Methods and apparatus for estimating a position dependent sound level in a reverberant environment

Abstract

Methods, apparatus, programs, and storage media for estimating a sound level of an audio source at a position in a reverberant environment are described. The method includes obtaining position information for the position, wherein the position information comprises a distance between the position and the audio source, and an angle of the position with respect to a main acoustic axis of the audio source. The sound level of the audio source is estimated at the position based on a reverberant sound field model of the audio source and a direct sound field model of the audio source. The direct sound field model is based on both the distance and the angle in the position information. The estimated sound level is output for further processing.

Description

[0001]METHODS AND APPARATUS FOR ESTIMATING A POSITION DEPENDENT SOUND LEVEL IN A REVERBERANT ENVIRONMENT CROSS-REFERENCE TO RELATED APPLICATIONS This application claims the benefit of priority from Spanish Patent Application No. P202430652, filed on August 5, 2024, United States Provisional Patent Application No.63 / 704,12, filed on October 7, 2024 and European Patent Application 24205067.2, filed on October 7, 2024, all of which are incorporated by reference herein in their entirety. TECHNICAL FIELD The present disclosure relates to estimation of a sound level of an audio source and more particular to methods and devices for estimating a sound level of an audio source at a specific position in a reverberant environment. BACKGROUND When a loudspeaker system includes multiple loudspeakers that are distributed over a listening environment, e.g., a living room, a listening experience of a user of the loudspeaker system may vary widely, depending on the position of the user with respect to each loudspeaker of the loudspeaker system. The (perceived) sound level of a loudspeaker may vary depending on a distance from the loudspeaker and depending on an angle from a main acoustic axis of the loudspeaker. To improve consistency of the listening experience, loudspeaker signals can be adjusted based on a position of the user with respect to each loudspeaker. Currently, two separate (independent) methods are used for adjusting the loudspeaker signal, namely directivity compensation and distance compensation. While directivity compensation targets an angle-specific gain adjustment and ignores the distance, distance compensation targets a distance-specific gain adjustment and ignores the angle of the user with respect to the main acoustic axis. The angle dependent attenuation of the sound waves emitted by the loudspeaker may however also be distance dependent, while the distance dependent attenuation may also be angle dependent. Therefore, treating angle and distance as independent variables for loudspeaker gain compensation may lead to suboptimal results. Thus, there is a need for an improved approach to estimation of sound levels in different positions in a reverberant environment. SUMMARY In view of the above, the present disclosure provides methods, apparatus, and programs, as well as computer-readable storage media for sound level estimation of an audio source at various positions in a reverberant environment. According to an aspect of the disclosure, a method of estimating a sound level of an audio source at a position in a reverberant environment is provided. The reverberant environment may include objects that reflect and diffuse sound emitted by the audio source. Position information for the position may be obtained (e.g. received, simulated or measured), wherein the position information may include a distance between the position and the audio source, and an angle of the position with respect to a main acoustic axis of the audio source. The main acoustic axis of the audio source may be a direction of a maximum sound pressure level emitted by the audio source. The sound level of the audio source may be estimated at the position based on a reverberant sound field model of the audio source and a direct sound field model of the audio source, wherein the direct sound field model may be based on both the distance and the angle in the position information. The estimated sound level may be output for further processing. As the model for the direct sound field takes into account the interdependency between angle and distance, the (perceived) sound level can be determined (estimated) more precisely. The estimated sound level can then be used for improving a sound experience for a user in various situations. In some embodiments, the reverberant environment may be a real environment and the audio source may be a loudspeaker or multiple loudspeakers of a loudspeaker system, i.e. the loudspeakers may be controllable by a single entity. In this case, the estimated sound level may be used for compensation of a loudspeaker signal. This compensation may include determining a compensation filter based on the estimated sound level and a target (e.g. desired) sound level. The target sound level may be used for matching the sound level of a first loudspeaker and a sound level of at least a second loudspeaker at the position. Further, the determined compensation filter may be applied to a signal fed to the loudspeaker. The compensation filter may be implemented as any one of an FIR filter, an IIR filter, or a transform-based filter. In a case of multiple loudspeakers, a compensation filter may be determined for each loudspeaker or all but one loudspeaker and applied to the signal fed to each loudspeaker or all but one loudspeaker. By compensating the signals fed to the loudspeaker in this way, the sound level at the specific position of each individual loudspeaker will be closer to the target level. Thereby, a sound experience may be improved for the user of the loudspeaker system.In some embodiments, a desired response of the compensation filter is defined as ^^(^) =^(^)^(^,^,^), where ^(^, ^, ^) may be the estimated sound level, ^(^) may be the target sound level, ^be frequency, d may be the distance and ^ may be In some embodiments, a smoothing function may be applied to ^(^, ^, ^). In addition oralternatively, regularization may be used for determining the desired response of the compensation filter. In some embodiments, obtaining the position information may include receiving the position information (e.g. the position information is determined externally). Alternatively, the position information may be determined based on measuring or estimating the position. In the case of measuring the position, position information may be determined based on any combination of camera sensor data, proximity sensor data, acoustic sensor data, inertial measurement unit data, virtual reality headset data and radio sensor data. In some embodiments, multiple listeners may be listening to the loudspeaker or the multiple loudspeakers. In this case, the listener position may correspond to a combination of multiple listener positions. The combination of multiple listener positions may include the combination of the distances between each listener and the audio source and the combination of the angles between each listener and the main acoustic axis of the audio source. The combination of the multiple listener positions may be a weighted average of the multiple listener positions or any other suitable combination of the individual values. In some embodiments, the method may further include tuning parameters for the reverberant sound field model and the direct sound field model based on the reverberant environment (i.e. the real environment). In some embodiments, tuning the parameters for the reverberant sound field model and the direct sound field model based on the reverberant environment may include measuring or simulating an impulse response of the audio source at one or more positions in the reverberant environment. The one or more positions may comprise the position used for estimating the sound level. The method may further include tuning the parameters based on the measured or simulated impulse response. Thereby, parameters of the models used for the sound level estimation can be further tuned based on environmental parameters and estimation precision may be increased. In some embodiments, before tuning the parameters based on the impulse response, the method may further include dividing the impulse response into a first part and a second part at truncation time τ, wherein the first part corresponds to a direct sound portion and the second part corresponds to a reverberant sound portion. Alternatively, the method may further include dividing the impulse response into a first part and a second part based on a frequency-dependent truncation kernel, wherein the first part may correspond to a direct sound portion and the second part corresponds to a reverberant sound portion. The first part may be defined as IRDS(^) = ^^^ FDT(^, ^^) IR(^′) ^^′ and the second part may be defined asIRRS(^) = (^) − ^^^ FDT(^, ^^) IR(^′) ^^′ , where FDT(^, ^^) may be the frequency-dependenttruncation kernel, may be the impulse response and ^ may be a time constant. In some embodiments, the reverberant environment may be a virtual reality environment or an augmented reality environment, and the audio source may be a virtual audio source. In this case, the estimated sound level may be used for rendering the virtual audio source by a virtual (or augmented) reality device used for rendering the virtual (or augmented) reality environment. A sound of the rendered virtual audio source may be output by a headphone of the virtual reality device, or a headphone or loudspeaker connected to the virtual reality device. In the case of headphones, the virtual audio source may be rendered binaurally with the estimated sound level. Thereby, the (perceived) sound level of an audio source in a virtual environment may be closer to a sound level expected by the user based on the perceived virtual environment. This may lead to an improved perceived spatial accuracy of the audio source in the virtual or augmented reality environment Therefore, immersion may be improved. In some embodiments, the sound level may be proportional to a power spectrum of the audio source. The power spectrum may be determined based on measurement data or simulation data of the audio source, or the power spectrum may be based on data from a database. In some embodiments, the direct sound field model of the audio source may be frequency dependent. The direct sound field model of the audio source may be further based on a distance decay model and the power spectrum of the audio source. The direct sound field model of theaudio source may then be defined as ^^^(^, ^, ^) = ^ ^^^^ ^^(^, ^), wherein ^^^(^, ^, ^) may be alevel of the direct sound field model, ^ may be ^ may be the distance, may be adecay parameter of the distance decay model and larger than 0, ^ may be the angle, and ^^(^, ^) may be the power spectrum of the audio source. The decay parameter may be equal to 2 in apreferred embodiment. ^^(^, ^) may be determined based on the power spectrum on the mainacoustic axis of the audio source and a high shelf filter model. The power spectrum may then bedefined as ^^(^, ^) = !"(#(^), ^)^^(^, ^ = 0), where ^^(^, ^ = 0) may be the powerspectrum on the main acoustic axis of the audio source, !"(#(^), ^) may be the high shelf filtermodel, and #(^) may be an angle dependent gain in decibels of the high shelf filter model. Theangle dependent gain of the high shelf filter model may be defined as #(^) =0 &^ ^ < ^^^+^,* ≤ ^ < + , gain in decibels of the high shelf filter model, ^^may be a first threshold angle, and ^0may be a second threshold angle. Alternatively, the power spectrum may be based on a database entry lookup. The database may include a value of the power spectrum for each angle. In some embodiments, the reverberant sound field model of the audio source may be independent of both the distance and the angle. In some embodiments, the reverberant sound field model of the audio source may be frequency dependent. In some embodiments, the reverberant sound field model of the audio source may depend on a critical distance. The critical distance may be a distance from the audio source at which a level of the direct sound field model and a level of the reverberant sound field model are equal. Thereverberant sound field model of the audio source may be defined as ^2^(^) = ^^ ^)34(^) ^^5-(^)^,wherein ^2^(^) may be a level of the reverberant sound field 67may be the critical distance, and ^^)34(^) may be an average power spectrum of the audio source. The average power spectrum may be an average of the power spectrum of the audio source over an angle range. The angle range may correspond to the full sphere or a sphericalsector. The average power spectrum may be defined as ^^)34(^) = ^^(^, ^ = 0)!"8#)34 , ^9,where # may be an average gain of #( ) )34^ . Instead reverberant sound field model may also be defined based on a flat power spectrum, i.e., a power spectrum that does not change with frequency. Alternatively, the reverberant sound field model may be based on a high-frequency roll-off power spectrum, i.e. the power spectrum may decay over frequency by a predefined rule, e.g., exponential decay. In some embodiments, the reverberant sound field model of the audio source may further depend on the distance between the position and the audio source. In this case, the reverberant sound field model may be defined as ^2^(^, ^) = ^^)34(^) : ^^;^(-<;) =, where ^2^(^, ^) may be a5 (^)level of the reverberant sound field model, 67may be the critical distance, ^^)34(^) may be an average power spectrum of the audio source and α may be a decay parameter with 0 ≤ > ≤ 2and > ≪ . Therefore, a decay parameter of the reverberant sound field model may be smallerthan a decay parameter of the direct sound field model. In some embodiments, the reverberant sound field model of the audio source may further depend on the angle. In this case, the reverberant sound field model may be defined as ^2^(^, ^, A) =!"