sound image as it was at the recording location. Standard approaches for spatial sound recording usually use spaced, omnidirectional microphones, for example, in AB stereophony, or coincident directional microphones, for example, in intensity stereophony, or more sophisticated microphones, such as a B-format microphone, e.g. in Ambisonics, see, for example, [1] R. K. Furness, "Ambisonics - An overview," in AES 8th International Conference, April 1990, pp. 181-189. For the sound reproduction, these non-parametric approaches derive the desired audio playback signals (e.g., the signals to be sent to the loudspeakers) directly from the recorded microphone signals. Alternatively, methods based on a parametric representation of sound fields can be applied, which are referred to as parametric spatial audio coders. These methods often employ microphone arrays to determine one or more audio downmix signals together with spatial side information describing the spatial sound. Examples are Directional Audio Coding (DirAC) or the so-called spatial audio microphones (SAM) approach. More details on DirAC can be found in [2] Pulkki, V., "Directional audio coding in spatial sound reproduction and stereo upmixing," in Proceedings of the AES 28th International Conference, pp. 251-258, Pitea, Sweden, June 30 - July 2, 2006, [3] V. Pulkki, "Spatial sound reproduction with directional audio coding," J . Audio Eng. Soc, vol. 55, no. 6, pp. 503-516, June 2007. For more details on the spatial audio microphones approach, reference is made to [4] C. Fallen "Microphone Front-Ends for Spatial Audio Coders", in Proceedings of the AES 125th International Convention, San Francisco, Oct. 2008. In DirAC, for instance the spatial cue information comprises the direction-of-arrival (DOA) of sound and the diffuseness of the sound field computed in a time-frequency domain. For the sound reproduction, the audio playback signals can be derived based on the parametric description. In some applications, spatial sound acquisition aims at capturing an entire sound scene. In other applications spatial sound acquisition only aims at capturing certain desired components. Close talking microphones are often used for recording individual sound sources with high signal-to-noise ratio (SNR) and low reverberation, while more distant configurations such as XY stereophony represent a way for capturing the spatial image of an entire sound scene. More flexibility in terms of directivity can be achieved with beamforming, where a microphone array can be used to realize steerable pick-up patterns. Even more flexibility is provided by the abovementioned methods, such as directional audio coding (DirAC) (see [2], [3]) in which it is possible to realize spatial filters with arbitrary pick-up patterns, as described in [5] M. Kallinger, H. Ochsenfeld, G. Del Galdo, F. Kuch, D. Mahne, R. Schultz-Amling. and O. Thiergart, "A spatial filtering approach for directional audio coding," in Audio Engineering Society Convention 126, Munich, Germany, May 2009, as well as other signal processing manipulations of the sound scene, see, for example, [6] R. Schultz-Amling, F. Kiich, O. Thiergart, and M. Kallinger, "Acoustical zooming based on a parametric sound field representation," in Audio Engineering Society Convention 128, London UK, May 2010, [7] J. Herre, C. Falch, D. Mahne, G. Del Galdo, M. Kallinger, and O. Thiergart, "Interactive teleconferencing combining spatial audio object coding and DirAC technology," in Audio Engineering Society Convention 8, London UK, May 2010. All the above-mentioned concepts have in common that the microphones are arranged in a fixed known geometry. The spacing between microphones is as small as possible for coincident microphonics, whereas it is normally a few centimeters for the other methods. In the following, we refer to any apparatus for the recording of spatial sound capable of retrieving direction of arrival of sound (e.g. a combination of directional microphones or a microphone array, etc.) as a spatial microphone. Moreover, all the above-mentioned methods have in common that they are limited to a representation of the sound field with respect to only one point, namely the measurement location. Thus, the required microphones must be placed at very specific, carefully selected positions, e.g. close to the sources or such that the spatial image can be captured optimally. In many applications however, this is not feasible and therefore it would be beneficial to place several microphones further away from the sound sources and still be able to capture the sound as desired. There exist several field reconstruction methods for estimating the sound field in a point in space other than where it was measured. One method is acoustic holography, as described in [8] E. G. Williams, Fourier Acoustics: Sound Radiation and Nearfield Acoustical Holography, Academic Press, 99. Acoustic holography allows to compute the sound field at any point with an arbitrary volume given that the sound pressure and particle velocity is known on