Adaptive rendering of reverberation from loudspeakers

The method and apparatus dynamically adjust reverberation signals based on listener position using a feedback delay network and adaptive 3D panning to maintain consistent spatial perception, addressing the challenge of listener movement in augmented and virtual reality environments.

GB2636429APending Publication Date: 2025-06-18NOKIA TECHNOLOGIES OY
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Patent Information

Application Number
GB2023019173
Authority / Receiving Office
GB · GB
Patent Type
Applications
Current Assignee / Owner
Filing Date
2023-12-14
Publication Date
2025-06-18

AI Technical Summary

Technical Problem

Existing methods for spatially rendering reverberation in augmented and virtual reality environments fail to accurately adapt to a listener's changing position, leading to inconsistent and localized reverberation perception when using multiple loudspeakers.

Method used

A method and apparatus that dynamically adjust gain, delay, and direction of reverberation signals based on the listener's position, using a feedback delay network (FDN) reverberator and adaptive 3D panning to maintain consistent reverberation perception across different listener positions.

Benefits of technology

Ensures that the perceived reverberation remains consistent and spatially accurate regardless of the listener's location within the environment, enhancing the immersive audio experience in augmented and virtual reality applications.

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Abstract

A method for rendering reverberation, the method comprising obtaining a reverberated signal based on a listener position within an acoustic environment and obtaining a listener position within a repro
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Description