(#^(^), ^)^^)34(^) : ^^;^(-<;) =, where ^2^(^, ^, A) may be a level of the reverberant sound5 (^)distance, ^^)34(^) may be an average power spectrum of the audio source, α may be a decay parameter with 0 ≤ > ≤ 2 and > ≪ , and #^(^) may be theangle dependent gain in decibels of the high shelf filter model with a maximum gain #′()* ≪#()*. Therefore, a maximum negative gain of a high shelf filter for reverberant sound fieldmodel may be larger than a maximum negative gain of a high shelf filter for the direct sound field. In summary, an angle and distance dependency of the reverberant sound field model may be less prominent compared with the angle and distance dependency of the direct sound field model. In some embodiments, estimating the sound level of the audio source at the position may include determining the sound level of the audio source as ^(^, ^, ^) = (B^^^(^, ^, ^)CD +B^2^(^, ^, ^)CD)^ / D, where ^2^(^, ^, A) may be a level of the reverberant sound field model,^, ^) may be a level of the direct sound field model, ^ may be frequency, ^ may be thedistance, ^ may be the angle, and F may be a coherence tuning parameter. In a preferred embodiment, F = 1, i.e. the sound level is estimated as the sum of the levels of the direct soundfield model and the reverberant sound field model. In some embodiments, the sound level of the audio source may be a measurable loudness or a perceived loudness. In some embodiments, the audio source may be moving and / or the position may be changing. In some embodiments, the position information may be updated continuously. According to another aspect, a method of estimating a sound level of an audio source at a position in a reverberant environment is provided. The reverberant environment may include objects that reflect and diffuse sound emitted by the audio source. Position information for the position may be obtained (e.g. received, simulated or measured), wherein the position information may include a distance between the position and the audio source, and an angle of the position with respect to a main acoustic axis of the audio source. The main acoustic axis of the audio source may be a direction of a maximum sound pressure level emitted by the audio source. The sound level of the audio source may be estimated at the position based on a reverberant sound field model of the audio source and a direct sound field model of the audio source. The direct sound field model may be based on the angle and a weight of the weighted sum depends on the distance. The estimated sound level may be output for further processing. In some embodiments, estimating the sound level of the audio source at the position may includedetermining the sound level of the audio source as ^(^, ^, ^) = HI&Jℎ^^^(^) ∗ ^^^(^, ^) +^2^(^, ^, ^), wherein ^2^(^, ^, A) may be a level of the reverberant sound field model, ^^^(^, ^)may be a level of the direct sound field model, HI&Jℎ^^^(^) may be the weights, ^ may be frequency, ^ may be the distance, and ^ may be the angle. In some embodiments, estimating the sound level of the audio source at the position may includedetermining the sound level of the audio source as ^(^, ^, ^) = HI&Jℎ^^^(^) ∗ ^^^(^, ^) +HI&Jℎ^2^(^) ∗ ^2^(^, ^), wherein ^2^(^, A) may be a level of the reverberant sound fieldmodel, ^^^(^, ^) may be a level of the direct sound field model, HI&Jℎ^^^(^) may be a firstweight of the weighted sum, HI&Jℎ^2^(^) may be a second weight of the weighted sum withHI&Jℎ^2^(^) = 1 − HI&Jℎ^^^(^), ^ may be frequency, ^ may be the distance, and ^ may bethe angle.. In some embodiments, HI&Jℎ^^^(^) may be equal to 1 when ^ is smaller than a minimaldistance. HI&Jℎ^^^(^) may be 0 when the distance is equal to a critical distance, andHI&Jℎ^^^(^) may decrease linearly between the minimimal distance and the critital distance.In some embodiments the reverberant sound field may be independent of the angle and the frequency. Aspects of the present disclosure may be implemented via an apparatus. The apparatus may include a processor and memory coupled to the processor. The processor may be adapted carry out the method according to aspects and embodiments of the present disclosure. The apparatus may be a loudspeaker or a device connected to the loudspeaker. Alternatively, the apparatus may be a virtual reality headset, an augmented reality headset or a device connected to the virtual reality headset or the augmented reality headset. The apparatus may be included in a device that also processes a received bitstream. The bitstream may include audio data and / or metadata to be used during rendering by the apparatus. The processing may be conducted in accordance with methods disclosed herein. Aspects of the present disclosure may be implemented via a program. When instructions of the program are executed by a processor, the processor may carry out aspects and embodiments of the present disclosure. A computer-readable storage medium may store the program. Such computer-readable storage media may include memory devices such as those described herein, including but not limited to random access memory (RAM) devices, read-only memory (ROM) devices, etc.. Accordingly, some innovative aspects of the subject matter described in this disclosure can be implemented via one or more computer-readable storage media having software stored thereon. It will be appreciated that apparatus features and method steps may be interchanged in many ways. In particular, the details of the disclosed method(s) can be realized by the corresponding apparatus (or system), and vice versa, as the skilled person will appreciate. Moreover, any of the above statements made with respect to the method(s) are understood to likewise apply to the corresponding apparatus (or system), and vice versa. BRIEF DESCRIPTION OF DRAWINGS Example embodiments of the disclosure are explained below with reference to the accompanying drawings, wherein Fig.1 schematically illustrates a loudspeaker off-axis listener in a reverberant environment, Fig.2 is a flowchart illustrating an example of a method of estimating a sound level of an audio source at a position in a reverberant environment according to embodiments of the disclosure, Fig.3 is a flowchart illustrating an example of a method of tuning parameters for the reverberant sound field model and the direct sound field model based on the reverberant environment according to embodiments of the disclosure, Fig.4 schematically illustrates examples of different off-axis positions with respect to a loudspeaker, Fig.5 depicts an example impulse response of a loudspeaker and its two components according to embodiments of the disclosure, Figs.6A, 6B, and 6C respectively depict a magnitude spectrum of a full impulse response, a direct sound component of the impulse response, and a reverberant sound component of the impulse response of a loudspeaker measured at the three different off-axis positions of Fig.4 according to embodiments of the disclosure, Fig.7 schematically illustrates an example of an apparatus for estimating a sound level of an audio source according to embodiments of the disclosure. DETAILED DESCRIPTION The Figures (Figs.) and the following description relate to preferred embodiments by way of illustration only. It should be noted that from the following discussion, alternative embodiments of the structures and methods disclosed herein will be readily recognized as viable alternatives that may be employed without departing from the principles of what is claimed. Reference will now be made in detail to several embodiments, examples of which are illustrated in the accompanying figures. It is noted that wherever practicable similar or like reference numbers may be used in the figures and may indicate similar or like functionality. The figures depict embodiments of the disclosed system (or method) for purposes of illustration only. One skilled in the art will readily recognize from the following description that alternative embodiments of the structures and methods illustrated herein may be employed without departing from the principles described herein. Fig.1 schematically illustrates a loudspeaker off-axis listener in a reverberant environment. In Fig.1 only on loudspeaker is depicted. In general however, a listener may use a loudspeaker system with multiple loudspeakers. A loudspeaker system may be tuned to generate the intended sound experience at a specific location, e.g. a center location. When the listener moves away from the optimal position, the sound experience may be degraded, as sound levels of the different loudspeakers at the new location may depart from an intended sound level. This change of sound level is caused by a change of an angle with respect to a main acoustic axis of a loudspeaker and a change of the distance to the loudspeaker. To alleviate these effects, loudspeaker loudness matching is performed. Generally, loudspeaker loudness matching ensures that the loudness from the different loudspeakers is substantially the same either at the listener position or across the listening area. Loudness may be defined to be as “the attribute of auditory sensation in terms of which sounds can be ordered on a scale extending from quiet to loud” [American National Standards Institute, "American national psychoacoustical terminology" S3.20, 1973, American Standards Association]. Empirical studies have shown that loudness correlates well with intensity estimates of the sound. In one example, loudness can be roughly estimated by the A-level intensity of the sound as measured in decibels. In another example, loudness can be estimated more accurately by the method detailed in ITU-R BS.1770. Currently, sound level compensation for angle and distance are treated completely independently in MPEG-I. MPEG-I is a set of international standards developed by the International Standards Organization (ISO) in conjunction with the International Electrotechnical Commission (IEC). In particular, the ISO / IEC Moving Picture Experts Group has developed the MPEG-I standard to enable the creation, storage, and transmission of immersive media experiences, including virtual reality (VR), 360-degree video, and 3D audio, on a wide range of devices—from smartphones to VR headsets. The MPEG-I standard document is published as ISO / IEC 23090-4. References throughout this document include some references to independent angle and distance sound level compensation in the context of the MPEG-I standard document ISO / IEC 23090-4, which is incorporated by reference herein in its entirety. Loudspeaker Directivity Compensation Current procedures to compensate for the directivity of loudspeakers in a reverberant room define a high-shelving filter that boosts high frequencies up to a maximum gain, purely based on the angle with which the user is located with respect to the on-axis direction of the loudspeaker. The assumption for loudspeaker directivity is that high frequencies will be more and more attenuated as the listener moves off-axis from main direction of the loudspeaker. In the example implementation of MPEG-I, in total, four different parameters are available to change the behavior of the Directivity Compensation (DC). The DC applies a 2nd order high shelf filter defined by (maximum) gain and frequency. A first angle defines the angle at which the DC starts to be applied, whereas a second angle defines the additional range with linearly increasing gain up to which the DC is fully applied. For angles larger than this, there is no further increase in the gain of the filter. Listener to loudspeaker distance is not considered. Distance compensation: feasible approaches The loudness at the listener position can be inferred by assuming a particular sound decay model. A common decay model is one of direct sound plus diffuse sound field. The model assumes that the sound has two main components: direct sound, whose corresponding intensity decays as the squared distance from the loudspeaker, accounting for the sound arriving directly from the loudspeaker to the user, and possibly some of the early reflections, and a diffuse sound field, which is constant for a source in the room, accounting for the reverberation produced by the late reflections in the room. This model has three parameters: the acoustic power of the source ^M, its directivity ^M, and the critical distance 67. All three parameters can be broadband or per frequency band, leading respectively to broadband or per frequency band loudness estimates. The critical distance 67is defined to be as the distance from the loudspeaker at which the direct sound and the reverberant sound field are equal. On axis, the loudness can be estimated from the total sound intensity in decibel scale as: ^^ ^ 1M = 10 log^^ Q M M 14S T^0 + 0U VM 67For rooms of comparable time, the smaller the critical distance. Alternatively, a decay model may only assume a direct sound component. In this case, the loudness can be estimated from the total sound intensity in decibel scale