Patent Description:
This invention relates to a method and a device for rendering an audio soundfield representation, and in particular an Ambisonics formatted audio representation, for audio playback.

Accurate localisation is a key goal for any spatial audio reproduction system. Such reproduction systems are highly applicable for conference systems, games, or other virtual environments that benefit from 3D sound. Sound scenes in 3D can be synthesised or captured as a natural sound field. Soundfield signals such as e.g. Ambisonics carry a representation of a desired sound field. The Ambisonics format is based on spherical harmonic decomposition of the soundfield. While the basic Ambisonics format or B-format uses spherical harmonics of order zero and one, the so-called Higher Order Ambisonics (HOA) uses also further spherical harmonics of at least <NUM>nd order. A decoding or rendering process is required to obtain the individual loudspeaker signals from such Ambisonics formatted signals. The spatial arrangement of loudspeakers is referred to as loudspeaker setup herein. However, while known rendering approaches are suitable only for regular loudspeaker setups, arbitrary loudspeaker setups are much more common. If such rendering approaches are applied to arbitrary loudspeaker setups, sound directivity suffers. <NPL>, discloses a three dimensional spatial sound reproduction using irregular loudspeaker layouts.

The present invention describes a method for rendering/decoding an audio sound field representation for both regular and non-regular spatial loudspeaker distributions, where the rendering/decoding provides highly improved localization properties and is energy preserving. In particular, the invention provides a new way to obtain the decode matrix for sound field data, e.g. in HOA format. Since the HOA format describes a sound field, which is not directly related to loudspeaker positions, and since loudspeaker signals to be obtained are necessarily in a channel-based audio format, the decoding of HOA signals is always tightly related to rendering the audio signal. Therefore the present invention relates to both decoding and rendering sound field related audio formats.

One advantage of the present invention is that energy preserving decoding with very good directional properties is achieved. The term "energy preserving" means that the energy within the HOA directive signal is preserved after decoding, so that e.g. a constant amplitude directional spatial sweep will be perceived with constant loudness. The term "good directional properties" refers to the speaker directivity characterized by a directive main lobe and small side lobes, wherein the directivity is increased compared with conventional rendering/decoding.

The invention discloses rendering sound field signals, Higher-Order Ambisonics (HOA), for arbitrary loudspeaker setups, where the rendering results in highly improved localization properties and is energy preserving. This is obtained by a new type of decode matrix for sound field data, and a new way to obtain the decode matrix. In a method for rendering an audio sound field representation for arbitrary spatial loudspeaker setups, the decode matrix for the rendering to a given arrangement of target loudspeakers is obtained by steps of obtaining a number of target speakers and their positions, positions of a spherical modeling grid and a HOA order, generating a mix matrix from the positions of the modeling grid and the positions of the speakers, generating a mode matrix from the positions of the spherical modeling grid and the HOA order, calculating a first decode matrix from the mix matrix and the mode matrix, and smoothing and scaling the first decode matrix with smoothing and scaling coefficients to obtain an energy preserving decode matrix.

In one embodiment, the invention relates to a method for generating a decode matrix used for rendering a Higher-Order Ambisonics audio sound field representation for audio playback as claimed in claim <NUM>. In another embodiment, the invention relates to a device for for generating a decode matrix used for rendering a Higher-Order Ambisonics sound field representation for audio playback as claimed in claim <NUM>. In yet another embodiment, the invention relates to a computer readable medium having stored on it executable instructions to cause a computer to perform a method according to claim <NUM>.

Generally, the invention uses the following approach. First, panning functions are derived that are dependent on a loudspeaker setup that is used for playback. Second, a decode matrix (e.g. Ambisonics decode matrix) is computed from these panning functions (or a mix matrix obtained from the panning functions) for all loudspeakers of the loudspeaker setup. In a third step, the decode matrix is generated and processed to be energy preserving. Finally, the decode matrix is filtered in order to smooth the loudspeaker panning main lobe and suppress side lobes. The filtered decode matrix is used to render the audio signal for the given loudspeaker setup. Side lobes are a side effect of rendering and provide audio signals in unwanted directions. Since the rendering is optimized for the given loudspeaker setup, side lobes are disturbing. It is one of the advantages of the present invention that the side lobes are minimized, so that directivity of the loudspeaker signals is improved.

Further objects, features and advantages of the invention will become apparent from a consideration of the following description and the appended claims when taken in connection with the accompanying drawings.

Exemplary embodiments of the invention are described with reference to the accompanying drawings, which show in.