Field The present application relates to apparatus and methods for spatial rendering of reverberation, and not exclusively generating listener position adaptive rendering of reverberation from loudspeakers in augmented reality and / or virtual reality apparatus. Background Reverberation refers to the persistence of sound in a space after the actual sound source has stopped. Different spaces are characterized by different reverberation characteristics. For conveying spatial impression of an environment, reproducing reverberation perceptually accurately is important. Room acoustics are often modelled with individually synthesized early reflection portion and a statistical model for the diffuse late reverberation. Figure 1 depicts an example of a synthesized room impulse response where the direct sound 101 is followed by discrete early reflections 103 (or reflection echoes) which have a direction of arrival (DOA) and diffuse late reverberation 105 which can be synthesized without any specific direction of arrival. The delay dl(t) 102 in Figure 1 can be seen to denote the direct sound arrival delay from the source to the listener and the delay d2(t) 104 can denote the delay from the source to the listener for one of the early reflections (in this case the first arriving reflection). One method of reproducing reverberation is to utilize a set of N loudspeakers (or virtual loudspeakers reproduced binaurally using a set of head-related transfer functions (HRTF)). The loudspeakers are positioned around the listener somewhat evenly. Mutually incoherent reverberant signals are reproduced from these loudspeakers, producing a perception of surrounding diffuse reverberation. The positioning and the number of the loudspeakers suitable for producing the diffuse perception has been studied, and found that somewhere around 6-12 loudspeakers are required, depending on the positioning of the loudspeakers. The reverberation produced by the different loudspeakers has to be mutually incoherent. In a simple case the reverberations can be produced using the different channels of the same reverberator, where the output channels are uncorrelated but otherwise share the same acoustic characteristics such as reverberation time and level (specifically, the diffuse-to-direct ratio or reverberant-to-direct ratio or diffuse-to-total ratio or diffuse-to-source ratio or any other suitable parameter for representing reverberation energy or level). Such uncorrelated outputs sharing the same acoustic characteristics can be obtained, for example, from the output taps of a feedback delay network (FDN) reverberator with suitable tuning of the delay line lengths and mixing matrix, or from a reverberator based on using decaying uncorrelated noise sequences by using a different uncorrelated noise sequence in each channel. In this case, the different reverberant signals effectively have the same features, and the reverberation is typically perceived to be similar in all directions. Summary There is provided according to a first aspect a method for rendering reverberation, the method comprising: obtaining a reverberated signal based on a listener position within an acoustic environment; obtaining a listener position within a reproduction space; obtaining at least one reverberator output channel position; obtaining a position of at least one loudspeaker based on the at least one reverberator output channel position and the listener position within the reproduction space; determining a gain based on the position of the at least one loudspeaker, the listener position within the reproduction space, and the reverberator output channel position; applying the gain on the reverberated signal; and generating a loudspeaker signal to be reproduced from the at least one loudspeaker based on the applied gain on the reverberated signal. Obtaining a reverberated signal based on a listener position within an acoustic environment may comprise: obtaining at least one reverberation parameter associated with the acoustic environment; configuring at least one reverberator based on the at least one reverberation parameter; and processing at least one audio signal with the configured reverberator to produce the reverberated signal. Obtaining at least one reverberation parameter associated with the acoustic environment may comprise determining at least one delay line parameter. Obtaining the listener position within the reproduction environment may comprise adaptively obtaining the listener position, the listener position being time-varying. The method may comprise obtaining based on the adaptively obtained listener position, at least one of: at least one gain; and at least one delay. Configuring at least one reverberator based on the at least one reverberation parameter may comprise selectively or partially disabling the at least one reverberator based on determining the reproduction space within which the at least one loudspeaker is located is reverberant. Selectively or partially disabling the at least one reverberator may be based on one of: a user input to selectively or partially disable the at least one reverberator; and a flag or indicator, the flag or indicator configured to indicate that loudspeaker rendering of reverberation is to be skipped in order to reduce computational complexity, or if the reproduction environment is reverberant, or a combination of both. Applying the gain on the one reverberated signal comprises gain on the reverberated signal may produce a panned audio signal. Generating a loudspeaker signal to be reproduced from the at least one loudspeaker based on the applied gain on the reverberated signal may comprise summing the panned audio signal to generate a loudspeaker signal to be reproduced from the at least one loudspeaker. Summing the panned audio signal to generate a loudspeaker signal to be reproduced from the at least one loudspeaker may comprise summing the panned audio signal for a sub-set of panned audio signal to generate the loudspeaker signal to be reproduced from the at least one loudspeaker. The method may further comprise: determining at least one distance between the listener position and the position of the at least one loudspeaker within the reproduction space; determining a further gain based on the determined at least one distance; and rendering the loudspeaker signal from the at least one loudspeaker by applying the further gain to the loudspeaker signal. Obtaining at least one reverberator output channel position may comprise adaptively controlling the at least one reverberator output channel position. Adaptively controlling the at least one reverberator output channel position may comprise mapping the at least one reverberator output channel position to a nearest position of the at least one loudspeaker. The acoustic environment may be the same as the reproduction space in an augmented reality application. The acoustic environment may be a virtual reality environment and the reproduction space may be a real space within which the at least one loudspeaker is located. According to a second aspect there is provided an apparatus, for rendering reverberation, the apparatus comprising at least one processor and at least one memory storing instructions that, when executed by the at least one processor, cause the system at least to perform: obtaining a reverberated signal based on a listener position within an acoustic environment; obtaining a listener position within a reproduction space; obtaining at least one reverberator output channel position; obtaining a position of at least one loudspeaker based on the at least one reverberator output channel position and the listener position within the reproduction space; determining a gain based on the position of the at least one loudspeaker, the listener position within the reproduction space, and the reverberator output channel position; applying the gain on the reverberated signal; and generating a loudspeaker signal to be reproduced from the at least one loudspeaker based on the applied gain on the reverberated signal. The apparatus caused to perform obtaining a reverberated signal based on a listener position within an acoustic environment may be caused to perform: obtaining at least one reverberation parameter associated with the acoustic environment; configuring at least one reverberator based on the at least one reverberation parameter; and processing at least one audio signal with the configured reverberator to produce the reverberated signal. The apparatus caused to perform obtaining at least one reverberation parameter associated with the acoustic environment may be caused to perform determining at least one delay line parameter. The apparatus caused to perform obtaining the listener position within the reproduction environment may be caused to perform adaptively obtaining the listener position, the listener position being time-varying. The apparatus may be caused to perform obtaining based on the adaptively obtained listener position, at least one of: at least one gain; and at least one delay. The apparatus caused to perform configuring at least one reverberator based on the at least one reverberation parameter may be caused to perform selectively or partially disabling the at least one reverberator based on determining the reproduction space within which the at least one loudspeaker is located is reverberant. The apparatus caused to perform selectively or partially disabling the at least one reverberator may be based on one of: a user input to selectively or partially disable the at least one reverberator; and a flag or indicator, the flag or indicator configured to indicate that loudspeaker rendering of reverberation is to be skipped in order to reduce computational complexity, or if the reproduction environment is reverberant, or a combination of both. The apparatus caused to perform applying the gain on the one reverberated signal comprises gain on the reverberated signal may be caused to produce a panned audio signal. The apparatus caused to perform generating a loudspeaker signal to be reproduced from the at least one loudspeaker based on the applied gain on the reverberated signal may be caused to perform summing the panned audio signal to generate a loudspeaker signal to be reproduced from the at least one loudspeaker. The apparatus caused to perform summing the panned audio signal to generate a loudspeaker signal to be reproduced from the at least one loudspeaker may be caused to perform summing the panned audio signal for a sub-set of panned audio signal to generate the loudspeaker signal to be reproduced from the at least one loudspeaker. The apparatus may further be caused to perform: determining at least one distance between the listener position and the position of the at least one loudspeaker within the reproduction space; determining a further gain based on the determined at least one distance; and rendering the loudspeaker signal from the at least one loudspeaker by applying the further gain to the loudspeaker signal. The apparatus caused to perform obtaining at least one reverberator output channel position may be caused to perform adaptively controlling the at least one reverberator output channel position. The apparatus caused to perform adaptively controlling the at least one reverberator output channel position may be caused to perform mapping the at least one reverberator output channel position to a nearest position of the at least one loudspeaker. The acoustic environment may be the same as the reproduction space in an augmented reality application. The acoustic environment may be a virtual reality environment and the reproduction space may be a real space within which the at least one loudspeaker is located. According to a third aspect there is provided an apparatus for rendering reverberation, the apparatus comprising means configured to: obtain a reverberated signal based on a listener position within an acoustic environment; obtain a listener position within a reproduction space; obtain at least one reverberator output channel position; obtain a position of at least one loudspeaker based on the at least one reverberator output channel position and the listener position within the reproduction space; determine a gain based on the position of the at least one loudspeaker, the listener position within the reproduction space, and the reverberator output channel position; apply the gain on the reverberated signal; and generating a loudspeaker signal to be reproduced from the at least one loudspeaker based on the applied gain on the reverberated signal. The means configured to obtain a reverberated signal based on a listener position within an acoustic environment may be configured to: obtain at least one reverberation parameter associated with the acoustic environment; configure at least one reverberator based on the at least one reverberation parameter; and process at least one audio signal with the configured reverberator to produce the reverberated signal. The means configured to obtain at least one reverberation parameter associated with the acoustic environment may be configured to determine at least one delay line parameter. The means configured to obtain the listener position within the reproduction environment may be configured to adaptively obtain the listener position, the listener position being time-varying. The means may be configured to obtain based on the adaptively obtained listener position, at least one of: at least one gain; and at least one delay. The means configured to configure at least one reverberator based on the at least one reverberation parameter may be configured to selectively or partially disable the at least one reverberator based on determining the reproduction space within which the at least one loudspeaker is located is reverberant. The means configured to selectively or partially disable the at least one reverberator may be based on one of: a user input to selectively or partially disable the at least one reverberator; and a flag or indicator, the flag or indicator configured to indicate that loudspeaker rendering of reverberation is to be skipped in order to reduce computational complexity, or if the reproduction environment is reverberant, or a combination of both. The means configured to apply the gain on the one reverberated signal comprises gain on the reverberated signal may be configured to produce a panned audio signal. The means configured to generate a loudspeaker signal to be reproduced from the at least one loudspeaker based on the applied gain on the reverberated signal may be configured to sum the panned audio signal to generate a loudspeaker signal