as: ^= 10 log ^Q M^M 1M ^^ 4S ^0 VThe current MPEG-I distance to fundamentally the same approach allowing for different slopes of the direct sound decay, diffuse decay and a further beta transition parameter. While adding degrees of freedom to fit real room behavior to the model, it is still fundamentally limited to just the distance as the input variable, disregarding the angle. The current MPEG-I implementation for distance compensation is introduced in section 6.7.2.1.2 (Physical compensation level (Level 1)) and the specific implementation can be found in section 6.7.2.2.2.4 (Update real-time parameter). Joint distance and directivity model With regards to directivity compensation, experiments in reverberant rooms have shown that in addition to the angular dependence, there is a further dependency on the distance between listener and loudspeaker defining how strong (and for which frequencies exactly) the mid and high frequencies are attenuated as the user moves away from the main axis of the loudspeaker. Likewise, the distance decay parameters chosen might depend on the listener-to-loudspeaker angle. Therefore, a more sophisticated model is needed for estimating the sound level based on an angle and a distance of a listener with respect to a loudspeaker. To achieve this, a more complete, distance and angle-based decay model that more accurately takes into account direct and reverberant components is proposed. Fig.2 illustrates an example of a method 300 of estimating a sound level of an audio source at a position in a reverberant environment. In particular, the method considers angle and distance of a position in the reverberant environment simultaneously to more accurately determine the amount of attenuation of sound, i.e. the sound level, at the position. In step S301, position information for a position is obtained. The position may be considered as any position (relative to an audio source) in a reverberant environment. The position information may be received, i.e., the method itself may not be aware of an actual position. Alternatively, the position information may be determined from an actual position in the reverberant environment. The reverberant environment may be any environment with objects that reflect and diffuse the sound emitted by an audio source. In such an environment, sound emitted by an audio source may not only be received directly (direct sound), but also through reflections in the environment (reverberant sound). The position information comprises a relative position with respect to an audio source, i.e., a distance to the audio source and an angle with respect to a main acoustic axis of the audio source. The main acoustic axis of an audio source may be an axis on which a sound pressure level of the audio source is maximal, i.e. the sound pressure in the direction of the axis may be larger compared to all other directions (around the audio source). The main acoustic axis may be defined by hardware constraints, e.g. a direction of a loudspeaker driver. Alternatively or additionally, the main acoustic axis may be defined by a beam steering direction of the audio source. The position may be a position of a listener for which an audio experience should be improved by this method. If the method determines the position, the position may be measured or estimated based on sensor data. Any sensor data suitable for tracking a position of a listener may be used. Without limitation, such data may be camera sensor data, proximity sensor data, acoustic sensor data, inertial measurement unit data, virtual reality headset data and radio sensor data. If not one, but multiple listeners are in the reverberant environment, optimizing the sound experience for a single listener may be inappropriate, as this may lead to a deterioration of a sound experience for all other listeners. In this case, the position information may refer to a combination of positions from the different listeners, i.e. a combination of angles and a combination of distances of all listeners or a subset of the listeners. Without limitation, the combination may refer to a weighted average. The weights may be based on the distances and the angles of a respective listeners, i.e., the weights may decrease with the distance from the audio source and angle with respect to the main acoustic axis of the audio source. Thereby, listeners with sub-optimal positioning with respect to a position with an optimal sound experience may be weighted less when determining the average position. Further, the audio source location or the position (e.g. the listener position) may change constantly. In this case, a (relative) position or the corresponding position information may have to be updated continuously. In step S302, a sound level of the audio source may be estimated at the position. The sound level is understood as referring to a measurable sound level and / or a perceived sound level. More generally, the estimated sound level may need to be at least suitable for adapting the sound characteristics (e.g. loudness and / or directivity) of an audio source such that an audio experience can be improved based in the estimated sound level. To estimate the sound level of the audio source at the position, the attenuation of the sound emitted by the audio source may need to be modeled. In a reverberant environment, the sound attenuation may be modeled by a model for direct sound (direct sound field) and a model for reverberant sound (reverberant sound field). Therefore, the sound level is estimated based on a direct sound field model and a reverberant sound field model. The sound level may be determined based on a sum of the sound level determined based on the direct sound field model and the sound level determined based on the reverberant sound field model. More generally, the (overall) sound level may be determined based on a p-norm of a vector with the sound level according to the direct sound field model and reverberant sound field model as entries: ^= (B^^ CD + B^ CD)^ / D^ 2^Wherein ^ may be the estimated sound level, ^^^may be the sound level according to the direct sound field model and ^2^may be the sound level according to the reverberant sound fieldmodel. F may be a correlation factor and larger or equal to 1. Preferably F = 1.The direct sound field model estimates the behavior of the direct sound in the reverberant environment. In contrast to the prior art, the direct sound field model may be dependent on both a distance of a position with respect to the audio source and an angle of the position with respect to the main acoustic axis of the audio source. In the following, specific examples for direct sound field models will be provided. The specific direct sound field models should however not be construed as to limit the scope of the disclosure. As long as a direct sound model considers both directivity and distance, the direct sound field model may be used for estimating the sound level in this disclosure. The direct sound field model may be proportional to the power spectrum of the audio source. The power spectrum may be based on directivity measurement (i.e. unit circle around the audio source) and / or simulation data provided by a loudspeaker manufacturer (e.g. ballon plot), a fitted model using a lower number of parameters (e.g. frequency dependent ellipsoidal fit), a physically motivated model of a piston in an infinite baffle. Alternatively, the power spectrum may be modeled based on a high shelf filter. The attenuation of the direct sound may also be highly dependent on the distance to the audio source. Therefore, the direct sound field model may also depend on a distance decay model. Further, the direct sound field model may also be frequency dependent, as a power spectrum of an audio source may vary over frequency. In the most general form, the sound level at a distance ^, angle ^ and a frequency ^ may be defined as: ^1^^(^, ^, ^) = :^W= ^^(^, ^)In this case, > 1 may be a decay. In a preferredimplementation, may bet set to 2, i.e., the distance loss is quadratic. Further, ^^(^, ^) may bethe frequency dependent power spectrum. When the power spectrum is modeled based on a high shelf filter, the power spectrum may be determined based on the power spectrum on the main acoustic axis and the high shelf filter: ^^(^, ^) = !"(#(^), ^)^^(^, ^ = 0)An angle equal to 0 may source. !"(#(^), ^) maydenote the high shelf filter and #(^) the angle dependent gain in decibels of the high shelf filter. The angle dependent gain may be defined as: ì0 &^ ^ < ^^^ ^^0−#()*may specify a may specify angles between which the gain decreases linearly from 0 to -#()*. In summary, the direct sound field model is dependent on both angle and distance and therefore can capture the attenuation of a direct sound part accurately at a specific position with respect to an audio source. In contrast to the direct sound field model, a model for the reverberant sound field may be completely independent of an angle and a distance specifying the position in the reverberant environment. To be more precise, the reverberant sound field model may be completely independent of an angle, and may be dependent on a distance only indirectly via the definition of a critical distance. In other words, the reverberant sound field model may depend on a critical distance and the critical distance may be defined as the distance at which the sound level of the direct sound field model and the reverberant sound field model are equal. The critical distance may also be frequency dependent. As with the direct sound field model, the specific reverberant sound field models provided in the following should not be understood as a limitation to the disclosure. Any reverberant sound field model suitable for modeling a reverberant component of a sound field of an audio source may be used to estimate the sound level of the audio source in this disclosure. In a specific example, the sound level according to the reverberant sound field model may be defined as: ^2^(^) = ^^)34(^) :1(^)=6](^) may be the critical of the power spectrum asdefined for the direct sound field model. To be more precise, the average of the power spectrum may be calculated over an angle range, e.g. the full sphere. This averaging process may be implemented by a high shelf filter with an average gain, wherein the average gain may be calculated by averaging the gain of the high shelf filter as defined for the direct sound field model over an angle range, e.g. the full sphere. The average power spectrum can then be defined as: ^^)34(^) = ^^(^, ^ = 0)!"8#)34, ^9#)34 may be the average ^^(^, ^ = 0) may be thepower spectrum on the main acoustic axis. The reverberant sound field model may be generalized to be at least slightly dependent on the distance. In this case, the sound level according to the reverberant sound field model may be defined as: 1^ ` > ≥ 0 may be a variable for defining the distance dependent decay. Notably, α may be muchsmaller than the distance decay variable for the direct sound field model. Therefore, α may be larger than zero, but close to zero if is equal to 2. The reverberant sound field model may be even more generalized to also be dependent on the angle of the position. In this case, the sound level according to the reverberant sound field model may be defined as: 2^( A) = !"(#^( 1^ ^, ^, ^), ^)^^)34(^) ^ (0+_) `#^(^) may define ^ Notably, # (^)may be defined as ì0 &^ ^ < ^^ï^ − ^^ ^0−#()*′ may be the filter and may be much larger compared to the maximum negative gain of the high shelf filter for the direct sound field model. In other words, the reverberant sound field model may be always less dependent on the angle of the position compared to the direct sound field model. Instead of the average power spectrum, the reverberant sound field model may also be defined based on a flat power spectrum, i.e., a power spectrum that does not change with frequency. Alternatively, the reverberant sound field model may be based on a high-frequency roll-off power spectrum, i.e. the power spectrum may decay over frequency by a predefined rule, e.g., exponential decay, starting at a certain frequency. For further improving the estimation of a sound level of an audio source at a specific position, parameters of the direct sound field model and the reverberant sound field model may be tuned, e.g. the decay parameter(s) and / or the parameters of the high shelf filter. The parameters may be tuned based on the reverberant environment, i.e., when the parameters of the reverberant environment are unknown. The parameters of the reverberant environment may be unknown if the reverberant environment is a real environment (not a virtual environment), and no previous measurements are available. Fig.3 illustrates an example of a method 400 for tuning parameters of the of the direct sound field model and the reverberant sound field model. In step S401, an impulse response may be measured or simulated at one or more positions in the reverberant environment to tune the parameters of the direct and reverberant sound field model. An impulse response may be understood as a response at a specific position when the audio source, e.g. a loudspeaker, is fed with an impulse, e.g. a Dirac impulse. The response may be measured by using a microphone at the position. If the impulse response is simulated, a room model may be generated based on sensor data. Any suitable sensor data may be used for generating the room model, e.g., camera sensors and proximity sensor. The positions for measurement or simulation may