its entire surface. Therefore, when the volume is large, an unpractically large number of sensors is required. Moreover, the method assumes that no sound sources are present inside the volume, making the algorithm unfeasible for our needs. The related wave field extrapolation (see also [8]) aims at extrapolating the known sound field on the surface of a volume to outer regions. The extrapolation accuracy however degrades rapidly for larger extrapolation distances as well as for extrapolations towards directions orthogonal to the direction of propagation of the sound, see [9] A. untz and R. Rabenstein, "Limitations in the extrapolation of wave fields from circular measurements," in 15th European Signal Processing Conference (EUSIPCO 2007), 2007. [10] A. Walther and C. Faller, "Linear simulation of spaced microphone arrays using bformat recordings," in Audio Engineering Society Convention 128, London UK, May 2010, describes a plane wave model, wherein the field extrapolation is possible only in points far from the actual sound sources, e.g., close to the measurement point. A major drawback of traditional approaches is that the spatial image recorded is always relative to the spatial microphone used. In many applications, it is not possible or feasible to place a spatial microphone in the desired position, e.g., close to the sound sources. In this case, it would be more beneficial to place multiple spatial microphones further away from the sound scene and still be able to capture the sound as desired. [ 1 1] US61/287,596: An Apparatus and a Method for Converting a First Parametric Spatial Audio Signal into a Second Parametric Spatial Audio Signal, proposes a method for virtually moving the real recording position to another position when reproduced over loudspeakers or headphones. However, this approach is limited to a simple sound scene in which all sound objects are assumed to have equal distance to the real spatial microphone used for the recording. Furthermore, the method can only take advantage of one spatial microphone. It is an object of the present invention to provide improved concepts for sound acquisition via the extraction of geometrical information. The object of the present invention is solved by an apparatus according to claim 1, by a method according to claim 24 and by a computer program according to claim 25. According to an embodiment, an apparatus for generating an audio output signal to simulate a recording of a virtual microphone at a configurable virtual position in an environment is provided. The apparatus comprises a sound events position estimator and an information computation module. The sound events position estimator is adapted to estimate a sound source position indicating a position of a sound source in the environment, wherein the sound events position estimator is adapted to estimate the sound source position based on a first direction information provided by a first real spatial microphone being located at a first real microphone position in the environment, and based on a second direction information provided by a second real spatial microphone being located at a second real microphone position in the environment. The information computation module is adapted to generate the audio output signal based on a first recorded audio input signal being recorded by the first real spatial microphone, based on the first real microphone position, based on the virtual position of the virtual microphone, and based on the sound source position. In an embodiment, the information computation module comprises a propagation compensator, wherein the propagation compensator is adapted to generate a first modified audio signal by modifying the first recorded audio input signal, based on a first amplitude decay between the sound source and the first real spatial microphone and based on a second amplitude decay between the sound source and the virtual microphone, by adjusting an amplitude value, a magnitude value or a phase value of the first recorded audio input signal, to obtain the audio output signal. In an embodiment, the first amplitude decay may be an amplitude decay of a sound wave emitted by a sound source and the second amplitude decay may be an amplitude decay of the sound wave emitted by the sound source. According to another embodiment, the information computation module comprises a propagation compensator being adapted to generate a first modified audio signal by modifying the first recorded audio input signal by compensating a first delay between an arrival of a sound wave emitted by the sound source at the first real spatial microphone and an arrival of the sound wave at the virtual microphone by adjusting an amplitude value, a magnitude value or a phase value of the first recorded audio input signal, to obtain the audio output signal. According to an embodiment, it is assumed to use two or more spatial microphones, which are referred to as real spatial microphones in the following. For each real spatial microphone, the DOA of the sound can be estimated in