to be reproduced from the at least one loudspeaker. The means configured to sum the panned audio signal to generate a loudspeaker signal to be reproduced from the at least one loudspeaker may be configured to sum the panned audio signal for a sub-set of panned audio signal to generate the loudspeaker signal to be reproduced from the at least one loudspeaker. The means may be further configured to: determine at least one distance between the listener position and the position of the at least one loudspeaker within the reproduction space; determine a further gain based on the determined at least one distance; and render the loudspeaker signal from the at least one loudspeaker by applying the further gain to the loudspeaker signal. The means configured to obtain at least one reverberator output channel position may be configured to adaptively control the at least one reverberator output channel position. The means configured to adaptively control the at least one reverberator output channel position may be configured to map the at least one reverberator output channel position to a nearest position of the at least one loudspeaker. The acoustic environment may be the same as the reproduction space in an augmented reality application. The acoustic environment may be a virtual reality environment and the reproduction space may be a real space within which the at least one loudspeaker is located. According to a fourth aspect there is provided an apparatus for rendering reverberation, the apparatus comprising: obtaining circuitry configured to obtain a reverberated signal based on a listener position within an acoustic environment; obtaining circuitry configured to obtain a listener position within a reproduction space; obtaining at least one reverberator output channel position; obtaining circuitry configured to obtain a position of at least one loudspeaker based on the at least one reverberator output channel position and the listener position within the reproduction space; determining circuitry configured to determine a gain based on the position of the at least one loudspeaker, the listener position within the reproduction space, and the reverberator output channel position; applying circuitry configured to apply the gain on the reverberated signal; and generating circuitry configured to generate a loudspeaker signal to be reproduced from the at least one loudspeaker based on the applied gain on the reverberated signal. According to a fifth aspect there is provided a computer program comprising instructions [or a computer readable medium comprising instructions] for causing an apparatus, for rendering reverberation, the apparatus caused to perform at least the following: obtaining a reverberated signal based on a listener position within an acoustic environment; obtaining a listener position within a reproduction space; obtaining at least one reverberator output channel position; obtaining a position of at least one loudspeaker based on the at least one reverberator output channel position and the listener position within the reproduction space; determining a gain based on the position of the at least one loudspeaker, the listener position within the reproduction space, and the reverberator output channel position; applying the gain on the reverberated signal; and generating a loudspeaker signal to be reproduced from the at least one loudspeaker based on the applied gain on the reverberated signal. According to a sixth aspect there is provided a non-transitory computer readable medium comprising program instructions for causing an apparatus, for rendering reverberation, to perform at least the following: obtaining a reverberated signal based on a listener position within an acoustic environment; obtaining a listener position within a reproduction space; obtaining at least one reverberator output channel position; obtaining a position of at least one loudspeaker based on the at least one reverberator output channel position and the listener position within the reproduction space; determining a gain based on the position of the at least one loudspeaker, the listener position within the reproduction space, and the reverberator output channel position; applying the gain on the reverberated signal; and generating a loudspeaker signal to be reproduced from the at least one loudspeaker based on the applied gain on the reverberated signal. According to a seventh aspect there is provided an apparatus, for rendering reverberation, comprising: means for obtaining a reverberated signal based on a listener position within an acoustic environment; means for obtaining a listener position within a reproduction space; means for obtaining at least one reverberator output channel position; means for obtaining a position of at least one loudspeaker based on the at least one reverberator output channel position and the listener position within the reproduction space; means for determining a gain based on the position of the at least one loudspeaker, the listener position within the reproduction space, and the reverberator output channel position; means for applying the gain on the reverberated signal; and means for generating a loudspeaker signal to be reproduced from the at least one loudspeaker based on the applied gain on the reverberated signal. According to an eighth aspect there is provided a computer readable medium comprising instructions for causing an apparatus, for rendering reverberation, to perform at least the following: obtaining a reverberated signal based on a listener position within an acoustic environment; obtaining a listener position within a reproduction space; obtaining at least one reverberator output channel position; obtaining a position of at least one loudspeaker based on the at least one reverberator output channel position and the listener position within the reproduction space; determining a gain based on the position of the at least one loudspeaker, the listener position within the reproduction space, and the reverberator output channel position; applying the gain on the reverberated signal; and generating a loudspeaker signal to be reproduced from the at least one loudspeaker based on the applied gain on the reverberated signal. An apparatus comprising means for performing the actions of the method as described above. An apparatus configured to perform the actions of the method as described above. A computer program comprising program instructions for causing a computer to perform the method as described above. A computer program product stored on a medium may cause an apparatus to perform the method as described herein. An electronic device may comprise apparatus as described herein. A chipset may comprise apparatus as described herein. Embodiments of the present application aim to address problems associated with the state of the art. Summary of the Figures For a better understanding of the present application, reference will now be made by way of example to the accompanying drawings in which: Figure 1 shows a model of room acoustics with regard to the room impulse response; Figure 2 shows schematically an example apparatus within which some embodiments may be implemented; Figure 3 shows a flow diagram of the operation of the example apparatus as shown in Figure 2 with respect to reverberant audio signal rendering; Figure 4 shows schematically an example reverberator controller as shown in Figure 2 in further detail according to some embodiments; Figure 5 shows a flow diagram of the operation of the example reverberator controller as shown in Figure 4; Figures 6a to 6d show example reverberation outputs rendered from constant user-relative positions using adaptive loudspeaker rendering; Figure 7 shows schematically an example adaptive loudspeaker renderer as shown in Figure 2 in further detail according to some embodiments; Figure 8 shows a flow diagram of the operation of the example adaptive loudspeaker renderer as shown in Figure 7; Figures 9a to 9d show further example reverberation outputs rendered from constant user-relative positions using adaptive loudspeaker rendering; Figure 10 shows a flow diagram of a further operation of the example adaptive loudspeaker renderer as shown in Figure 7; Figures 11a and 11b show an example system within which some embodiments can be implemented; Figure 12 shows a reverberator which includes an example feedback delay network (FDN) reverberator as shown in Figure 2 in further detail according to some embodiments; and Figure 13 shows an example device suitable for implementing the apparatus shown in previous figures. Embodiments of the Application The following describes in further detail suitable apparatus and possible mechanisms for controlling the rendering of audio scenes. As indicated above a method of reproducing reverberation is to utilize a set of N loudspeakers (or virtual loudspeakers reproduced binaurally using a set of head-related transfer functions (HRTF)). The loudspeakers are positioned around the listener somewhat evenly. Mutually incoherent reverberant signals are reproduced from these loudspeakers, producing a perception of surrounding diffuse reverberation. The positioning and the number of the loudspeakers suitable for producing the diffuse perception has been studied, e.g., in K. Hiyama, S. Komiyama, and K. Hamasaki, The Minimum Number of Loudspeakers and Its Arrangement for Reproducing the Spatial Impression of Diffuse Sound Field, AES 113th Convention, 2002 and C. Kirch, J Poppitz, T. Wendt, S. van der Par, and S. Ewert, Spatial Resolution of Late Reverberation in Virtual Acoustic Environments. Submitted to Trends in Hearing (currently available in the Carl von Ossietzky Universitat Oldenburg website), 2021. It has been found that somewhere around 6-12 loudspeakers are required, depending on the positioning of the loudspeakers. The reverberation produced by the different loudspeakers is designed to be mutually incoherent, but, otherwise their content can be otherwise configured by the algorithm designer. In a simple case, for example, the reverberations can be produced using the different channels of the same reverberator, where the output channels are uncorrelated but otherwise share the same acoustic characteristics such as RT60 time and level (specifically, the diffuse-to-direct ratio or reverberant-to-direct ratio or source-energy-to-diffuse energy ratio). Such uncorrelated outputs sharing the same acoustic characteristics can be obtained, for example, from the output taps of a Feedback-Delay-Network (FDN) reverberator with suitable tuning of the delay line lengths, or from a reverberator based on using decaying uncorrelated noise sequences by using a different uncorrelated noise sequence in each channel. In this case, the different reverberant signals basically have the same features, and the reverberation is typically perceived to be similar to all directions. Traditional loudspeaker listening situations assume that the listener is static and remains in a so-called sweet spot. In a 6DoF listening situation, the listener is free to move in the space surrounded by loudspeakers and in some situations also outside the space surrounded by loudspeakers (the space within which the loudspeakers are arranged is also referred to as a loudspeaker circle for pure horizontal or 2D loudspeaker configurations). This potential movement of the listener thus requires any audio rendering process to be continuously adapted to follow any changed in the listener position. A reproduction space is usually the real space of the listener. That is, the room where the physical loudspeakers are located. Acoustic environment is either: a VR acoustic environment describing a virtual space having acoustic characteristics or dimensions; or it can be an AR environment describing the acoustic characteristics and dimensions of the reproduction space. In the case of the AR reproduction the acoustic environment and the reproduction space are the same. A currently discussed approach, such as found in the current working draft of ISO / IEC 23090-4, is one where the renderer can adapt to the changing listener position by applying gains and delays to loudspeaker signals such that the loudspeaker signals arrive at the listener position (off the sweet spot position) with a similar level and delay as they would arrive at the sweet spot position. As a part of the process the directions and distances of the sound sources and loudspeakers from the listener are updated based on the listener position. This approach is referred as Level 1 loudspeaker rendering. Another approach found in the current working draft of ISO / IEC 23090-4 is one where the listener perceives a 6DoF audio scene (e.g., with an audio object) while moving within the loudspeaker arrangement, referred as Level 2 loudspeaker rendering. The updated distances and directions of sound sources and loudspeakers are considered when performing 3D amplitude panning with the loudspeaker setup. The loudspeaker setup is provided as an input within the Listening Space Description Format (LSDF) file to the renderer. As a result, the renderer can reproduce 6DoF audio using physical loudspeakers while the listener moves within the loudspeaker area. The application of adaptive time varying rendering prevents the perceived sound from being perceived to ‘collapse’ towards a nearest loudspeaker but is located at the sound source place in the virtual space (which corresponds to a place in the real listening space). In a virtual acoustics rendering system, the reverberation is typically rendered as a combination of a certain number of distinct early reflections and a stochastic model for the late reverberation. The early reflection synthesis is typically time varying whereas the late reverberation synthesis is not time varying. Together these two early and late components create a plausible reverberation rendering for a physical or virtual space. When reproducing reverberation from physical loudspeakers, uncorrelated reverberation signals can each be reproduced from physical loudspeakers. However, as discussed above rendering in 6DoF scenes can adaptively render the scene based on listener position to prevent a nearest speaker sound source collapse, meaning that the user will hear too much of the reverberation signal of the closest loudspeaker and too little of the reverberation from the farther away speakers. The concept as discussed herein in further detail in the following examples and embodiments is one of application of 6DoF adaptive rendering methods for rendering reverberation from physical loudspeakers. This can, for example, in some embodiments be implemented with respect to