be chosen to capture a large variety of responses. An example for different positions is depicted in Fig.4, in which three positions with nearly constant distance but varying angle are used for measurement or simulation, e.g.0°, 50° and 100° with respect to the main acoustic axis of a loudspeaker. Ideally, the positions include a position that is close or identical to a position for which the sound level of the audio source is to be estimated according to method 300. Before tuning, the impulse response may be divided in a first and second part, corresponding to a direct sound portion and a reverberant sound portion, respectively. The impulse response may be dived at a truncation time ^. Truncation time ^ may be a fixed value or may also be dependent on the environment. The division of the impulse response may be implemented by multiplying a window function with the impulse response: IRDS(^) = IR(^) H(^), where a H(^) > 0 &^ ^ < ^H = ^ ≥ ^^^IR(^)may be the ,(^)and IRRS(^)the first and second part of the impulse response, respectively. The direct sound field model may then be tuned based on the direct sound portion and the reverberant sound field model based on the reverberant sound portion. Using a small and fixed truncation time may have the inconvenience that frequencies approximately lower than the inverse truncation time cannot be adequately represented. Especially, when using unequal window lengths for IRDS and IRRS, the corresponding power or magnitude spectra may be too distorted to compare, are impacted by their respective windowing function. Therefore, the impulse response may alternatively be divided based on a frequency-dependent truncation kernel. The frequency-dependent truncation kernel may truncate all frequency components of the impulse response to a time ^ or smaller. It may truncate the lowest frequency under consideration to a time ^ and higher frequencies to a time smaller than ^. This approach may have the advantage of providing a better representation of the lower frequencies without compromising the truncation of the impulse response at higher frequencies. The corresponding first and second parts of the impulse response may then be defined as: ^IRDS(^) = b FDT(^, ^^) IR(^′) ^^′^^IRRS(^) = IR(^) − b FDT(^, ^^) IR(^′) ^^′^FDT(^, ^^) may respresent the trunction kernel.To illustrate the division, Fig.5 illustrated an example for dividing the measured or simulated impulse response into the first part (direct sound response) and the second part (reverberantresidual sound response). Notably, the direct sound does not start at ^ = 0, as the sound wavespropagate through the air with the speed of sound, thereby inducing a delay based on the distance of the position at which the impulse response is measured or simulated. This delay has to be considered when dividing the impulse response. Further, Figs.6A, 6B, and 6C illustrate the impulse response in the frequency domain, for all three positions of Fig.4, and for the full response (Fig.6A), the direct sound portion (Fig.6B), and the reverberant sound portion (Fig.6C). Notably, while the reverberant sound portion is nearly independent of the angle, the direct sound portion shows a strong attenuation for high frequencies and large off-axis angle, i.e.100°. Therefore, the gain and the two angles of the high shelf filter for the direct sound field model may modified to captures this effect. It is however noted that Figs.6A, 6B, and 6C depict the impulse response for an example for a specific environment, which is not to be construed to limit this disclosure. Impulse responses for different environments may have different properties when the angle or distance is varied with respect to the audio source. Finally, when impulse responses are determined for the environment, e.g. for a room, parameters of the direct sound field model and the reverberant sound field model may be tuned based on the impulse responses in step S402. As previously mentioned, this may include all parameters of the models that are not fixed, e.g., the gain of the high shelf filter. By estimating the sound level at a position based on a tuned model, the estimated sound level may be used to improve the sound experience for a listener, especially at an off-axis positions, e.g., an angle larger than 50° of the main acoustic axis of a loudspeaker. After estimating the sound level based on direct sound field model and the reverberant sound field model, method 300 outputs the estimated sound level for further processing in step S303. In the following, two particular use cases for the estimated sound level will be presented. Method 300 should however not be construed to be limited to these two specific use cases. Other use cases for the estimated sound level may be envisioned by the skilled person. The first use case for the estimated sound level may be for a real environment, e.g. a room, with a sound system comprising one or multiple loudspeakers. In such an environment, the signal fed to a loudspeaker may be adapted for a position of a listener. For example, the estimated sound level could be used for constantly adapting the signal fed to a loudspeaker such that the sound level or the sound quality experienced by a moving listener is constant or close to constat. Further, in a case of a sound system with multiple loudspeakers, a sound level from one loudspeaker may be a reference sound level for the remaining loudspeakers such that all loudspeakers have the same loudness at the position of a listener. To achieve this, a compensation filter is determined based on the estimated sound level and a target sound level in step 304a. The target sound level is used for matching the sound level of the different loudspeakers. The target sound level may be a sound level of one of the loudspeakers. A response of the compensation filter may be defined as: ^(^)^^(^) =^^(^, ^, ^) may be the estimated method 300 and ^(^) may be thetarget sound level. To implement the compensation filter, ^(^, ^, ^) may be smoothed beforedetermined the compensation filter or regularization may be used when determined the response of the compensation filter. The compensation filter may be implemented as an FIR filter, an IIR filter, or a transform-based filter. In step S304b the compensation filter is applied to the signal fed to a loudspeaker, e.g. the input signal of the loudspeaker may be multiplied by the compensation filter in the frequency domain. When one loudspeaker of the multiple loudspeakers represents the reference sound level, the compensation filter may only need to be applied to all loudspeakers but the loudspeaker corresponding to the reference sound level. By tuning the sound emitted by the loudspeakers based on a sound level estimated at a position of a listener, a sound experience of the listener can be improved, specifically as the underlying model for estimating the sound level is based on both angle and distance of the position. An alternative use case for the estimated sound level may be a virtual or augmented reality environment. In such an environment, different audio sources may be rendered. As the position of a listener, i.e., the user of virtual or augmented reality headset, may be exactly known, the sound level of an audio source may be estimated continuously. The estimation of the sound level at the position may be very efficient way of producing a realistic sound scene for the user, compared with simulating the sound level based on sound wave propagation in the virtual environment. Therefore, in this use case, in step S304c a virtual audio source in the virtual or augmented reality environment is rendered by the virtual or augmented reality device with the estimated sound level. The sound for the virtual environment may be reproduced by a headset that is part of the virtual or augmented reality device or that is connected to a virtual or augmented reality device. Alternatively, sound for the virtual environment may be reproduced by loudspeakers connected to the virtual or augmented reality device. Alternative method of sound level estimation Instead of considering a direct sound level model that is dependent on both distance and angle of the position (i.e. instead of step S302), a direct sound level model that is only dependent on the angle may be used in combination with a distance dependent weighting factor. In this case, the estimated sound level may be expressed as: ^(^, ^, ^) = HI&Jℎ^^^(^) ∗ ^^^(^, ^) + ^2^(^, ^, ^)In this example ^^^ of the audiosource. Further, HI&Jℎ^^^(^) may be equal to 1 when ^ is smaller than a minimal distance, e.g. 0,5m. Further, HI&Jℎ^^^(^) may be 0 when the distance is equal to the critical distance as previously defined for the revereberant sound field model. Between the minimimal distance and the critical distance HI&Jℎ^^^(^)may decrease linearly. Notably, the reverberant sound fieldmay also be independent of the distance and angle in this example. i.e. ^2^(^, ^, ^) = ^2^(^).Further, instead of modelling a distance dependent reverberant sound field, a weighting factor may also be used for the reverberant sound field. In this case the estimated sound level may be expressed as: ^(^, ^, ^) = HI&Jℎ^^^(^) ∗ ^^^(^, ^) + HI&Jℎ^2^(^) ∗ ^2^(^, ^)In this example, HI&Jℎ^2^(^) = 1 − HI&Jℎ^^^(^).By using this alternative method, determination / estimation of the overall sound level at a certain position may be simplified, while accuracy of the sound level estimation may decrease compared to method 300. Further, accuracy of an existing implementation with distance independent directivity compensation may be improved by adding the distance dependent weight. While a method of estimating a sound level has been described above, the disclosure likewise relates to corresponding apparatus, and the like. An embodiment providing such apparatus will be described next with reference to Fig.7. As shown in Fig.7, the apparatus 800 includes a processor 801 and memory 802. The memory 802 is configured to store program code. The processor 801 is configured to run instructions in the program code, so that the apparatus 800 performs the sound level estimation methods in any one of the above embodiments and implementations. The processor 801 may also receive, among others, suitable input data (e.g., position information, tuning parameters, etc.), depending on use cases and / or implementations. The processor 801 may be adapted to carry out the methods / techniques (e.g., methods 300 and 400 as illustrated above with reference to Figs.3 and 4, respectively) described throughout the present disclosure and to generate corresponding output data (e.g., the estimated sound level, etc.), depending on use cases and / or implementations. The apparatus may be part of a Virtual reality (VR), augmented reality (AR), mixed reality (MR), and / or extended reality (XR) device. Alternatively, the apparatus may be part of a loudspeaker or connected to a loudspeaker or a loudspeaker system. A device including apparatus 800 may also process a received bitstream. The bitstream may be processed by a decoder. The bitstream may include audio data and / or metadata for use during rendering by apparatus 800. One or more of the components, blocks, processes or other functional components may be implemented through a computer program that controls execution of a processor-based computing device of the system. It should also be noted that the various functions disclosed herein may be described using any number of combinations of hardware, firmware, and / or as data and / or instructions embodied in various machine-readable or computer-readable media, in terms of their behavioral, register transfer, logic component, and / or other characteristics. Computer-readable media in which such formatted data and / or instructions may be embodied include, but are not limited to, physical (non-transitory), non-volatile storage media in various forms, such as optical, magnetic or semiconductor storage media. While one or more implementations have been described by way of example and in terms of the specific embodiments, it is to be understood that one or more implementations are not limited to the disclosed embodiments. To the contrary, it is intended to cover various modifications and similar arrangements as would be apparent to those skilled in the art. Therefore, the scope of the appended claims should be accorded the broadest interpretation so as to encompass all such modifications and similar arrangements. Interpretation A computing device implementing the techniques described above can have the following example architecture. Other architectures are possible, including architectures with more or fewer components. In some implementations, the example architecture includes one or more processors (e.g., dual-core Intel® Xeon® Processors), one or more output devices (e.g., LCD), one or more network interfaces, one or more input devices (e.g., mouse, keyboard, touch- sensitive display) and one or more computer-readable mediums (e.g., RAM, ROM, SDRAM, hard disk, optical disk, flash memory, etc.). These components can exchange communications and data over one or more communication channels (e.g., buses), which can utilize various hardware and software for facilitating the transfer of data and control signals between components. The term “computer-readable medium” refers to a medium that participates in providing instructions to processor for execution, including without limitation, non-volatile media (e.g., optical or magnetic disks), volatile media (e.g., memory) and transmission