the time-frequency domain. From the information gathered by the real spatial microphones, together with the knowledge of their relative position, it is possible to constitute the output signal of an arbitrary spatial microphone virtually placed at will in the environment. This spatial microphone is referred to as virtual spatial microphone in the following. Note that the Direction of Arrival (DOA) may be expressed as an azimuthal angle if 2D space, or by an azimuth and elevation angle pair in 3D. Equivalently, a unit norm vector pointed at the DOA may be used. In embodiments, means are provided to capture sound in a spatially selective way, e.g., sound originating from a specific target location can be picked up, just as if a close-up "spot microphone" had been installed at this location. Instead of really installing this spot microphone, however, its output signal can be simulated by using two or more spatial microphones placed in other, distant positions. The term "spatial microphone" refers to any apparatus for the acquisition of spatial sound capable of retrieving direction of arrival of sound (e.g. combination of directional microphones, microphone arrays, etc.) . The term "non-spatial microphone" refers to any apparatus that is not adapted for retrieving direction of arrival of sound, such as a single omnidirectional or directive microphone. It should be noted, that the term "real spatial microphone" refers to a spatial microphone as defined above which physically exists. Regarding the virtual spatial microphone, it should be noted, that the virtual spatial microphone can represent any desired microphone type or microphone combination, e.g. it can, for example, represent a single omnidirectional microphone, a directional microphone, a pair of directional microphones as used in common stereo microphones, but also a microphone array. The present invention is based on the finding that when two or more real spatial microphones are used, it is possible to estimate the position in 2D or 3D space of sound events, thus, position localization can be achieved. Using the determined positions of the sound events, the sound signal that would have been recorded by a virtual spatial microphone placed and oriented arbitrarily in space can be computed, as well as the corresponding spatial side information, such as the Direction of Arrival from the point-ofview of the virtual spatial microphone. For this purpose, each sound event may be assumed to represent a point like sound source, e.g. an isotropic point like sound source. In the following "real sound source" refers to an actual sound source physically existing in the recording environment, such as talkers or musical instruments etc.. On the contrary, with "sound source" or "sound event" we refer in the following to an effective sound source, which is active at a certain time instant or in a certain time-frequency bin, wherein the sound sources may, for example, represent real sound sources or mirror image sources. According to an embodiment, it is implicitly assumed that the sound scene can be modeled as a multitude of such sound events or point like sound sources. Furthermore, each source may be assumed to be active only within a specific time and frequency slot in a predefined time-frequency representation. The distance between the real spatial microphones may be so, that the resulting temporal difference in propagation times is shorter than the temporal resolution of the timefrequency representation. The latter assumption guarantees that a certain sound event is picked up by all spatial microphones within the same time slot. This implies that the DOAs estimated at different spatial microphones for the same time-frequency slot indeed correspond to the same sound event. This assumption is not difficult to meet with real spatial microphones placed at a few meters from each other even in large rooms (such as living rooms or conference rooms) with a temporal resolution of even a few ms. Microphone arrays may be employed to localize sound sources. The localized sound sources may have different physical interpretations depending on their nature. When the microphone arrays receive direct sound, they may be able to localize the position of a true sound source (e.g. talkers). When the microphone arrays receive reflections, they may localize the position of a mirror image source. Mirror image sources are also sound sources. A parametric method capable of estimating the sound signal of a virtual microphone placed at an arbitrary location is provided. In contrast to the methods previously described, the proposed method does not aim directly at reconstructing the sound field, but rather aims at providing sound that is perceptually similar to the one which would be picked up by a microphone physically placed at this location. This may be achieved by employing a parametric model of the sound field based on point-like sound sources, e.g. isotropic pointlike sound sources (IPLS). The required geometrical information, namely the instantaneous position