reproduction of (late) reverberation according to reverberation parameters of a (virtual or physical) acoustic space where apparatus and methods propose that rendering of reverberation of an acoustic environment under time-varying listener-position to loudspeakers so that the position dependent changes in the reverberation perceived by the listener depend on the listener position within the acoustic space but not on the listener position with regard to the loudspeakers. In some embodiments a reverberation apparatus and methods of applying reverberation can be configured to obtain or receive, the reverberation parameters (RT60, direct-to-reverberant ratio, etc.) which control the desired characteristics of reverberation. When reproducing the reverberated signals via loudspeakers, the rendering aims to adaptively control the gain, delay, and direction of reverberation signals such that the listener perceives reverberation according to the reverberation parameters and the reverberation is not affected by the changing listener position with respect to the loudspeakers. This can, in some embodiments, be achieved by: obtaining reverberation parameters containing at least a reverberation time; configuring a digital reverberator with reverberator parameters derived from the reverberation parameters, where at least one attenuation filter is configured based on the reverberation time; rendering a signal using the reverberator and at least one input signal to produce at least one reverberation signal; obtaining a user position; obtaining a loudspeaker position; calculating a distance between the user position and the loudspeaker position; based on the distance, calculating a gain; using the gain, rendering the reverberation signal from the loudspeaker; In some embodiments, the reverberator is configured to produce N nearly mutually decorrelated outputs. The reverberator outputs are given spatial positions surrounding the listener. Adaptive loudspeaker rendering is achieved by updating the spatial positions based on the current listener position in the real room and then applying adaptive 3D panning to the updated spatial positions and adjusting the gain and delay for the loudspeaker signals. The adaptive 3D panning to the updated spatial positions and the adjustment of the gain and delay for the loudspeaker signals can keep the rendered spatial positions of reverberation static with regards to the listener and the adaptive panning ensures that the levels and delays of the reverberation signals react to changing listener position, avoiding being localized to the closest loudspeaker. In some further embodiments, the spatial positions of the reverberation signals are updated based on the listener position such that each reverberation is reproduced from the physical loudspeaker the direction of which is closest to the direction of the sound source. In such embodiments panning between loudspeakers is effectively not applied as the directions of reverberation sources coincide with loudspeaker directions (thus making the panning gain for each reverberation source 1 for a single loudspeaker and zero for others). The distances to reverberation sources are kept constant and thus the 3D panning keeps the level of reverberation signals constant. In another set of embodiments, decorrelation is applied to the reverberation signals. With respect to some embodiments, the reverberation is reproduced from a subset of the loudspeakers. An example subset of the loudspeakers comprises front left and right speakers and back left and right speakers. In another set of embodiments a real room reverberation is taken into account when adjusting the reverberator based on the reverberation time. In some embodiments bitstream signalling is employed to control how reverberation is rendered into loudspeakers. Example signalling can include: omitting reverberation rendering if physical room is reverberant; and compensating for physical room reverberation; In some example embodiment, a Feedback-Delay-Network (FDN) reverberator is employed with a number of output channels equal to the number of recirculating delay lines. The reverberator can, for example, be configured with 3, 7, 15, 31, or 63 delay lines. The FDN as described herein and summarized herein later can be one similar to the those described in WO2022167720. Figure 2 shows an example system or apparatus 299 suitable for implementing embodiments within a late or later reverberation audio signal processing system. The system comprises inputs such as audio signal 200 and reverberation parameters 202. The system or apparatus 299 is configured to render reverberant loudspeaker audio signals 212 as an output, containing late reverberation which are perceived according to the acoustic parameters carried in the reverberation parameters 202. The acoustic parameters can describe the acoustics of an acoustic environment such as a virtual environment. The acoustic parameters can also describe the acoustics of a reproduction space, that is, a real listening environment or room. In some embodiments the reverberation parameters 202 comprises suitable parameters for configuring the reverberator 201. Suitable reverberation parameters 202 can be: predelay tpre, RT60 or reverberation times T60(fc) in frequency bands (where k is the frequency band index), reverberation ratio (such as reverberant-to-direct ratio RDR(ky), room dimensions, and number of delay lines. Alternatively to the RDR, the diffuse-to-source energy ratio (DSR) can be used. The reverberation parameters 202 can, for example, be obtained from a bitstream (representing the reverberation parameters of a virtual scene) or from a listening space description format (LSDF) input to the renderer (representing the parameters of a physical scene). Additionally, the system or apparatus 299 can be configured to obtain or receive a listener position and orientation 210 or listener pose from a head tracking apparatus. The system or apparatus 299 can furthermore be configured to obtain or receive a loudspeaker setup 208 from the LSDF input. In some embodiments the system 299 comprises a Reverberator controller 203 configured to obtain the reverberation parameters 202 and to convert the reverberation parameters 202 into suitable reverberator parameters 204 for controlling the reverberator 201. In some embodiments the system 299 comprises a reverberator 201 configured to receive the audio signal 200 (t), where t is the sample (time) index. Furthermore, the reverberator 201 is configured to receive the reverberator parameters 204. The reverberator 201 is shown herein using a feedback network (FN) implemented as a feedback delay network (FDN) but in other embodiments the FN can be implemented using other suitable feedback architectures. In this example embodiment, the reverberator 201 has D output channels indexed with d = 1,2, The resulting reverberant audio signals 210 srev(t,d) are mutually incoherent, and they have acoustical characteristics according to the reverberator parameters 204 and their number D depends on the input diffuseness (number of delay loops). Here, t denotes the sample index (time). In some embodiments the system 299 comprises an adaptive loudspeaker renderer 205. The reverberant audio signals 206 srev(t, d) are forwarded to the adaptive loudspeaker renderer 205. The adaptive loudspeaker renderer 205 in some embodiments is configured to further obtain the listener position and orientation 210 and the loudspeaker setup 208 and render the reverberated audio signals to reverberant loudspeaker audio signals 212 slSiadaptive(t,i) (where i is the loudspeaker channel index), which can, for example, be reproduced using physical loudspeakers. The adaptive loudspeaker renderer 205 thus takes the listener position and orientation 210 and loudspeaker setup 208 into account when panning the directional reverberant audio signals 206 into reverberant loudspeaker audio signals 212 such that the listener position affects the listener position in the virtual audio scene and is compensated for during panning. The reverberant loudspeaker audio signals 212 can thus be perceived as surrounding and enveloping with acoustical characteristics according to reverberation parameters 202 and such that acoustical characteristics change dependent on the listener position in the virtual audio scene, as intended. Figure 3 shows an example flow diagram of the operations of the system shown in Figure 2 with respect to the reverberator and adaptive loudspeaker renderer associated with the reverberator. First there is obtaining the audio signals, and reverberation parameters (predelay, RT60, reverberation ratio, room dimensions, number of delay lines D) as shown by 301. Then there is the operation of determining reverberator parameters (or otherwise reverberator configuration) from the reverberation parameters input as shown by 303. Then reverberated audio signals are generated based on the audio signal and the configured reverberator as shown by 305. Then reverberant adaptive loudspeaker audio signals are rendered using the listener position and orientation and loudspeaker setup parameters as shown by 307. Then reverberant loudspeaker signals are output as shown by 309. Figure 4 shows schematically an example reverberator controller 203 as shown in Figure 2 in further detail. The reverberator controller 203 in some embodiments is configured to obtain the reverberation parameters 202 (predelay, RT60, reverberation ratio, room dimensions, number of delay lines D) and generate suitable reverberator parameters 204 (delay line length; delay line attenuation filter; reverberation ratio control filter; pre-delay line length; feedback matrix coefficients; and output channel position). For example, the reverberator parameter determiner 203 comprises a delay line length determiner 401. The delay line length determiner 401 in some embodiments is configured to receive the room dimensions and delay line number D to determine the delay line lengths. The delay line length determiner 401 can, in some implementations, be configured to determine delay line lengths md for each of D delay lines based on virtual room dimensions. Here, we use the dimensions of the enclosure, that is, a bounding box that encloses or is aligned with the walls of the physical or virtual room. For example, a shoebox shaped room can be defined with dimensions xDim,yDim,zDim. If the room is not shaped as a shoebox (or cuboid) then a shoebox can be fit inside the room and the dimensions of the fitted shoebox can be utilized for the delay line lengths. Alternatively, the dimensions can be obtained as three longest orthogonal dimensions in the non-shoebox shaped room, or by another suitable method. The delays can in some embodiments be set proportionally to standing wave resonance frequencies in the virtual room or physical room. In some embodiments there can be a first set of roomModes as below (with the three values depicting the x, y, and z of the mode, respectively): roomModes{ {1.0, 0.0, 0.0}, {2.0, 1.0, 1.0}, {0.0, 2.0, 1.0}, {1.0, 0.0, 1.0}, {2.0, 1.0, 0.0}, {0.0, 1.0, 1.0}, {1.0, 1.0, 1.0}, {1.0, 1.0, 0.0}, {0.0, 1.0, 2.0}, {1.0, 2.0, 1.0}, {1.0, 2.0, 0.0}, {0.0, 0.0, 1.0}, {0.0, 1.0, 0.0}, {1.0, 0.0, 2.0}, {2.0, 0.0, 1.0}, With these roomModes the delay line lengths can be calculated for a maximum of D=15. In case there are more delay lines (D>15), additional room modes can be calculated by modifying these modes and collecting the modified modes which are between a certain minimum and maximum frequency. A frequency corresponding to a mode can be calculated as freq = sqrt(mode.x * mode.x + mode.y * mode.y + mode.z * mode.z); The above modes can be modified by incrementing the x, y, z values by fractional steps such as 0.0, 0.4, 0.8, 1.2, 1.6, 2.0 or other suitable amounts. A modified mode can be checked if its frequency is between the minimum and maximum frequency of the first set of roomModes, and selected to be used if it falls in between. In such a manner a necessary number of additional modes can be created this way until enough room modes are obtained (as many as there are delay lines). In some alternative embodiments other methods can be used for the delay line length determination. The delay line lengths md can further be made to be mutually prime integers. This choice can then minimize coherent repetition in the impulse response of the FN. The sieve of the Sundaram algorithm can be used to find the prime numbers up to the maximum delay line length. Each delay line length can then be mapped to the closest prime number in the obtained set of prime numbers. In some embodiments the reverberator controller 203 further comprises a delay line attenuation filter parameter determiner 403 which is configured to determine attenuation filter coefficients. The attenuation filter coefficients in the delay lines can be configured so that an amount of attenuation happens at each signal recirculation through the delay line so that the desired RT60(k) time is achieved. This is done in a frequency dependent manner to ensure the appropriate rate of decay of signal energy at specified frequencies. For a frequency bin k, the desired attenuation per signal sample is calculated as attenuationPerSample^k) = —60 / (fs * RT60(ky), where fs is the sampling rate. The attenuation in decibels for a delay line of length md is then attenuationDb^k) = md * attenuationPerSample(ky Furthermore, the attenuation filters are designed as cascade graphic equalizer filters as described in V. Valimaki and J. Liski, “Accurate cascade graphic equalizer,” IEEE Signal Process. Lett., vol. 24, no. 2, pp. 176-180, Feb. 2017, for each delay line. The design procedure outlined takes as input a set of command gains at octave bands. There are also methods for a similar graphic EQ structure which can support third octave bands, increasing the number of biquad filters to 31 and providing a better match for detailed target responses such as indicated in J. Ramb, J. Liski, and V. Valimaki, “Third-Octave and Bark Graphic-Equalizer Design with Symmetric Band Filters,” Applied Sciences, vol. 10, no. 4, p. 1222, Feb. 2020. In some embodiments the reverberator controller 203 comprises a reverberation ratio control filter parameter determiner 405 which can be configured to determine reverberation ratio control filter parameters. The control filter parameters can be designed such that, when the filter is applied to the input signal of the reverberator, the output reverberation will have the desired energy ratio defined by the RDR(k). The input to the design procedure can in some embodiments be the vector of RDR values