media. Transmission media includes, without limitation, coaxial cables, copper wire and fiber optics. Computer-readable medium can further include operating system (e.g., a Linux® operating system), network communication module, audio interface manager, audio processing manager and live content distributor. Operating system can be multi-user, multiprocessing, multitasking, multithreading, real time, etc. Operating system performs basic tasks, including but not limited to: recognizing input from and providing output to network interfaces and / or devices; keeping track and managing files and directories on computer-readable mediums (e.g., memory or a storage device); controlling peripheral devices; and managing traffic on the one or more communication channels. Network communications module includes various components for establishing and maintaining network connections (e.g., software for implementing communication protocols, such as TCP / IP, HTTP, etc.). Architecture can be implemented in a parallel processing or peer-to-peer infrastructure or on a single device with one or more processors. Software can include multiple software components or can be a single body of code. The described features can be implemented advantageously in one or more computer programs that are executable on a programmable system including at least one programmable processor coupled to receive data and instructions from, and to transmit data and instructions to, a data storage system, at least one input device, and at least one output device. A computer program is a set of instructions that can be used, directly or indirectly, in a computer to perform a certain activity or bring about a certain result. A computer program can be written in any form of programming language (e.g., Objective-C, Java), including compiled or interpreted languages, and it can be deployed in any form, including as a stand-alone program or as a module, component, subroutine, a browser-based web application, or other unit suitable for use in a computing environment. Suitable processors for the execution of a program of instructions include, by way of example, both general and special purpose microprocessors, and the sole processor or one of multiple processors or cores, of any kind of computer. Generally, a processor will receive instructions and data from a read-only memory or a random-access memory or both. The essential elements of a computer are a processor for executing instructions and one or more memories for storing instructions and data. Generally, a computer will also include, or be operatively coupled to communicate with, one or more mass storage devices for storing data files; such devices include magnetic disks, such as internal hard disks and removable disks; magneto-optical disks; and optical disks. Storage devices suitable for tangibly embodying computer program instructions and data include all forms of non-volatile memory, including by way of example semiconductor memory devices, such as EPROM, EEPROM, and flash memory devices; magnetic disks such as internal hard disks and removable disks; magneto-optical disks; and CD-ROM and DVD-ROM disks. The processor and the memory can be supplemented by, or incorporated in, ASICs (application-specific integrated circuits). To provide for interaction with a user, the features can be implemented on a computer having a display device such as a CRT (cathode ray tube) or LCD (liquid crystal display) monitor or a retina display device for displaying information to the user. The computer can have a touch surface input device (e.g., a touch screen) or a keyboard and a pointing device such as a mouse or a trackball by which the user can provide input to the computer. The computer can have a voice input device for receiving voice commands from the user. The features can be implemented in a computer system that includes a back-end component, such as a data server, or that includes a middleware component, such as an application server or an Internet server, or that includes a front-end component, such as a client computer having a graphical user interface or an Internet browser, or any combination of them. The components of the system can be connected by any form or medium of digital data communication such as a communication network. Examples of communication networks include, e.g., a LAN, a WAN, and the computers and networks forming the Internet. The computing system can include clients and servers. A client and server are generally remote from each other and typically interact through a communication network. The relationship of client and server arises by virtue of computer programs running on the respective computers and having a client-server relationship to each other. In some embodiments, a server transmits data (e.g., an HTML page) to a client device (e.g., for purposes of displaying data to and receiving user input from a user interacting with the client device). Data generated at the client device (e.g., a result of the user interaction) can be received from the client device at the server. A system of one or more computers can be configured to perform particular actions by virtue of having software, firmware, hardware, or a combination of them installed on the system that in operation causes or cause the system to perform the actions. One or more computer programs can be configured to perform particular actions by virtue of including instructions that, when executed by data processing apparatus, cause the apparatus to perform the actions. While this specification contains many specific implementation details, these should not be construed as limitations on the scope of any inventions or of what may be claimed, but rather as descriptions of features specific to particular embodiments of particular inventions. Certain features that are described in this specification in the context of separate embodiments can also be implemented in combination in a single embodiment. Conversely, various features that are described in the context of a single embodiment can also be implemented in multiple embodiments separately or in any suitable sub-combination. Moreover, although features may be described above as acting in certain combinations and even initially claimed as such, one or more features from a claimed combination can in some cases be excised from the combination, and the claimed combination may be directed to a sub-combination or variation of a sub-combination. Similarly, while operations are depicted in the drawings in a particular order, this should not be understood as requiring that such operations be performed in the particular order shown or in sequential order, or that all illustrated operations be performed, to achieve desirable results. In certain circumstances, multitasking and parallel processing may be advantageous. Moreover, the separation of various system components in the embodiments described above should not be understood as requiring such separation in all embodiments, and it should be understood that the described program components and systems can generally be integrated together in a single software product or packaged into multiple software products. Unless specifically stated otherwise, as apparent from the following discussions, it is appreciated that throughout the present invention discussions utilizing terms such as “processing”, “computing”, “calculating”, “determining”, “analyzing” or the like, refer to the action and / or processes of a computer or computing system, or similar electronic computing devices, that manipulate and / or transform data represented as physical, such as electronic, quantities into other data similarly represented as physical quantities. Reference throughout this invention to “one example embodiment”, “some example embodiments” or “an example embodiment” means that a particular feature, structure or characteristic described in connection with the example embodiment is included in at least one example embodiment of the present invention. Thus, appearances of the phrases “in one example embodiment”, “in some example embodiments” or “in an example embodiment” in various places throughout this invention are not necessarily all referring to the same example embodiment. Furthermore, the particular features, structures or characteristics may be combined in any suitable manner, as would be apparent to one of ordinary skill in the art from this invention, in one or more example embodiments. As used herein, unless otherwise specified the use of the ordinal adjectives “first”, “second”, “third”, etc., to describe a common object, merely indicate that different instances of like objects are being referred to and are not intended to imply that the objects so described must be in a given sequence, either temporally, spatially, in ranking, or in any other manner. Also, it is to be understood that the phraseology and terminology used herein are for the purpose of description and should not be regarded as limiting. The use of “including,” “comprising,” or “having” and variations thereof are meant to encompass the items listed thereafter and equivalents thereof as well as additional items. Unless specified or limited otherwise, the terms “mounted”, “connected”, “supported”, and “coupled” and variations thereof are used broadly and encompass both direct and indirect mountings, connections, supports, and couplings. In the claims below and the description herein, any one of the terms comprising, comprised of or which comprises is an open term that means including at least the elements / features that follow, but not excluding others. Thus, the term comprising, when used in the claims, should not be interpreted as being limitative to the means or elements or steps listed thereafter. For example, the scope of the expression a device comprising A and B should not be limited to devices consisting only of elements A and B. Any one of the terms including or which includes or that includes as used herein is also an open term that also means including at least the elements / features that follow the term, but not excluding others. Thus, including is synonymous with and means comprising. It should be appreciated that in the above description of example embodiments of the present invention, various features of the present invention are sometimes grouped together in a single example embodiment, Fig., or description thereof for the purpose of streamlining the present invention and aiding in the understanding of one or more of the various inventive aspects. This method of invention, however, is not to be interpreted as reflecting an intention that the claims require more features than are expressly recited in each claim. Rather, as the following claims reflect, inventive aspects lie in less than all features of a single foregoing disclosed example embodiment. Thus, the claims following the Description are hereby expressly incorporated into this Description, with each claim standing on its own as a separate example embodiment of this invention. Furthermore, while some example embodiments described herein include some, but not other features included in other example embodiments, combinations of features of different example embodiments are meant to be within the scope of the present invention, and form different example embodiments, as would be understood by those skilled in the art. For example, in the following claims, any of the claimed example embodiments can be used in any combination. In the description provided herein, numerous specific details are set forth. However, it is understood that example embodiments of the present invention may be practiced without these specific details. In other instances, well-known methods, structures and techniques have not been shown in detail in order not to obscure an understanding of this description. Thus, while there has been described what are believed to be the best modes of the present invention, those skilled in the art will recognize that other and further modifications may be made thereto without departing from the spirit of the present invention, and it is intended to claim all such changes and modifications as fall within the scope of the present invention. For example, any formulas given above are merely representative of procedures that may be used. Functionality may be added or deleted from the block diagrams and operations may be interchanged among functional blocks. Steps may be added or deleted to methods described within the scope of the present disclosure. Enumerated Example Embodiments Various aspects and implementations of the present disclosure may also be appreciated from the following enumerated example embodiments (EEEs), which are not claims. EEE 1. A method of estimating a sound level of an audio source at a position in a reverberant environment, the method comprising: obtaining position information for the position, wherein the position information comprises a distance between the position and the audio source, and an angle of the position with respect to a main acoustic axis of the audio source; estimating the sound level of the audio source at the position based on a reverberant sound field model of the audio source and a direct sound field model of the audio source, wherein the direct sound field model is based on both the distance and the angle in the position information; outputting the estimated sound level for further processing. EEE 2. The method according to EEE 1, wherein the reverberant environment is a real environment, and the audio source is a loudspeaker. EEE 