of all IPLS, may be obtained by conducting triangulation of the directions of arrival estimated with two or more distributed microphone arrays. This might be achieved, by obtaining knowledge of the relative position and orientation of the arrays. Notwithstanding, no a priori knowledge on the number and position of the actual sound sources (e.g. talkers) is necessary. Given the parametric nature of the proposed concepts, e.g. the proposed apparatus or method, the virtual microphone can possess an arbitrary directivity pattern as well as arbitrary physical or non-physical behaviors, e. g., with respect to the pressure decay with distance. The presented approach has been verified by studying the parameter estimation accuracy based on measurements in a reverberant environment. While conventional recording techniques for spatial audio are limited in so far as the spatial image obtained is always relative to the position in which the microphones have been physically placed, embodiments of the present invention take into account that in many applications, it is desired to place the microphones outside the sound scene and yet be able to capture the sound from an arbitrary perspective. According to embodiments, concepts are provided which virtually place a virtual microphone at an arbitrary point in space, by computing a signal perceptually similar to the one which would have been 11 071629 picked up, if the microphone had been physically placed in the sound scene. Embodiments may apply concepts, which may employ a parametric model of the sound field based on point-like sound sources, e.g. point-like isotropic sound sources. The required geometrical information may be gathered by two or more distributed microphone arrays. According to an embodiment, the sound events position estimator may be adapted to estimate the sound source position based on a first direction of arrival of the sound wave emitted by the sound source at the first real microphone position as the first direction information and based on a second direction of arrival of the sound wave at the second real microphone position as the second direction information. In another embodiment, the information computation module may comprise a spatial side information computation module for computing spatial side information. The information computation module may be adapted to estimate the direction of arrival or an active sound intensity at the virtual microphone as spatial side information, based on a position vector of the virtual microphone and based on a position vector of the sound event. According to a further embodiment, the propagation compensator may be adapted to generate the first modified audio signal in a time-frequency domain, by compensating the first delay or amplitude decay between the arrival of the sound wave emitted by the sound source at the first real spatial microphone and the arrival of the sound wave at the virtual microphone by adjusting said magnitude value of the first recorded audio input signal being represented in a time-frequency domain. In an embodiment, the propagation compensator may be adapted to conduct propagation compensation by generating a modified magnitude value of the first modified audio signal by applying the formula: wherein di(k, n) is the distance between the position of the first real spatial microphone and the position of the sound event, wherein s(k, n) is the distance between the virtual position of the virtual microphone and the sound source position of the sound event, wherein Pref(k, n) is a magnitude value of the first recorded audio input signal being represented in a time-frequency domain, and wherein P (k, n) is the modified magnitude value. In a further embodiment, the information computation module may moreover comprise a combiner, wherein the propagation compensator may be furthermore adapted to modify a second recorded audio input signal, being recorded by the second real spatial microphone, by compensating a second delay or amplitude decay between an arrival of the sound wave emitted by the sound source at the second real spatial microphone and an arrival of the sound wave at the virtual microphone, by adjusting an amplitude value, a magnitude value or a phase value of the second recorded audio input signal to obtain a second modified audio signal, and wherein the combiner may be adapted to generate a combination signal by combining the first modified audio signal and the second modified audio signal, to obtain the audio output signal. According to another embodiment, the propagation compensator may furthermore be adapted to modify one or more further recorded audio input signals, being recorded by the one or more further real spatial microphones, by compensating delays between an arrival of the sound wave at the virtual microphone and an arrival of the sound wave emitted by the sound source at each one of the further real spatial microphones. Each of the delays or amplitude decays may be compensated by adjusting an amplitude value, a magnitude value or a phase value of