RDR(k) obtained by the reverberator parameter determiner 203. The GEQratio is configured to match the reverberator spectrum energy to the target spectrum energy. To do this, an estimate of the RDR of the reverberator output and the target RDR is determined. The RDR of the reverberator output can be obtained by rendering a unit impulse through the reverberator using the first reverberator parameters and measuring the energy of the reverberator output and energy of the unit impulse and calculating the ratio of these energies. In some embodiments the energy of the reverberator output can be measured from a certain time onwards, for example, from an input predelay time or other time indicating the beginning of where the RDR data has been given / provided / measured. In some embodiments the reverberation ratio control filter parameter determiner 405 is configured to create an input of a unit impulse followed by zeroes of a sufficient length to capture the reverberation tail. In practice, the length of the zero tail is adjusted to equal max( / ?T60( / c)) plus the tpre in samples. The monophonic output signal srev(t), which is a function of time t, can be obtained by summation of the outputs of the feedback network . A long FFT (of length NF FT) is calculated over srev(t) and its absolute value can be obtained as FFA(kk) = abs (FFT(svev(l)) Here, kk are the FFT bin indices. We furthermore obtain the positive half spectral energy density as S(kk) = 1 / NFFT * FFA(kk)2 where we add the energy from the negative frequency indices kk into the corresponding positive frequency indices kk. The energy of a unit impulse can be calculated or obtained analytically. Denote it as Su(kk). In some embodiments the band energies are calculated of both the positive half spectral energy density of the reverberator S(kk) and the positive half spectral energy density of the unit impulse Su(kky Band energies can be calculated as bhigh S{kk~) kk=biow where biow and bhigh are the lowest and highest bin indices belonging to band k, respectively. The band bin indices can be obtained by comparing the frequencies of the bins to the lower and upper frequencies of each band. The reproduced RDRrev(k~) of the reverberator output at the frequency band k is obtained as RDRrev(k~) = S(k) / Su(k) The target linear magnitude response for GEQratio can be obtained as rdrFilterTargetResponse(k') = sqrt^RDR(k')') / sqrt(RDRrev(k)') where RDR^k) is the linear target RDR value at band k. The target response controlGain^k) = 20 * log1Q(rdrFilterTargetResponse(ky) is input as for the graphic equalizer design routine as discussed in the previously cited documents of V. Valimaki and J. Liski, “Accurate cascade graphic equalizer,” IEEE Signal Process. Lett, vol. 24, no. 2, pp. 176-180, Feb. 2017 and J. Ramd, J. Liski, and V. Valimaki, “Third-Octave and Bark Graphic-Equalizer Design with Symmetric Band Filters,” Applied Sciences, vol. 10, no. 4, p. 1222, Feb. 2020. The RDR filter target response (control gains for the graphic EQ design routine) can also be obtained directly in the logarithmic domain as controlGain(k) = 10 * log10(RDR(k)) — 10 * log10(RDRrev(k)) In some embodiments the reverberator controller 203 comprises a pre-delay line length determiner 407. The pre-delay line length determiner 407 is configured to determine and adjust the length mpre in samples of the pre-delay line based on the input pre-delay time tpre. The input pre-delay can be converted to samples and the length of the shortest FN delay line is subtracted from it. This can be set as the length mpre. This will cause the first echoes from the FN to occur after time tpre. In some other embodiments, the diffuse portion of the FN is set to start after time tpre. In this case mpre can be lower and can be set based on an estimate of the time at which the FDN output becomes diffuse and by setting the pre-delay line such that the desired time can be obtained. The reverberator parameter determiner 203 further comprises a feedback matrix coefficient and output channel position determiner 409 configured to generate feedback matrix coefficients. The core processing of the FDN calculation can in some embodiments be realized by the below method process: process (sampleCount) { m oneOverM = l.Of / sqrt(m outputcount + 1); / / m outputcount is equal to the number of delay line loops D m outputBuffer->fill (0.F); outputScaler{ l.Of / m outputcount }; vector updated^v(m^outputCount); for (samplelndex = 0; samplelndex <sampleCount; samplelndex++) { d = accumulate(m v[0], m v[m outputcount], 0); alphaTimesD = m alpha * d; fill(updated v, O.Of); for (delayLinelndex = 0; delayLinelndex <m^outputCount; delayLine!ndex++) { / / Read position starts from the beginning of the ring buffer and moves backwards (from / / the beginning of the second half which repeats the first half). float* s = &m_v[m^outputCount - delayLinelndex]; b = 0; c = 0 ; for (dlld2 = 0; dlld2 <m outputcount; dlld2++) { if (m hasPositiveFeedbackGain[dlld2]) { b += s[dlld2]; ) else { c += s[dlld2]; } } a = m oneOverM * (b - c) + alphaTimesD; delayedSample = m delayLines[delayLinelndex].getDelayedSample(m inputBuffer->at(samplelndex) + a); filteredSample = m^filters[delayLinelndex] - >filter(delayedSample); updated v[delayLinelndex] = filteredSample; (*m_outputBuffer)[delayLinelndex][samplelndex] += filteredSample; } copy (&updated v[0], m outputcount, &m v[0]); copy (&updated_v[0], m^outputCount, &m_v[m^outputCount]); for (outputChannellndex = 0; outputChannellndex <m outputcount; outputChannelIndex++) { (*m outputBuffer)[outputChannellndex][samplelndex] * = outputScaler; } } } The following parameters are needed to implement the core feedback matrix calculations of the FDN reverberator. In the example FDN shown later, the feedback matrix is depicted with A. The parameters required for feedback matrix calculation are listed for different numbers of reverb delay line loops D: / / D = 3 m hasPositiveFeedbackGain = { 0, 1, 0, }; m alpha = -0.1666666667f; / / D = 7 m hasPositiveFeedbackGain = { 0, 1, 1, 0, 1, 0, 0, }; m_alpha = -0.0923495156f; / / D = 15 m hasPositiveFeedbackGain = { 0, 1, 1, 1, 0, 1, 1, 0, 0, 1, 0, 1, 0, 0, 0, }; m^alpha = -0.0500000000f; / / D = 31 m hasPositiveFeedbackGain = { 0, 1, 1, 1, 1, 0, 1, 1, 0, 1, 0, 0, 1, 1, 0, 0, 0, 0, 0, 1, 1, 1, 0, 0, 1, 0, 0, 0, 1, 0, 1, }; m^alpha = -0.0265555905f; / / D = 63 m_hasPositiveFeedbackGain = { 0, 1, 1, 1, 1, 1, 0, 1, 1, 1, 1, 0, 0, 1, 1, 1, 0, 1, 0, 1, 1, 0, 0, 0, 0, 1, 0, 1, 1, 1, 0, 0, 0, 1, 1, 0, 1, 1, 0, 1, 0, 0, 1, 0, 0, 0, 1, 0, 0, 1, 1, 0, 0, 1, 0, 1, 0, 1, 0, 0, 0, 0, 0, }; m^alpha = -0.0138888889f; / / D = 127 m hasPositiveFeedbackGain = { o, 1, i, 1, 1, 1, 1, o, 1, 1, 1, 1, 1, 0, 0, 1, 1, 1, 1, o. i, o. 1, 1, 1, o, o, o, 0, 1, 1, 0, 1, 1, 1, 0, 1, 0, o, 1, 1, o, o. 0, 1, o, 1, 0, 1, 1, 0, 0, 0, o, 0, 1, o, 1, 1, 1, 1, 0, o. o. 1, 1, 1, 0, 1, 1, 0, 1, 1, o, o. 1, o. o, 1, o. 1, o, 0, 1, 0, 0, 0, 0, 1, o. o, 1, 1, 1, o, o, 1, o, 1, 1, 0, 1, 0, 0, 0, 1, 0, o, o, 1, 1, o, o, 1, 1, o, 1, o, 1, o, 1, o, 0, 0, 0, o, o, }; m alpha = -0.0071780445f; In the above embodiment, m_hasPositiveFeedbackGain lists, for each delay line index d, whether the delay line value should be summed to the positive sum accumulator b (value of 1) or the negative sum accumulator c (value of zero). The benefit of the above embodiment is that m hasPositiveFeedbackGain consists of zeros and ones and can thus be compactly stored in computer memory. In some embodiments the feedback matrix coefficient and output channel position determiner 409 is further configured to output channel positions based on a selection from a stored table. For example, the azimuth and elevation angles in degrees for output directions can be selected from the table below: Azimuth degrees (degrees) Elevation (degrees) 90 0 -90 0 114 20 -60 -6 85 -44 -130 -21 49 41 -67 52 154 -9 -48 -55 19 -31 -162 21 151 73 -9 20 180 -63 The output of this step is the reverberator output directions setup, 0rev(i), 0rev(O- As another example, the loudspeaker output positions can be selected using 5 a suitable algorithm or software routine which places the output channel positions substantially uniformly (equidistant on azimuth, elevation or both) on a sphere, or at horizontal circles. The delay lines can be assigned the Loudspeaker setup positions such that the delay lines are arranged into ascending order according to their lengths and the 10 output positions are assigned starting from the top of the table above. Figure 5 shows example operations of the reverberator controller 203. The first operation is obtaining the reverberation parameters (predelay, RT60, reverberation ratio, room dimensions, number of delay lines D) as shown by 501. 15 Then is shown determining delay line lengths based on the room dimensions and D as shown by 503. Following this is determining delay line attenuation filter parameters based on the delay line lengths, D and RT60(k') as shown by 505. Then determining reverberation ratio control filter parameters based on the reverberation ratio as shown by 507. After this is determining length of a pre-delay line based on the pre-delay and the delay line lengths as shown by 509. Then is the determination of the feedback matrix calculation parameters and the output channel positions based on D as shown by 511. Figure 12 shows in further detail the reverberator 201 introduced in Figure 2. As discussed above the reverberator 201 can be used to produce D uncorrelated outputs. In some embodiments the reverberator 201 is configured to produce D=15 (or some other determined number for D) outputs which are subsequently encoded to be rendered from different spatial directions defined by the directional configuration. In some embodiments the reverberator 201 comprises a pre-delay line z-mPre 1201, configured to receive and delay the input audio signal. The reverberator 201 also comprises a reverberation ratio control filter GEQratio 1203 which is configured to receive the pre-delay line output. The reverberator 201 further comprises a number D of feedback delay lines z~md 1251 and corresponding feedback delay line attenuation filters GEQd 1253. The signals which are output from GEQd 1253 are sent to inputs of a feedback matrix A 1257. D signal combiners 1254 (adders) sum the outputs of the feedback matrix A 1257 with the output of GEQratio 1203 to be used as inputs to each of the feedback delay lines z-md <| 251 The output of the feedback delay line attenuation filters 1253 are routed to D signal multipliers 1261 which in turn output the reverberant audio signals 206. The output of the feedback delay line attenuation filters 1253 are also the outputs of the FDN and outputs of the feedback delay line loops. The D feedback delay line loops of the FN thus comprise the adders 1254, feedback delay lines 1251, and feedback delay line attenuation filters 1253. The number of these feedback delay line loops determines the complexity of the FN calculation as the calculations need to be done for each delay line loop in all these elements, and also in the feedback matrix A 1257, for each incoming audio sample of Audio signal 200. Thus, in some embodiments the reverberator 201 is configured to receive reverberator parameters which comprise a delay length mpre, in samples, for pre-delay line z-mp™ 1201, coefficients of a reverberation ratio control filter GEQratio 1203, delay lengths md for each of D feedback delay lines z~md 1251, coefficients for each of D feedback delay line attenuation filters GEQd 1253, and coefficients for the feedback matrix A 1257. The reverberator parameters also in some embodiments comprise output channel gains Cd which are used to configure D signal multipliers 1261. In some embodiments the attenuation filter GEQd is a graphic equalizer (EQ) filter using M biquad HR band filters. In the case of octave-band filtering, M = 10. Thus, the reverberator parameters corresponding to each graphic EQ filter comprise the feedforward and feedback coefficients for 10 biquad HR filters, the gains for biquad band filters, and the overall gain. The feedback delay lines z^md 1251 can also be referred as loop delay lines or recirculating delay lines and the feedback delay line attenuation filters GEQd 1253 can be referred to as loop filters or recirculating filters. In some embodiments the coefficients of feedback matrix A 1257 are hardcoded in software code rather than provided as parameters. The reverberator thus comprises multiple recirculating delay lines (also referred as delay line loops) associated with the feedback network (FN) 1250. The feedback matrix A 1257 is used to control the recirculation gain and routing within the network. The feedback delay line attenuation filters GEQd 1253 can be implemented in some embodiments as graphic EQ filters implemented as cascades of second-order section HR filters and can facilitate controlling the energy decay rate at different frequencies. The feedback delay line attenuation filters GEQd 1253 furthermore are designed such that they attenuate the signal by the desired amount with each pass through the FN such that the desired reverberation time (RT60) is achieved. With respect to Figure 7 is shown a schematic view of an example adaptive loudspeaker renderer 205. The adaptive loudspeaker renderer 205 in some embodiments is configured to receive or obtain the listener position and orientation 210, Loudspeaker setup 208 0is(i), cpLs(i^, i = 1, and Reverberated audio signal 206 srev(t, d) and generate the reverberant loudspeaker audio signals 212. In some embodiments the loudspeaker setup can be one of the CICP loudspeaker setups defined in ISO / IEC 23091 3:2017(E) Information technology — Coding independent code-points — Part 3: Audio. As an example, two setups are given below, 5 is the common 5.1 setup and 12 is a horizontal setup with more loudspeakers. Value "Front / Surr. LFE" notation Loudspeaker names in loudspeaker layout Loudspeaker abbrev. Informative geometric position Azim., Elev. Ch. is LFE 5 3 / 2.0 centre front, C 0, o 0 left front, L 30, 0 0 right front, R -30, 0 0 left surround, Ls 110, 0 0 right surround Rs -110, 0 0 12 3 / 4.1 centre front, C 0, o 0 left front, L 30, 0 0 right front, R -30, 0 0 left surround, Ls 110, 0 0 right surround, Rs -110, 0 0 rear surround left, Lsr 135, 0 0 rear surround right, Rsr -135, 0 0 LFE LFE 0, -15 1 5 The LSDF file can in some embodiments contain the index of the CICP setup and the loudspeaker setup information such as loudspeaker setup origin or reference point and the loudspeaker positions with regard to the reference point. The loudspeaker setup origin can in some situations be also referred to as the 