3. The method according to EEE 2, wherein outputting the estimated sound level for further processing comprises: determining a compensation filter based on the estimated sound level and a target sound level; and applying the compensation filter to a signal fed to the loudspeaker. EEE 4. The method according to EEE 3, wherein a desired response of the compensation filter is defined as ^^(^) = ^(^)^where ^(^, ^, ^) is the estimated target sound level, ^ is frequency, d isthe distance and ^ is the angle.EEE 5. The method according to EEE 4, wherein a smoothing function is applied to ^(^, ^, ^)and / or wherein regularization is used for determining the desired response of the compensation filter. EEE 6. The method according to EEE 4 or 5, wherein the target sound level is used for matching the sound level of the loudspeaker and a sound level of at least one second loudspeaker at the position. EEE 7. The method according to any one of EEEs 3 to 6, wherein the compensation filter is implemented as any one of an FIR filter, an IIR filter, or a transform-based filter. EEE 8. The method according to any one of EEEs 2 to 7, wherein obtaining the position information comprises receiving the position information or determining the position information based on measuring or estimating the position. EEE 9. The method according to any one of the previous EEEs, wherein the position is a listener position of a listener, and the position information is listener position information. EEE 10. The method according to EEE 9 when dependent on EEE 8, wherein the listener position information is determined based on any combination of camera sensor data, proximity sensor data, acoustic sensor data, inertial measurement unit data, virtual reality headset data and radio sensor data. EEE 11. The method according to EEE 9 when dependent on EEE 8, or EEE 10, wherein the listener position corresponds to a combination of multiple listener positions if at least one or more second listeners are in the reverberant environment. EEE 12. The method according to EEE 11, wherein the combination of multiple listener positions comprises the combination of the distance and the distances between each second listener and the audio source and the combination of the angle and the angles between each second listener and the main acoustic axis of the audio source. EEE 13. The method according to EEE 11 or 12, wherein the combination of the multiple listener positions is a weighted average of the multiple listener positions. EEE 14. The method according to any one of EEEs 2 to 13, wherein the method further comprises: tuning parameters for the reverberant sound field model and the direct sound field model based on the reverberant environment. EEE 15. The method according to EEE 14, wherein tuning the parameters for the reverberant sound field model and the direct sound field model based on the reverberant environment comprises: measuring or simulating an impulse response of the audio source at one or more second positions in the reverberant environment; and tuning the parameters based on the measured or simulated impulse response. EEE 16. The method according EEE 15, wherein the one or more second positions comprise the position. EEE 17. The method according to EEE 15 or 16, wherein before tuning the parameters based on the impulse response, the method further comprises: dividing the impulse response into a first part and a second part at truncation time ^, wherein the first part corresponds to a direct sound portion and the second part corresponds to a reverberant sound portion. EEE 18. The method according to EEE 15 or 16, wherein before tuning the parameters based on the impulse response, the method further comprises: dividing the impulse response into a first part and a second part based on a frequency-dependent truncation kernel, wherein the first part corresponds to a direct sound portion and the second part corresponds to a reverberant sound portion. EEE 19. The method according to EEE 18, wherein the first part is defined as ^IRDS(^) = b FDT(^, ^^) IR(^′) ^^′^and the second part is defined as ^ IRRS(^) = IR(^) − b FDT(^, ^^) IR(^′) ^^′^where FDT(^, ^^) is the frequency-dependent truncation kernel, IR(^′) is the impulse responseand ^ is a time constant. EEE 20. The method according to EEE 1, wherein the reverberant environment is a virtual reality environment or an augmented reality environment, and the audio source is a virtual audio source. EEE 21. The method according to EEE 20, wherein outputting the estimated sound level for further processing comprises: rendering the virtual audio source with the estimated sound level by a virtual reality device used for rendering the virtual environment or the augmented reality environment. EEE 22. The method according to EEE 21, wherein a sound of the rendered virtual audio source is output by a headphone of the virtual reality device or a headphone connected to the virtual reality device. EEE 23. The method according to any one of the previous EEEs, wherein the sound level is proportional to a power spectrum of the audio source. EEE 24. The method according to any one of the previous EEEs, wherein the direct sound field model of the audio source is frequency dependent. EEE 25. The method according to EEE 24 when dependent on EEE 23, wherein the direct sound field model of the audio source is based on a distance decay model and the power spectrum of the audio source. EEE 26. The method according to EEE 25, wherein the direct sound field model of the audio source is defined as ^^^ = 1^^ ^) wherein ^^^(^, ^, ^) is a level of the direct sound field model, ^ is the frequency, ^ is thedistance, is a decay parameter of the distance decay model and larger than 0, ^ is the angle, and ^^(^, ^) is the power spectrum of the audio source.method according to EEE 26, wherein = 2. EEE 28. The method according to any one of EEEs 25 to 27, wherein the power spectrum is determined based on measurement data or simulation data of the audio source, or the power spectrum is based on data from a database. EEE 29. The method according to any one of EEEs 25 to 27, wherein the power spectrum is determined based on the power spectrum on the main acoustic axis of the audio source and a high shelf filter model. EEE 30. The method according to EEE 29, wherein the power spectrum is defined as^^(^, ^) = !"(#(^), ^)^^(^, ^ = 0)where ^^(^, ^ = 0) is the power spectrum on the main acoustic axis of the audio source,!"(#(^), ^) is the high shelf filter model, and #(^) is an angle dependent gain in decibels of thehigh shelf filter model. EEE 31. The method according to EEE 30, wherein the angle dependent gain of the high shelf filter model is defined as 0&^ ^ < ^^where of the high shelf filter model, ^^is a first threshold angle, and ^0is a second threshold angle. EEE 32. The according to any one of the previous EEEs, wherein the reverberant sound field model of the audio source is independent of both the distance and the angle. EEE 33. The method according to any one of EEEs 1 to 31, wherein the reverberant sound field model of the audio source is frequency dependent. EEE 34. The method according to EEE 33, wherein the reverberant sound field model of the audio source depends on a critical distance. EEE 35. The method according to EEE 34, wherein the critical distance is a distance from the audio source at which a level of the direct sound field model and a level of the reverberant sound field model are equal. EEE 36. The method according to EEE 34 or 35, wherein the reverberant sound field model of the audio source is defined as ^(^) = ^^ 12^ )34(^) :6 =70(^)wherein ^2^(^)is a level of the ^ is the frequency, 67is the critical distance, and ^^ )34(^) of the audio source. EEE 37. The method according to EEE 36 when depending on any one of EEEs 25 to 31, wherein the average power spectrum is an average of the power spectrum of the audio source over an angle range. EEE 38. The method according to EEE 37, wherein the angle range corresponds to the full sphere. EEE 39. The method according to any one of EEEs 36 to 38 when depending on EEE 30 or 31,wherein the average power spectrum is defined as^^)34(^) = ^^(^, ^ = 0)!"8#)34, ^9where #)34is an average EEE 40. The method according to any one of EEE 33 to 35, wherein the reverberant sound field model of the audio source further depends on the distance between the position and the audio source. EEE 41. The method according to EEE 40 when depending on EEE 26 or 27, wherein the reverberant sound field model is defined as ^^2^(^, ^) = ^^)34(^) :^;^(-<;) =,where ^2^(^, ^) is a level of 67 is the critical distance,^^)34(^) is an average power spectrum of the audio source and α is a decay parameter with 0 ≤> ≤ 2 and > ≪ .EEE 42. The method according to EEE 41, wherein the reverberant sound field model of the audio source further depends on the angle. EEE 43. The method according to EEE 42 when dependent on EEE 31, wherein the reverberant sound field model is defined as 1^ ` where ^2^(^, ^, A) is a level of the reverberant sound field model, 67 is the critical distance,^^)34(^) is an average power spectrum of the audio source, α is a decay parameter with 0 ≤> ≤ 2 and > ≪ , and #^(^) is the angle dependent gain in decibels of the high shelf filtermodel with a maximum gain #′()* ≪ #()*.EEE 44. The method according to any one of EEE 1 to 31 and 33 to 43, wherein estimating the sound level of the audio source at the position comprises determining the sound level of theaudio source as^(^, ^, ^) = (B^^^(^, ^, ^)CD + B^2^(^, ^, ^)CD)^ / D,where ^2^(^, ^, A) is ^^^ ^, ^) is a level of the direct sound field and F is a coherence tuning parameter.EEE 45. The method according to EEE 44, wherein F = 1.EEE 46. The method according to any one of the previous EEEs, wherein the sound level of the audio source is a measurable loudness or a perceived loudness. EEE 47. The method according to any one of the previous EEEs, wherein the audio source is moving and / or the position is changing. EEE 48. The method according to any one of the previous EEEs, wherein the reverberant environment comprises objects that reflect and diffuse sound emitted by the audio source. EEE 49. The method according to any one of the previous EEEs, wherein the position information is updated continuously. EEE 50. The method according to any one of the previous EEEs, wherein the main acoustic axis of the audio source is a direction of a maximum sound pressure level emitted by the audio source. EEE 51. A method of estimating a sound level of an audio source at a position in a reverberant environment, the method comprising: obtaining position information for the position, wherein the position information comprises a distance between the position and the audio source, and an angle of the position with respect to a main acoustic axis of the audio source; estimating the sound level of the audio source at the position based on a weighted sum of a reverberant sound field model of the audio source and direct sound field model of the audio source, wherein the direct sound field model is based on the angle and a weight of the weighted sum depends on the distance; outputting the estimated sound level for further processing. EEE 52. The method according to EEE 51, wherein estimating the sound level of the audiosource at the position comprises determining the sound level of the audio source as^(^, ^, ^) = HI&Jℎ^^^(^) ∗ ^^^(^, ^) + ^2^(^, ^, ^)where ^2^(^, ^, A) is a level of the reverberant sound field model, ^^^(^, ^) is a level ofthe direct sound ^ is the distance, and ^ is the angle,. EEE 53. The method according to EEE 51, wherein estimating the sound level of the audiosource at the position comprises determining the sound level of the audio source as^(^, ^, ^) = HI&Jℎ^^^(^) ∗ ^^^(^, ^) + HI&Jℎ^2^(^) ∗ ^2^(^, ^)where ^2^(^, A) is a level of the reverberant sound field model, ^^^(^, ^) is a level of thedirect sound field model, HI&Jℎ^^^(^) is a first weight of the weighted sum, HI&Jℎ^2^(^) is asecond weight of the weighted sum with HI&Jℎ^2^(^) = 1 − HI&Jℎ^^^(^), ^ is frequency, ^ isthe distance, and ^ is the angle. EEE 54. The method according to EEE 52 or 53, wherein HI&Jℎ^^^(^) is equal to 1 when ^ issmaller than a minimal distance;HI&Jℎ^^^(^) is 0 when the distance is equal to a critical distance; andHI&Jℎ^^^(^) decreases linearly between the minimal distance and the critical distance.EEE 55. The method according to any one of EEEs 52 to 54, wherein the reverberant sound field is independent of the angle and the frequency. EEE 56. An apparatus, comprising a processor and a memory coupled to the processor, wherein the processor is adapted to carry out the method according to any one of EEEs 1 to 55. EEE 57. The apparatus according to EEE 56, wherein the apparatus is a loudspeaker or a device connected to the loudspeaker. EEE 58. The apparatus according to EEE 56, wherein the apparatus is a virtual reality headset, an augmented reality headset or a device connected to the virtual reality headset or the augmented reality headset. EEE 59. A device comprising: a decoder for decoding a received bitstream; and the apparatus according to any one of EEEs 56 to 58; wherein the bitstream includes audio data and / or metadata for use during rendering by the apparatus. EEE 60. A computer program comprising instructions that, when executed by a processor, cause the processor to carry out the method according to any one of EEEs 1 to 55. EEE 61. A computer-readable storage medium storing the computer program according to EEE 59.