each one of the further recorded audio input signals to obtain a plurality of third modified audio signals. The combiner may be adapted to generate a combination signal by combining the first modified audio signal and the second modified audio signal and the plurality of third modified audio signals, to obtain the audio output signal. In a further embodiment, the information computation module may comprise a spectral weighting unit for generating a weighted audio signal by modifying the first modified audio signal depending on a direction of arrival of the sound wave at the virtual position of the virtual microphone and depending on a virtual orientation of the virtual microphone to obtain the audio output signal, wherein the first modified audio signal may be modified in a time-frequency domain. Moreover, the information computation module may comprise a spectral weighting unit for generating a weighted audio signal by modifying the combination signal depending on a direction of arrival or the sound wave at the virtual position of the virtual microphone and a virtual orientation of the virtual microphone to obtain the audio output signal, wherein the combination signal may be modified in a time-frequency domain. According to another embodiment, the spectral weighting unit may be adapted to apply the weighting factor + (l-oc)cos((pv(k, n)), or the weighting factor 0.5 + 0.5 cos((pv(k, n)) on the weighted audio signal, wherein p v(k, n) indicates a direction of arrival vector of the sound wave emitted by the sound source at the virtual position of the virtual microphone. In an embodiment, the propagation compensator is furthermore adapted to generate a third modified audio signal by modifying a third recorded audio input signal recorded by an omnidirectional microphone by compensating a third delay or amplitude decay between an arrival of the sound wave emitted by the sound source at the omnidirectional microphone and an arrival of the sound wave at the virtual microphone by adjusting an amplitude value, a magnitude value or a phase value of the third recorded audio input signal, to obtain the audio output signal. In a further embodiment, the sound events position estimator may be adapted to estimate a sound source position in a three-dimensional environment. Moreover, according to another embodiment, the information computation module may further comprise a diffuseness computation unit being adapted to estimate a diffuse sound energy at the virtual microphone or a direct sound energy at the virtual microphone. The diffuseness computation unit may, according to a further embodiment, be adapted to estimate the diffuse sound energy at the virtual microphone by applying the formula: V _ - S wherein N is the number of a plurality of real spatial microphones comprising the first and the second real spatial microphone, and wherein is the diffuse sound energy at the i-th real spatial microphone. In a further embodiment, the diffuseness computation unit may be adapted to estimate the direct sound energy by applying the formula: distance SMi - IPLS distance VM IPLS wherein "distance SMi - IPLS" is the distance between a position of the i-th real microphone and the sound source position, wherein "distance VM - IPLS" is the distance between the virtual position and the sound source position, and wherein E r M is the direct energy at the i-th real spatial microphone. Moreover, according to another embodiment, the diffuseness computation unit may furthermore be adapted to estimate the diffuseness at the virtual microphone by estimating the diffuse sound energy at the virtual microphone and the direct sound energy at the virtual microphone and by applying the formula: wherein y ' indicates the diffuseness at the virtual microphone being estimated, wherein indicates the diffuse sound energy being estimated and wherein E indicates the direct sound energy being estimated. Preferred embodiments of the present invention will be described in the following, in which: illustrates an apparatus for generating an audio output signal according to an embodiment, Fig. 2 illustrates the inputs and outputs of an apparatus and a method for generating an audio output signal according to an embodiment, Fig. 3 illustrates the basic structure of an apparatus according to an embodiment which comprises a sound events position estimatior and an information computation module, Fig. 4 shows an exemplary scenario in which the real spatial microph depicted as Uniform Linear Arrays of 3 microphones each, Fig. 5 depicts two spatial microphones in 3D for estimating the direction of arrival in 3D space, Fig. 6 illustrates a geometry where an isotropic point-like sound source of the current time-frequency bin (k, n) is located at a position piPLs (k, n), Fig. 7 depicts the information computation module according to an embodiment, Fig. 8 depicts the information computation module according to another embodiment, Fig. 9 shows two real spatial microphones, a localized sound event and a position of a virtual spatial microphone, together