10 sweet spot and denoted with label s. The loudspeaker setup 0k(i), (pi^i) denotes loudspeaker directions relative to s. The adaptive loudspeaker rendering of reverberated signals can involve the following: updating the listener position in the real space; 15 updating the reverberator output directions based on the updated listener position information; updating the loudspeaker directions based on the listener position; calculating panning gains for the reverberator outputs based on the updated reverberator output directions and the updated loudspeaker directions; rendering loudspeaker signals using the panning gains and the reverberator output signals; calculating additional gain and delay parameters for the loudspeaker signals to compensate for the difference of listener position from the sweet spot; and rendering reverberated output signals using the loudspeaker signals and the gain and delay parameters. Figures 6a to 6d show examples of rendering reverberator outputs from constant user relative positions in adaptive loudspeaker rendering. For example, in Figure 6a, a reverb output direction 0rev, (prev, 601 is depicted. In Figure 6b, the reverberation output direction 605 is shown to be relative to the real listener position 603 within the listening space with loudspeakers. In Figure 6c, the real listener position p 603, the reverb output direction 0rev, (prev 605 relative to the real listener position 603, the sweet spot s 607 and the loudspeakers 609 having directions 0is(j), <pis(J) relative to the sweet spot 607 are depicted. In Figure 6d, the loudspeaker directions 0iSiust(j), (pis,ust(J) 611 with respect to the listener position p are depicted 603, along with the reverberation output position 605 with respect to the listener position 603 is shown. Adaptive loudspeaker panning is then performed such that the loudspeaker directions are updated to be the listener relative ones diSiust(J), <pis,ust(D 611. With respect to Figure 7 is shown a schematic view of an example adaptive loudspeaker renderer 205. The adaptive loudspeaker renderer 205 in some embodiments comprises a loudspeaker position calculator 701. The loudspeaker position calculator 701 is configured to calculate the position of the loudspeakers with regard to user position in the real scene. Furthermore, in some embodiments the adaptive loudspeaker renderer 205 comprises a relative reverberator output generator 703. The relative reverberator output generator 703 is configured to generate a reverberation output which is relative to the user position in the real scene. Additionally, in some embodiments the adaptive loudspeaker renderer 205 comprises a reverberated audio signal renderer 705. The reverberated audio signal renderer 705 is configured to render a reverberated signal from the user relative position using adaptive loudspeaker rendering with loudspeaker positions relative to the user position. Figure 8 shows example operations of the adaptive loudspeaker renderer 205. The first operation is obtaining a user position in the real scene, reverberation output position, loudspeaker positions, and reverberated audio signal as shown by 801. Then is shown determining or calculating a position of loudspeakers with regard to user position in the real scene as shown by 803. Following this is generating reverberation output relative to the user position in the real scene as shown by 805. Then is shown the rendering of the reverberated signal from the user relative position using adaptive loudspeaker rendering with loudspeaker positions relative to the user position as shown by 807. Figures 9a to 9d show examples of rendering reverberator outputs where the reverberation outputs are rendered as loudspeaker signals using adaptive loudspeaker rendering. For example in Figure 9a, a reverb output direction 0rev, (prev, 901 is depicted. In Figure 9b, the reverberation output direction 905 is shown to be relative to the real listener position 903 within the listening space with loudspeakers. In Figure 9c, the real listener position p 903, the reverb output direction 0rev, (prev 905 relative to the real listener position 903, and the loudspeakers 909 having directions 0is(j), <pis(J) relative to the real listener position 903 are depicted. Figures 7 and 10 further show a further embodiment implementation of the adaptive loudspeaker renderer 205. In Figure 7 as shown by the dashed line is also a direction mapper configured to map the direction of the reverberation output to the nearest loudspeaker direction. Figure 10 shows example operations of the adaptive loudspeaker renderer 205 with the mapping operation. The first operation is obtaining a user position in the real scene, reverberation output position, loudspeaker positions, and reverberated audio signal as shown by 801. Then is shown determining or calculating a position of loudspeakers with regard to user position in the real scene as shown by 803. Following this is generating reverberation output relative to the user position in the real scene as shown by 805. Then is shown the mapping of the direction of the reverberation output to the nearest loudspeaker as shown by 1006. Then is shown the rendering of the reverberated signal from the loudspeaker mapped reverberation output position using adaptive loudspeaker rendering with loudspeaker positions relative to the user position as shown by 1007. In Figure 9d, the loudspeaker directions 0iSiust(j), 911 with respect to the listener position p are depicted 903, along with the reverberation output position 915 coinciding with direction of a physical loudspeaker and with respect to the listener position 903 is shown. The benefit of the mapping the reverberation output direction 6rev, (prev such that it matches with the closest physical loudspeaker direction from the listener position is that no panning between loudspeakers is then required and has the benefit that no extra coherence is created to the rendered reverberation signal. Based on the listener relative loudspeaker setup 0iSiust(J), <Pis,ust(j) and the user-relative reverb output directions setup 0rev(i), 0rev(O, amplitude panning gains g(i) for each of the directions in 0rev(i), <prev(j) are obtained. The amplitude panning gains g, are obtained for three loudspeakers corresponding to unit vectors / 1,1, / 2,1, / 3,1 (the vector base from real listener position) forming a loudspeaker triangle (triplet) enclosing each the direction 0rev(i), 0rev(i). A column unit vector v is formulated pointing towards the desired amplitude panning direction, and a vector g containing the amplitude panning gains can be solved by a matrix multiplication gT = VT I7T1 *1 JT l2 ]T -1 where the reverberator output channel index i is omitted for clarity, and where T denotes transpose and -1 the matrix inverse. After formulating gains g, their overall level is normalized such that for the final gains the energy sum gTg = 1. The above example uses Vector-Base Amplitude Panning. Alternatively, in some embodiments, other methods of panning can be used. It is noted that there is difference in these embodiments compared to conventional VBAP in that the loudspeaker unit vectors are constantly, and repeatedly recalculated based on the listener position. Since the listener-to-loudspeaker distance also changes, the loudspeaker vectors are made unit vectors by dividing by their length after each user position update. In some embodiments where reverberator output directions are mapped to the direction of the closest loudspeaker to obtain 0rer_is(O, <prev,is(.Q> the gain g^i = 1 for the mapped loudspeaker Is and the gain for other speakers is 0. This maps the output of each reverberator channel to a single speaker only. When the number of reverberator channels is larger than the number of speakers then each speaker is mapped multiple reverberator channels. Since the loudspeaker to which a reverberation output is mapped can change, smoothing of the gain values can be performed, in some embodiments, to prevent switching artefacts. The smoother can be an HR filter or other suitable smoothing mechanism. In some embodiments, switching of several reverberation outputs at the same time (rendering frame) is not allowed, but only one (or predetermined number) of reverberation outputs can change the target loudspeaker at once, and others are queued for changing later. With the gains g, the output signal corresponding to the unit vector A.i for the reverberator output channel i can be obtained as ^Is^’ 0 — 91,i^rev^> 0 Then, the loudspeaker signals for the different loudspeaker channels j are summed by yielding the reverberant loudspeaker signals s^njy Adaptive loudspeaker rendering can also involve receiving the user position p and calculating additional gain and delay parameters y(f),8(f) for each loudspeaker j based on the difference of the listener position and sweet spot p-s. The goal of the gain parameter is to compensate for the difference in distance gain attenuation due to different listener-to-loudspeaker distance compared to the sweet spot, and the goal of the delay parameter is to compensate for the difference in sound propagation delay. The distance from the user position p to a loudspeaker lj is ||s + - p|| . The difference of the distance compared to the distance from the sweet spot ||s + lj\\ = 1 indicates the necessary delay 8(j") to be applied: <5(j) = 1 fs, where c=343m / s and fs=48000Hz. For example, if the distance from the user position to loudspeaker lj is ||s + lj - p|| = 1.1, then the additional delay needed is 5( / ) =* 48000 « 13.9942 samples. Such an additional delay can be implemented with a variable delay line (VDL). In a free field, the sound level attenuates with 1 / r, where r is the distance traveled by the sound. The difference in distance gain for speaker lj at position p can be compensated with y(f) = —1—+ ij — p|| assuming free field conditions. Thus, if ||s + lj - p|| = 1.1 then the gain to be applied to lj signal is y(j) = 1.1. Since the method is commonly applied in rooms which contain reflections and reverberation due to the real space acoustics, the sound level attenuates slower than 1 / r. Such a behaviour can be modelled by configuring the system to apply less level compensation depending on the room reverberation time, for example. A decay parameter can be provided as a input to the LSDF for the renderer to control this behaviour. The output of the adaptive loudspeaker rendering are the signals sis,adaptiVe(.n,D = y(j)sls(n - 8(jyjy which are the output of the system. Figures 11a and 11b show schematically an example system where the embodiments are implemented in an encoder device 1901 which performs part of the functionality; writes data into a bitstream 1921 and transmits that for a renderer device 1941, which decodes the bitstream, performs reverberator processing according to the embodiments and outputs audio for headphone listening. The encoder side 1901 of Figure 11 can be performed on content creator computers and / or network server computers. The output of the encoder is the bitstream 1921 which is made available for downloading or streaming. The decoder / renderer 1941 functionality runs on an end-user-device, which can be a mobile device, personal computer, sound bar, tablet computer, car media system, home HiFi or theatre system, head mounted display for AR or VR, smart watch, or any suitable system for audio consumption. The encoder 1901 is configured to receive the virtual scene description 1900 and the audio signals 1904. The virtual scene description 1900 can be provided in the MPEG-I encoder input format (EIF) or in another suitable format. Generally, the virtual scene description contains an acoustically relevant description of the contents of the virtual scene, and contains, for example, the scene geometry as a mesh or as voxels, acoustic materials, acoustic environments with reverberation parameters, positions of sound sources, and other audio element related parameters such as whether reverberation is to be rendered for an audio element or not. The encoder 1901 in some embodiments comprises a scene and reverberation payload encoder 1913 configured to generate reverberation parameters. The encoder 1901 further comprises a MPEG-H 3D audio encoder 1914 configured to obtain the audio signals 1904 and MPEG-H encode them and pass them to a bitstream encoder 1915. The encoder 1901 furthermore in some embodiments comprises a bitstream encoder 1915 which is configured to receive the output of the scene and reverberation payload encoder 1913 and the encoded audio signals from the MPEG-H encoder 1914 and generate the bitstream 1921 which can be passed to the bitstream decoder 1941. The bitstream 1921 in some embodiments can be streamed to end-user devices or made available for download or stored. In other implementations, the audio signals 1904 may be encoded via any suitable codec such as EVS, IVAS, AAC, OPUS, etc. The decoder 1941 in some embodiments comprises a bitstream decoder 1951 configured to decode the bitstream. The decoder 1941 further can comprise a scene payload decoder 1953 configured to obtain the encoded reverberation parameters and decode these in an opposite or inverse operation to the reverberation payload encoder 1913. The reverberator controller 203 / 1952 is configured to receive the decoded reverberation parameters and room dimensions and loudspeaker configuration or setup 1950 information and generate the reverberator control parameters discussed herein. Furthermore, the head pose generator 1957 receives information from a head mounted device 1970 or similar and generates head pose information or parameters which can be passed to the adaptive loudspeaker renderer 205 / 1967. The decoder 1941 comprises MPEG-H 3D audio decoder 1954 which is configured to decode the audio signals and pass them to the reverberator 201 / 1961, early reflection renderer 1999 and direct sound processing 1965. In other implementations, the MPEG-H 3D audio decoder 1954 may be replaced by any suitable audio decoder depending on the audio coding method utilized. The decoder 1941 furthermore comprises a reverberator 201 / 1961 configured to implement a suitable reverberation of the audio signals from the MPEG-H 3D audio decoder 1954. The output of the reverberator 201 / 1961 is configured to output reverberated audio based on the reverberator parameters to an adaptive loudspeaker renderer 205 / 1959. The decoder 1941 furthermore comprises an early reflection renderer 1999 configured to obtain the output of the MPEG-H 3D audio decoder 1954 and generate early reflections using, for example, geometric acoustic rendering with image sources based