Claims

1. CLAIMS 1. A method of estimating a sound level of an audio source at a position in a reverberant environment, the method comprising: obtaining position information for the position, wherein the position information comprises a distance between the position and the audio source, and an angle of the position with respect to a main acoustic axis of the audio source; estimating the sound level of the audio source at the position based on a reverberant sound field model of the audio source and a direct sound field model of the audio source, wherein the direct sound field model is based on both the distance and the angle in the position information; outputting the estimated sound level for further processing.

2. The method according to claim 1, wherein the reverberant environment is a real environment, and the audio source is a loudspeaker.

3. The method according to claim 2, wherein outputting the estimated sound level for further processing comprises: determining a compensation filter based on the estimated sound level and a target sound level; and applying the compensation filter to a signal fed to the loudspeaker.

4. The method according to claim 3, wherein a desired response of the compensation filter is defined as ^(^)^^(^) =^where ^(^, ^, ^) is theis the target sound level, ^ isfrequency, d is the distance and ^ is the angle.

5. The method according to claim 4, wherein a smoothing function is applied to ^(^, ^, ^)and / or wherein regularization is used for determining the desired response of the compensation filter.

6. The method according to claim 4 or 5, wherein the target sound level is used for matching the sound level of the loudspeaker and a sound level of at least one second loudspeaker at the position.

7. The method according to any one of claims 3 to 6, wherein the compensation filter is implemented as any one of an FIR filter, an IIR filter, or a transform-based filter.

8. The method according to any one of claims 2 to 7, wherein obtaining the position information comprises receiving the position information or determining the position information based on measuring or estimating the position.

9. The method according to any one of the previous claims, wherein the position is a listener position of a listener, and the position information is listener position information.

10. The method according to claim 9 when dependent on claim 8, wherein the listener position information is determined based on any combination of camera sensor data, proximity sensor data, acoustic sensor data, inertial measurement unit data, virtual reality headset data and radio sensor data.

11. The method according to claim 9 when dependent on claim 8, or claim 10, wherein the listener position corresponds to a combination of multiple listener positions if at least one or more second listeners are in the reverberant environment.

12. The method according to claim 11, wherein the combination of multiple listener positions comprises the combination of the distance and the distances between each second listener and the audio source and the combination of the angle and the angles between each second listener and the main acoustic axis of the audio source.

13. The method according to claim 11 or 12, wherein the combination of the multiple listener positions is a weighted average of the multiple listener positions.

14. The method according to any one of claims 2 to 13, wherein the method further comprises:tuning parameters for the reverberant sound field model and the direct sound field model based on the reverberant environment.