with the corresponding delays and amplitude decays, Fig. 10 illustrates, how to obtain the direction of arrival relative to a virtual microphone according to an embodiment, Fig. 11 depicts a possible way to derive the DOA of the sound from the point of view of the virtual microphone according to an embodiment, Fig. 12 illustrates an information computation block additionally comprising a diffuseness computation unit according to an embodiment, Fig. 13 depicts a diffuseness computation unit according to an embodiment, Fig. 14 illustrates a scenario, where the sound events position estimation is not possible, and Fig. 15a- 15c illustrate scenarios where two microphone arrays receive direct sound, sound reflected by a wall and diffuse sound. Fig. 1 illustrates an apparatus for generating an audio output signal to simulate a recording of a virtual microphone at a configurable virtual position posVmic in an environment. The apparatus comprises a sound events position estimator 110 and an information computation module 120. The sound events position estimator 110 receives a first direction information dil from a first real spatial microphone and a second direction information di2 from a second real spatial microphone. The sound events position estimator 110 is adapted to estimate a sound source position ssp indicating a position of a sound source in the environment, the sound source emitting a sound wave, wherein the sound events position estimator 110 is adapted to estimate the sound source position ssp based on a first direction information dil provided by a first real spatial microphone being located at a first real microphone position poslmic in the environment, and based on a second direction information di2 provided by a second real spatial microphone being located at a second real microphone position in the environment. The information computation module 120 is adapted to generate the audio output signal based on a first recorded audio input signal isl being recorded by the first real spatial microphone, based on the first real microphone position poslmic and based on the virtual position posVmic of the virtual microphone. The information computation module 120 comprises a propagation compensator being adapted to generate a first modified audio signal by modifying the first recorded audio input signal is 1 by compensating a first delay or amplitude decay between an arrival of the sound wave emitted by the sound source at the first real spatial microphone and an arrival of the sound wave at the virtual microphone by adjusting an amplitude value, a magnitude value or a phase value of the first recorded audio input signal isl, to obtain the audio output signal. Fig. 2 illustrates the inputs and outputs of an apparatus and a method according to an embodiment. Information from two or more real spatial microphones 111, 12, 1IN is fed to the apparatus/is processed by the method. This information comprises audio signals picked up by the real spatial microphones as well as direction information from the real spatial microphones, e.g. direction of arrival (DOA) estimates. The audio signals and the direction information, such as the direction of arrival estimates may be expressed in a timefrequency domain. If, for example, a 2D geometry reconstruction is desired and a traditional STFT (short time Fourier transformation) domain is chosen for the representation of the signals, the DOA may be expressed as azimuth angles dependent on k and n, namely the frequency and time indices. In embodiments, the sound event localization in space, as well as describing the position of the virtual microphone may be conducted based on the positions and orientations of the real and virtual spatial microphones in a common coordinate system. This information may be represented by the inputs 121 ... 12N and input 104 in Fig. 2. The input 104 may additionally specify the characteristic of the virtual spatial microphone, e.g., its position and pick-up pattern, as will be discussed in the following. If the virtual spatial microphone comprises multiple virtual sensors, their positions and the corresponding different pick-up patterns may be considered. The output of the apparatus or a corresponding method may be, when desired, one or more sound signals 105, which may have been picked up by a spatial microphone defined and placed as specified by 104. Moreover, the apparatus (or rather the method) may provide as output corresponding spatial side information 106 which may be estimated by employing the virtual spatial microphone. Fig. 3 illustrates an apparatus according to an embodiment, which comprises two main processing units, a sound events position estimator 201 and an information computation module 202. The sound events position estimator 201 may carry out geometrical reconstruction on the basis of the DOAs comprised in inputs 111 ... 