on the physical or virtual scene geometry and pass these to the adaptive loudspeaker renderer 205 / 1967. The decoder further comprises a direct sound processor 1965 configured to apply direct sound processing such as air absorption, distance-gain attenuation and pass the results to a direct sound renderer 1963. The decoder 1941 further comprises a direct sound renderer 1963 configured to render direction sound values and pass these to the adaptive loudspeaker renderer 205 / 1967. The decoder 1941 furthermore comprises the adaptive loudspeaker renderer 205 / 1967 configured to generate loudspeaker audio signals. In some embodiments, the reverberation parameters of the real room are provided as RDRreal(k), tpre>real. In some embodiments, when rendering the reverberation for an AR scene, the rendering of reverberation can be omitted as the real room already contains reverberation of the space. This omitting can be controlled by bitstream signalling. In some embodiments, the real room reverberation can be compensated when rendering virtual reverberation with parameters T60(fc), RDR^k), tpre. Additionally in some embodiments, the compensation is implemented by modifying the reverberation parameters to be T60iComp( / c) = max (0,T60( / c)-T&0,realW), RDRcomp(k) = max (0,RDR(k)- RDRreal(_k)) and tpre,comp max(0,tpre-tpre,real) ■ The bitstream signaling for configuration of reverb rendering for loudspeakers: Semantics: payloadReverbForLSPresent Indicates presence of payload for controlling reverb rendering to physical loudspeakers. A value equal to 1 indicates presence and a value equal to 0 indicates absence of bitstream information for controlling reverb rendering to physical loudspeakers. reverbForLSParamsPresent Indicates the presence of control parameters for reverb rendering to physical LS isARScene true if the scene is an AR scene, false for VR scene thresholdReverbPresent indicates the presence of reverberation threshold data performDecorrelation toggle applying additional decorrelation to reverberation signals numReverbDelayLinesPresent indicates presence of reverb delay lines for loudspeaker rendering reverbLSMethod indicates method for reverb rendering to LS reverberationThresholdStruct indicates the threshold RT60 and RDR data such that if the environment has reverberation time larger than threshold and RDR larger than threshold no reverb rendering from physical loudspeakers is performed. This can be applied to both AR and VR scenes. class aligned(8) pay load ReverbQ{ unsigned int(1) payloadReverbForLSPresent; if(payloadReverbForLSPresent){ unsigned int(1) reverbForLSParamsPresent; unsigned int(1) isARScene; if(isARScene){ unsigned int(1) thresholdReverbPresent; unsigned int(1) compensateRoomReverb; if(thresholdReverbPresent) reverberationThresholdStructQ; } if(reverbForLSParamsPresent){ unsigned int(1) performDecorrelation; unsigned int(1) numReverbDelayLinesPresent; unsigned int(4) reverbLSMethod; if(numReverbDelayLinesPresent) unsigned int(8) numDelayLines; / / 3, 7, 15, 31,63, 128 reverberationThresholdStructO; } } aligned(8) reverberationThresholdStructO! unsigned int(1) reverbThresholdPresent; <any other parameter for reverberation threshold> if(reverbParamsStruct){ EncodedRT60StructO; EncodedRDRStructO; } } aligned(8) EncodedRT60StructO{ unsigned int(8) num_frequency_bands; for (i=O;i<num_frequency_bands;i++){ unsigned int(32) frequency_encoded_value; unsigned int(32) scene_rt60_encoded_value; } } aligned(8) EncodedRDRStruct(){ unsigned int(8) num_frequency_bands; for (i=O;i<num_frequency_bands;i++){ unsigned int(32) frequency_encoded_value; unsigned int(32) scene_rdr_encoded_value; / / represented as logRDR, Huffman coded } } reverbLSMethod Value Reserved 0 This indicates that the reverb is rendered as listener relative position compared to virtual loudspeakers and perform panning to nearest physical loudspeakers 1 This indicates that the reverb is rendered as listener relative position to the virtual loudspeakers that are aligned with the physical loudspeakers (direction mapped to the closest) 2 Reserved 3-15 In embodiments when additional decorrelation is to be performed on the reverberation signals, each of the reverberated signals will be run through a decorrelator. The decorrelator can be a known decorrelator used in MPEG audio codecs, or another type of decorrelator such as a velvet noise decorrelator or any suitable decorrelator producing sufficient amount of decorrelation while not introducing significant spectral coloration. In implementation embodiments related to AR scenes, loudspeaker rendering of reverberation is performed only when the loudspeaker rendering of the reverberation does not increase the reverberation level. In implementation embodiments related to VR scenes, the loudspeaker rendering of reverberation only if the room reverberation introduced by the loudspeaker output is not sufficient or aligned with the VR scene content creator requirements. Consequently, an additional data structure carrying the threshold reverberation information is included in the bitstream. In an embodiment, the bitstream can also carry a flag to indicate that loudspeaker rendering of reverberation may be skipped in order to reduce computational complexity or if the reproduction environment is reverberant or a combination of both. With respect to Figure 13 an example electronic device which may be used as any of the apparatus parts of the system as described above. The device may be any suitable electronics device or apparatus. For example, in some embodiments the device 2000 is a mobile device, user equipment, tablet computer, computer, audio playback apparatus, etc. The device may for example be configured to implement the encoder or the renderer or any functional block as described above. In some embodiments the device 2000 comprises at least one processor or central processing unit 2007. The processor 2007 can be configured to execute various program codes such as the methods described herein. In some embodiments the device 2000 comprises a memory 2011. In some embodiments the at least one processor 2007 is coupled to the memory 2011. The memory 2011 can be any suitable storage means. In some embodiments the memory 2011 comprises a program code section for storing program codes implementable upon the processor 2007. Furthermore, in some embodiments the memory 2011 can further comprise a stored data section for storing data, for example data that has been processed or to be processed in accordance with the embodiments as described herein. The implemented program code stored within the program code section and the data stored within the stored data section can be retrieved by the processor 2007 whenever needed via the memory-processor coupling. In some embodiments the device 2000 comprises a user interface 2005. The user interface 2005 can be coupled in some embodiments to the processor 2007. In some embodiments the processor 2007 can control the operation of the user interface 2005 and receive inputs from the user interface 2005. In some embodiments the user interface 2005 can enable a user to input commands to the device 2000, for example via a keypad. In some embodiments the user interface 2005 can enable the user to obtain information from the device 2000. For example, the user interface 2005 may comprise a display configured to display information from the device 2000 to the user. The user interface 2005 can in some embodiments comprise a touch screen or touch interface capable of both enabling information to be entered to the device 2000 and further displaying information to the user of the device 2000. In some embodiments the user interface 2005 may be the user interface for communicating. In some embodiments the device 2000 comprises an input / output port 2009. The input / output port 2009 in some embodiments comprises a transceiver. The transceiver in such embodiments can be coupled to the processor 2007 and configured to enable a communication with other apparatus or electronic devices, for example via a wireless communications network. The transceiver or any suitable transceiver or transmitter and / or receiver means can in some embodiments be configured to communicate with other electronic devices or apparatus via a wire or wired coupling. The transceiver can communicate with further apparatus by any suitable known communications protocol. For example, in some embodiments the transceiver can use a suitable universal mobile telecommunications system (UMTS) protocol, a wireless local area network (WLAN) protocol such as for example IEEE 802.X, a suitable short-range radio frequency communication protocol such as Bluetooth, or infrared data communication pathway (IRDA). The input / output port 2009 may be configured to receive the signals. In some embodiments the device 2000 may be employed as at least part of the renderer. The input / output port 2009 may be coupled to headphones (which may be a headtracked or a non-tracked headphones) or similar. In general, the various embodiments of the invention may be implemented in hardware or special purpose circuits, software, logic or any combination thereof. For example, some aspects may be implemented in hardware, while other aspects may be implemented in firmware or software which may be executed by a controller, microprocessor or other computing device, although the invention is not limited thereto. While various aspects of the invention may be illustrated and described as block diagrams, flow charts, or using some other pictorial representation, it is well understood that these blocks, apparatus, systems, techniques or methods described herein may be implemented in, as non-limiting examples, hardware, software, firmware, special purpose circuits or logic, general purpose hardware or controller or other computing devices, or some combination thereof. The embodiments of this invention may be implemented by computer software executable by a data processor of the mobile device, such as in the processor entity, or by hardware, or by a combination of software and hardware. Further in this regard it should be noted that any blocks of the logic flow as in the Figures may represent program steps, or interconnected logic circuits, blocks and functions, or a combination of program steps and logic circuits, blocks and functions. The software may be stored on such physical media as memory chips, or memory blocks implemented within the processor, magnetic media such as hard disk or floppy disks, and optical media such as for example DVD and the data variants thereof, CD. The memory may be of any type suitable to the local technical environment and may be implemented using any suitable data storage technology, such as semiconductor-based memory devices, magnetic memory devices and systems, optical memory devices and systems, fixed memory and removable memory. The data processors may be of any type suitable to the local technical environment, and may include one or more of general-purpose computers, special purpose computers, microprocessors, digital signal processors (DSPs), application specific integrated circuits (ASIC), gate level circuits and processors based on multi-core processor architecture, as non-limiting examples. Embodiments of the inventions may be practiced in various components such as integrated circuit modules. The design of integrated circuits is by and large a highly automated process. Complex and powerful software tools are available for converting a logic level design into a semiconductor circuit design ready to be etched and formed on a semiconductor substrate. Programs, such as those provided by Synopsys, Inc. of Mountain View, California and Cadence Design, of San Jose, California automatically route conductors and locate components on a semiconductor chip using well established rules of design as well as libraries of pre-stored design modules. Once the design for a semiconductor circuit has been completed, the resultant design, in a standardized electronic format (e.g., Opus, GDSII, or the like) may be transmitted to a semiconductor fabrication facility or "fab" for fabrication. As used in this application, the term “circuitry” may refer to one or more or all of the following: (a) hardware-only circuit implementations (such as implementations in only analog and / or digital circuitry) and (b) combinations of hardware circuits and software, such as (as applicable): (i) a combination of analog and / or digital hardware circuit(s) with software / firmware and (ii) any portions of hardware processor(s) with software (including digital signal processor(s)), software, and memory(ies) that work together to cause an apparatus, such as a mobile phone or server, to perform various functions) and I hardware circuit(s) and or processor(s), such as a microprocessor(s) or a portion of a microprocessor(s), that requires software (e.g., firmware) for operation, but the software may not be present when it is not needed for operation. This definition of circuitry applies to all uses of this term in this application, including in any claims. As a further example, as used in this application, the term circuitry also covers an implementation of merely a hardware circuit or processor (or multiple processors) or portion of a hardware circuit or processor and its (or their) accompanying software and / or firmware. The term circuitry also covers, for example and if applicable to the particular claim element, a baseband integrated circuit or processor integrated circuit for a mobile device or a similar integrated circuit in server, a cellular network device, or other computing or network device. The term “non-transitory,” as used herein, is a limitation of the medium itself (i.e., tangible, not a signal) as opposed to a limitation on data storage persistency (e.g., RAM vs. ROM). As used herein, “at least one of the following: ” 5 and “at least one of ” and similar wording, where the list of two or more elements are joined by “and” or “or”, mean at least any one of the elements, or at least any two or more of the elements, or at least all the elements The foregoing description has provided by way of exemplary and non-limiting 10 examples a full and informative description of the exemplary embodiment of this invention. However, various modifications and adaptations may become apparent to those skilled in the relevant arts in view of the foregoing description, when read in conjunction with the accompanying drawings and the appended claims. However, all such and similar modifications of the teachings of this invention will 15 still fall within the scope of this invention as defined in the appended claims.