15. The method according to claim 14, wherein tuning the parameters for the reverberant sound field model and the direct sound field model based on the reverberant environment comprises: measuring or simulating an impulse response of the audio source at one or more second positions in the reverberant environment; and tuning the parameters based on the measured or simulated impulse response.

16. The method according claim 15, wherein the one or more second positions comprise the position.

17. The method according to claim 15 or 16, wherein before tuning the parameters based on the impulse response, the method further comprises: dividing the impulse response into a first part and a second part at truncation time ^, wherein the first part corresponds to a direct sound portion and the second part corresponds to a reverberant sound portion.

18. The method according to claim 15 or 16, wherein before tuning the parameters based on the impulse response, the method further comprises: dividing the impulse response into a first part and a second part based on a frequency- dependent truncation kernel, wherein the first part corresponds to a direct sound portion and the second part corresponds to a reverberant sound portion.

19. The method according to claim 18, wherein the first part is defined as ^IRDS(^) = b FDT(^, ^^) IR(^′) ^^′^and the second part is defined as ^IRRS(^) = IR(^) − b FDT(^, ^^) IR(^′) ^^′^where FDT(^, ^^) is the frequency-dependent truncation kernel, IR(^′) is the impulseresponse and ^ is a time constant.

20. The method according to claim 1, wherein the reverberant environment is a virtual reality environment or an augmented reality environment and the audio source is a virtual audio source.

21. The method according to claim 20, wherein outputting the estimated sound level for further processing comprises: rendering the virtual audio source with the estimated sound level by a virtual reality device used for rendering the virtual environment or the augmented reality environment.

22. The method according to claim 21, wherein a sound of the rendered virtual audio source is output by a headphone of the virtual reality device or a headphone connected to the virtual reality device.

23. The method according to any one of the previous claims, wherein the sound level is proportional to a power spectrum of the audio source.

24. The method according to any one of the previous claims, wherein the direct sound field model of the audio source is frequency dependent.

25. The method according to claim 24 when dependent on claim 23, wherein the direct sound field model of the audio source is based on a distance decay model and the power spectrum of the audio source.

26. The method according to claim 25, wherein the direct sound field model of the audio source is defined as ^(^, ^, ) 1^^ ^ = : ^^(^, ^)wherein ^^^(^, ^, ^) is amodel, ^ is the frequency, ^ is thedistance, is a decay parameter of the distance decay model and larger than 0, ^ is the angle,and ^^(^, ^) is the power spectrum of the audio source.

27. The method according to claim 26, wherein = 2.

28. The method according to any one of claims 25 to 27, wherein the power spectrum is determined based on measurement data or simulation data of the audio source, or the power spectrum is based on data from a database.

29. The method according to any one of claims 25 to 27, wherein the power spectrum is determined based on the power spectrum on the main acoustic axis of the audio source and a high shelf filter model.

30. The method according to claim 29, wherein the power spectrum is defined as^^(^, ^) = !"(#(^), ^)^^(^, ^ = 0)where ^^(^, ^ = 0) is the power spectrum on the main acoustic axis of the audio source,!"(#(^), ^) is the high shelf filter model, and #(^) is an angle dependent gain in decibels of thehigh shelf filter model.

31. The method according to claim 30, wherein the angle dependent gain of the high shelf filter model is defined as 0&^ ^ < ^^^+^^whereof the high shelf filter model, ^^is a first threshold angle, and ^0is a second threshold angle.

32. The method according to any one of the previous claims, wherein the reverberant sound field model of the audio source is independent of both the distance and the angle.

33. The method according to any one of claims 1 to 31, wherein the reverberant sound field model of the audio source is frequency dependent.

34. The method according to claim 33, wherein the reverberant sound field model of the audio source depends on a critical distance.

35. The method according to claim 34, wherein the critical distance is a distance from the audio source at which a level of the direct sound field model and a level of the reverberant sound field model are equal.

36. The method according to claim 34 or 35, wherein the reverberant sound field model of the audio source is defined as 2^( ) 1^ ^ = ^^)34(^) :6 =70(^)wherein ^2^(^) is a model, ^ is the frequency, 67is the critical distance, and ^^)34of the audio source.

37. The method according to claim 36 when depending on any one of claims 25 to 31, wherein the average power spectrum is an average of the power spectrum of the audio source over an angle range.

38. The method according to claim 37, wherein the angle range corresponds to the full sphere.

39. The method according to any one of claims 36 to 38 when depending on claim 30 or31, wherein the average power spectrum is defined as^^)34(^) = ^^(^, ^ = 0)!"8#)34, ^9where #)34is an average gain of #(^).

40. The method according to any one of claim 33 to 35, wherein the reverberant sound field model of the audio source further depends on the distance between the position and the audio source.

41. The method according to claim 40 when depending on claim 26 or 27, wherein the reverberant sound field model is defined as ^(^, ^) = ^2^ ^^)34(^) (-<;) =,sound field model, 67 is the critical distance,^^)34(^) is an average power spectrum of the audio source and α is a decay parameter with 0 ≤> ≤ 2 and > ≪ .

42. The method according to claim 41, wherein the reverberant sound field model of the audio source further depends on the angle.

43. The method according to claim 42 when dependent on claim 31, wherein the reverberant sound field model is defined as ^2^(^, ^, A) = !"(#^( 1^), ^)^^)34(^) ^ (0+ `^_6 _)where ^2^ 67 is the criticaldistance, ^^)34(^) a decay parameterwith 0 ≤ > ≤ 2 and > ≪ , and #^(^) is the angle dependent gain in decibels of the high shelffilter model with a maximum gain #′()* ≪ #()*.

44. The method according to any one of the previous claims, wherein estimating the sound level of the audio source at the position comprises determining the sound level of theaudio source as^(^, ^, ^) = (B^^^(^, ^, ^)CD + B^2^(^, ^, ^)CD)^ / Dwhere ^2^(^, ^, A) is a level of the reverberant sound field model, ^^^(^, ^, ^) is a levelof the direct sound field model, ^ is frequency, ^ is the distance, ^ is the angle, and F is a coherence tuning parameter.

45. The method according to claim 44, wherein F = 1.

46. The method according to any one of the previous claims, wherein the sound level of the audio source is a measurable loudness or a perceived loudness.

47. The method according to any one of the previous claims, wherein the audio source is moving and / or the position is changing.

48. The method according to any one of the previous claims, wherein the reverberant environment comprises objects that reflect and diffuse sound emitted by the audio source.

49. The method according to any one of the previous claims, wherein the position information is updated continuously.

50. The method according to any one of the previous claims, wherein the main acoustic axis of the audio source is a direction of a maximum sound pressure level emitted by the audio source.

51. A method of estimating a sound level of an audio source at a position in a reverberant environment, the method comprising: obtaining position information for the position, wherein the position information comprises a distance between the position and the audio source, and an angle of the position with respect to a main acoustic axis of the audio source; estimating the sound level of the audio source at the position based on a weighted sum of a reverberant sound field model of the audio source and direct sound field model of the audio source, wherein the direct sound field model is based on the angle and a weight of the weighted sum depends on the distance; outputting the estimated sound level for further processing.

52. The method according to claim 51, wherein estimating the sound level of the audiosource at the position comprises determining the sound level of the audio source as^(^, ^, ^) = HI&Jℎ^^^(^) ∗ ^^^(^, ^) + ^2^(^, ^, ^)where ^2^ ^^^(^, ^) is a level ofthe direct sound field model, HI&Jℎ^^^(^) is the weights, ^ is frequency, ^ is the distance, and ^ is the angle.

53. The method according to claim 51, wherein estimating the sound level of the audiosource at the position comprises determining the sound level of the audio source as^(^, ^, ^) = HI&Jℎ^^^(^) ∗ ^^^(^, ^) + HI&Jℎ^2^(^) ∗ ^2^(^, ^)where ^2^(^, A) is a level of the reverberant sound field model, ^^^(^, ^) is a level of thedirect sound field model, HI&Jℎ^^^(^) is a first weight of the weighted sum, HI&Jℎ^2^(^) is asecond weight of the weighted sum with HI&Jℎ^2^(^) = 1 − HI&Jℎ^^^(^), ^ is frequency, ^ isthe distance, and ^ is the angle.

54. The method according to claim 52 or 53, wherein HI&Jℎ^^^(^) is equal to 1 when ^is smaller than a minimal distance;HI&Jℎ^^^(^) is 0 when the distance is equal to a critical distance; andHI&Jℎ^^^(^) decreases linearly between the minimal distance and the critical distance.

55. The method according to any one of claims 52 to 54, wherein the reverberant sound field is independent of the angle and the frequency.

56. An apparatus, comprising a processor and a memory coupled to the processor, wherein the processor is adapted to carry out the method according to any one of claims 1 to 55.

57. The apparatus according to claim 56, wherein the apparatus is a loudspeaker or a device connected to the loudspeaker.

58. The apparatus according to claim 56, wherein the apparatus is a virtual reality headset, an augmented reality headset or a device connected to the virtual reality headset or the augmented reality headset.

59. A device comprising: a decoder for decoding a received bitstream; and the apparatus according to any one of claims 56 to 58; wherein the bitstream includes audio data and / or metadata for use during rendering by the apparatus.

60. A computer program comprising instructions that, when executed by a processor, cause the processor to carry out the method according to any one of claims 1 to 55.

61. A computer-readable storage medium storing the computer program according to claim 59.

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