1IN and based on the knowledge of the position and orientation of the real spatial microphones, where the DOAs have been computed. The output of the sound events position estimator 205 comprises the position estimates (either in 2D or 3D) of the sound sources where the sound events occur for each time and frequency bin. The second processing block 202 is an information computation module. According to the embodiment of Fig. 3, the second processing block 202 computes a virtual microphone signal and spatial side information. It is therefore also referred to as virtual microphone signal and side information computation block 202. The virtual microphone signal and side information computation block 202 uses the sound events' positions 205 to process the audio signals comprised in 111... N to output the virtual microphone audio signal 105. Block 202, if required, may also compute the spatial side information 106 corresponding to the virtual spatial microphone. Embodiments below illustrate possibilities, how blocks 201 and 202 may operate. In the following, position estimation of a sound events position estimator according to an embodiment is described in more detail. Depending on the dimensionality of the problem (2D or 3D) and the number of spatial microphones, several solutions for the position estimation are possible. If two spatial microphones in 2D exist, (the simplest possible case) a simple triangulation is possible. Fig. 4 shows an exemplary scenario in which the real spatial microphones are depicted as Uniform Linear Arrays (ULAs) of 3 microphones each. The DOA, expressed as the azimuth angles al(k, n) and a2(k, n), are computed for the time-frequency bin (k, n). This is achieved by employing a proper DOA estimator, such as ESPRIT, [13] R. Roy, A. Paulraj, and T. Kailath, "Direction-of-arrival estimation by subspace rotation methods - ESPRIT," in IEEE International Conference on Acoustics, Speech, and Signal Processing (ICASSP), Stanford, CA, USA, April 1986, or (root) MUSIC, see [14] R. Schmidt, "Multiple emitter location and signal parameter estimation," IEEE Transactions on Antennas and Propagation, vol. 34, no. 3, pp. 276-280, 1986 to the pressure signals transformed into the time-frequency domain. In Fig. 4, two real spatial microphones, here, two real spatial microphone arrays 410, 420 are illustrated. The two estimated DOAs al(k, n) and a2(k, n) are represented by two lines, a first line 430 representing DOA al(k, n) and a second line 440 representing DOA a2(k, n). The triangulation is possible via simple geometrical considerations knowing the position and orientation of each array. The triangulation fails when the two lines 430, 440 are exactly parallel. In real applications, however, this is very unlikely. However, not all triangulation results correspond to a physical or feasible position for the sound event in the considered space. For example, the estimated position of the sound event might be too far away or even outside the assumed space, indicating that probably the DOAs do not correspond to any sound event which can be physically interpreted with the used model. Such results may be caused by sensor noise or too strong room reverberation. Therefore, according to an embodiment, such undesired results are flagged such that the information computation module 202 can treat them properly. Fig. 5 depicts a scenario, where the position of a sound event is estimated in 3D space. Proper spatial microphones are employed, for example, a planar or 3D microphone array. In Fig. 5, a first spatial microphone 510, for example, a first 3D microphone array, and a second spatial microphone 520, e.g. , a first 3D microphone array, is illustrated. The DOA in the 3D space, may for example, be expressed as azimuth and elevation. Unit vectors 530, 540 may be employed to express the DOAs. Two lines 550, 560 are projected according to the DOAs. In 3D, even with very reliable estimates, the two lines 550, 560 projected according to the DOAs might not intersect. However, the triangulation can still be carried out, for example, by choosing the middle point of the smallest segment connecting the two lines. Similarly to the 2D case, the triangulation may fail or may yield unfeasible results for certain combinations of directions, which may then also be flagged, e.g. to the information computation module 202 of Fig. 3. If more than two spatial microphones exist, several solutions are possible. For example, the triangulation explained above, could be carried out for all pairs of the real spatial microphones (if N = 3, 1 with 2, 1 with 3, and 2 with 3). The resulting positions may then be averaged (along x and y, and, if 3D is considered, z). Alternatively, more complex concepts may be used. For example, probabilistic approaches may be applied as described in [15] J . Michael Steele, "Optimal Triangulation of Random Samples in the Plane", The Annals of Probability, Vol. 10, No.3 (Aug., 1982), pp. 548-553. According to an embodiment, the sound field may be analyzed in the time-frequency domain, for example, obtained via a short-time Fourier transform (STFT), in which k and n denote the frequency index k and time index n, respectively. The complex pressure P (k, n) at