Claims

1. A method for rendering reverberation, the method comprising:obtaining a reverberated signal based on a listener position within an acoustic environment;obtaining a listener position within a reproduction space;obtaining at least one reverberator output channel position;obtaining a position of at least one loudspeaker based on the at least one reverberator output channel position and the listener position within the reproduction space;determining a gain based on the position of the at least one loudspeaker, the listener position within the reproduction space, and the reverberator output channel position;applying the gain on the reverberated signal; andgenerating a loudspeaker signal to be reproduced from the at least one loudspeaker based on the applied gain on the reverberated signal.

2. The method as claimed in claim 1, wherein obtaining a reverberated signal based on a listener position within an acoustic environment comprises:obtaining at least one reverberation parameter associated with the acoustic environment;configuring at least one reverberator based on the at least one reverberation parameter; andprocessing at least one audio signal with the configured reverberator to produce the reverberated signal.

3. The method as claimed in claim 2, wherein obtaining at least one reverberation parameter associated with the acoustic environment comprises determining at least one delay line parameter.

4. The method as claimed in any of claims 1 to 3, wherein obtaining the listener position within the reproduction environment comprises adaptively obtaining the listener position, the listener position being time-varying.

5. The method as claimed in claim 4, comprising obtaining based on the adaptively obtained listener position, at least one of:at least one gain; andat least one delay.

6. The method as claimed in any of claims 2, 4 or 5, or claim 4 when dependent on claim 2, wherein configuring at least one reverberator based on the at least one reverberation parameter comprises selectively or partially disabling the at least one reverberator based on determining the reproduction space within which the at least one loudspeaker is located is reverberant.

7. The method as claimed in claim 6, wherein selectively or partially disabling the at least one reverberator based on one of:a user input to selectively or partially disable the at least one reverberator; anda flag or indicator, the flag or indicator configured to indicate that loudspeaker rendering of reverberation is to be skipped in order to reduce computational complexity, or if the reproduction environment is reverberant, or a combination of both.

8. The method as claimed in any of claims 1 to 7, wherein applying the gain on the one reverberated signal comprises gain on the reverberated signal to produce a panned audio signal.

9. The method as claimed in claim 8, wherein generating a loudspeaker signal to be reproduced from the at least one loudspeaker based on the applied gain on the reverberated signal comprises summing the panned audio signal to generate a loudspeaker signal to be reproduced from the at least one loudspeaker.

10. The method as claimed in claim 9, wherein summing the panned audio signal to generate a loudspeaker signal to be reproduced from the at least one loudspeaker comprises summing the panned audio signal for a sub-set of pannedaudio signal to generate the loudspeaker signal to be reproduced from the at least one loudspeaker.

11. The method as claimed in any of claims 1 to 10, further comprising: determining at least one distance between the listener position and the position of the at least one loudspeaker within the reproduction space;determining a further gain based on the determined at least one distance; andrendering the loudspeaker signal from the at least one loudspeaker by applying the further gain to the loudspeaker signal.

12. The method as claimed in any of claims 1 to 11, wherein obtaining at least one reverberator output channel position comprises adaptively controlling the at least one reverberator output channel position.

13. The method as claimed in claim 12, wherein adaptively controlling the at least one reverberator output channel position comprises mapping the at least one reverberator output channel position to a nearest position of the at least one loudspeaker.

14. The method as claimed in any of claims 1 to 13, wherein the acoustic environment is the same as the reproduction space in an augmented reality application.

15. The method as claimed in any of claims 1 to 13, wherein the acoustic environment is a virtual reality environment and the reproduction space is a real space within which the at least one loudspeaker is located.

16. An apparatus comprising means for performing the method of any of claims 1 to 15.

17. A computer program comprising instructions, which, when executed by an apparatus, cause the apparatus to perform the method of any of claims 1 to 15.

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