an arbitrary position p for a certain k and n is modeled as a single spherical wave emitted by a narrow-band isotropic point-like source, e.g. by employing the formula: P v (k, n ) = P P s ( , n ) (ft, IP LS (ft, n), p v ) , where PiPLs(k, n) is the signal emitted by the IPLS at its position p Ls( , n). The complex factor y(k, P P S Pv) expresses the propagation from pipLs(k, n) to pv, e.g., it introduces appropriate phase and magnitude modifications. Here, the assumption may be applied that in each time-frequency bin only one IPLS is active. Nevertheless, multiple narrow-band IPLSs located at different positions may also be active at a single time instance. Each IPLS either models direct sound or a distinct room reflection. Its position pipLs (k, n) may ideally correspond to an actual sound source located inside the room, or a mirror image sound source located outside, respectively. Therefore, the position piPLs ( , n) may also indicates the position of a sound event. Please note that the term "real sound sources" denotes the actual sound sources physically existing in the recording environment, such as talkers or musical instruments. On the contrary, with "sound sources" or "sound events" or "IPLS" we refer to effective sound sources, which are active at certain time instants or at certain time-frequency bins, wherein the sound sources may, for example, represent real sound sources or mirror image sources. Fig. 15a-15b illustrate microphone arrays localizing sound sources. The localized sound sources may have different physical interpretations depending on their nature. When the microphone arrays receive direct sound, they may be able to localize the position of a true sound source (e.g. talkers). When the microphone arrays receive reflections, they may localize the position of a mirror image source. Mirror image sources are also sound sources. Fig. 15a illustrates a scenario, where two microphone arrays 1 1 and 152 receive direct sound from an actual sound source (a physically existing sound source) 153. Fig. 15b illustrates a scenario, where two microphone arrays 161, 162 receive reflected sound, wherein the sound has been reflected by a wall. Because of the reflection, the microphone arrays 161, 162 localize the position, where the sound appears to come from, at a position of an mirror image source 165, which is different from the position of the speaker 163. Both the actual sound source 153 of Fig. 15a, as well as the mirror image source 165 are sound sources. Fig. 15c illustrates a scenario, where two microphone arrays 171, 172 receive diffuse sound and are not able to localize a sound source. While this single-wave model is accurate only for mildly reverberant environments given that the source signals fulfill the W-disjoint orthogonality (WDO) condition, i.e. the timefrequency overlap is sufficiently small. This is normally true for speech signals, see, for example, [12] S. Rickard and Z. Yilmaz, "On the approximate W-disjoint orthogonality of speech," in Acoustics, Speech and Signal Processing, 2002. ICASSP 2002. IEEE International Conference on, April 2002, vol. 1. However, the model also provides a good estimate for other environments and is therefore also applicable for those environments. In the following, the estimation of the positions piPLs(k, n) according to an embodiment is explained. The position i Ls (k, n) of an active IPLS in a certain time-frequency bin, and thus the estimation of a sound event in a time-frequency bin, is estimated via triangulation on the basis of the direction of arrival (DOA) of sound measured in at least two different observation points. Fig. 6 illustrates a geometry, where the IPLS of the current time-frequency slot (k, n) is located in the unknown position piPLs (k, n). In order to determine the required DOA information, two real spatial microphones, here, two microphone arrays, are employed having a known geometry, position and orientation, which are placed in positions 610 and 620, respectively. The vectors i and p2 point to the positions 610, 620, respectively. The array orientations are defined by the unit vectors c i and c2. The DOA of the sound is determined in the positions 610 and 620 for each (k, n) using a DOA estimation algorithm, for instance as provided by the DirAC analysis (see [2], [3]). By this, a first point-of-view unit vector e ov (k, n) and a second point-of-view unit vector e 0V(k, n) with respect to a point of view of the microphone arrays (both not shown in Fig. 6) may be provided as output of the DirAC analysis. For example, when operating in 2D, the first point-of-view unit vector results to: (2) Here, cpi(k, n) represents the azimuth of the DOA estimated at the first microphone array, as depicted in Fig. 6. The corresponding DOA unit vectors ei(k, n) and e (k, n), with respect to the global coordinate system in the origin, may be computed by applying the formulae: e \ k , n ) = R e k , n), e 2 (fc, n ) = · e (fc, ) , (3) where R are coordinate transformation matrices, e.g., (4) when operating in 2D and C C