In general, the invention relates to rendering (i.e. decoding) sound field formatted audio signals such as Higher Order Ambisonics (HOA) audio signals to loudspeakers, where the loudspeakers are at symmetric or asymmetric, regular or non-regular positions. The audio signals may be suitable for feeding more loudspeakers than available, e.g. the number of HOA coefficients may be larger than the number of loudspeakers. The invention provides energy preserving decode matrices for decoders with very good directional properties, i.e. speaker directivity lobes generally comprise a stronger directive main lobe and smaller side lobes than speaker directivity lobes obtained with conventional decode matrices. Energy preserving means that the energy within the HOA directive signal is preserved after decoding, so that e.g. a constant amplitude directional spatial sweep will be perceived with constant loudness.

<FIG> shows a flow-chart of a method according to one embodiment of the invention. In this embodiment, the method for rendering (i.e. decoding) a HOA audio sound field representation for audio playback uses a decode matrix that is generated as follows: first, a number L of target loudspeakers, the positions <IMG> of the loudspeakers, a spherical modeling grid <IMG> and an order N (e.g. HOA order) are determined <NUM>. From the positions <IMG> of the speakers and the spherical modeling grid <IMG>, a mix matrix G is generated <NUM>, and from the spherical modeling grid <IMG> and the HOA order N, a mode matrix Ψ̃ is generated <NUM>. A first decode matrix D̂ is calculated <NUM> from the mix matrix G and the mode matrix Ψ̃. The first decode matrix D̂ is smoothed <NUM> with smoothing coefficients <IMG>, wherein a smoothed decode matrix D̃ is obtained, and the smoothed decode matrix D̃ is scaled <NUM> with a scaling factor obtained from the smoothed decode matrix D̃, wherein the decode matrix D is obtained. In one embodiment, the smoothing <NUM> and scaling <NUM> is performed in a single step.

In one embodiment, the smoothing coefficients <IMG> are obtained by one of two different methods, depending on the number of loudspeakers L and the number of HOA coefficient channels O3D=(N+<NUM>)<NUM>. If the number of loudspeakers L is below the number of HOA coefficient channels O3D , a new method for obtaining the smoothing coefficients is used.

In one embodiment, a plurality of decode matrices corresponding to a plurality of different loudspeaker arrangements are generated and stored for later usage. The different loudspeaker arrangements can differ by at least one of the number of loudspeakers, a position of one or more loudspeakers and an order N of an input audio signal. Then, upon initializing the rendering system, a matching decode matrix is determined, retrieved from the storage according to current needs, and used for decoding.

In one embodiment which does not from part of the claimed invention, the decode matrix D is obtained by performing a compact singular value decomposition of the product of the mode matrix Ψ̃ with the Hermitian transposed mix matrix GH according to U S VH = Ψ̃GH, and calculating a first decode matrix D̂ from the matrices U,V according to D̂ = V UH. The U,V are derived from Unitary matrices, and S is a diagonal matrix with singular value elements of said compact singular value decomposition of the product of the mode matrix Ψ̃ with the Hermitian transposed mix matrix GH. Decode matrices obtained according to this embodiment are often numerically more stable than decode matrices obtained with an alternative embodiment described below. The Hermitian transposed of a matrix is the conjugate complex transposed of the matrix.

In the alternative embodiment which does not from part of the claimed invention, the decode matrix D is obtained by performing a compact singular value decomposition of the product of the Hermitian transposed mode matrix Ψ̃H with the mix matrix G according to U S VH = GΨ̃H, wherein a first decode matrix is derived by D̂ = U VH.

In one embodiment which does not from part of the claimed invention, a compact singular value decomposition is performed on the mode matrix Ψ̃ and mix matrix G according to U S VH = GΨ̃H , where a first decode matrix is derived by D̂ = U Ŝ VH, where Ŝ is a truncated compact singular value decomposition matrix that is derived from the singular value decomposition matrix S by replacing all singular values larger or equal than a threshold thr by ones, and replacing elements that are smaller than the threshold thr by zeros. The threshold thr depends on the actual values of the singular value decomposition matrix and may be, exemplarily, in the order of <NUM>,<NUM> * S<NUM> (the maximum element of S).

In one embodiment, a compact singular value decomposition is performed on the mode matrix Ψ̃ and mix matrix G according to V S UH = GΨ̃H , where a first decode matrix is derived by D̂ = V Ŝ UH. The Ŝ and threshold thr are as described above for the previous embodiment. The threshold thr is usually derived from the largest singular value.

In one embodiment, two different methods for calculating the smoothing coefficients are used, depending on the HOA order N and the number of target speakers L: if there are less target speakers than HOA channels, i.e. if O3D = (N<NUM>+<NUM>) > L, the smoothing and scaling coefficients <IMG> corresponds to a conventional set of max rE coefficients that are derived from the zeros of the Legendre polynomials of order N + <NUM>; otherwise, if there are enough target speakers, i.e. if O3D = (N<NUM>+<NUM>) ≤ L, the coefficients of <IMG> are constructed from the elements <IMG> of a Kaiser window with len=(2N+<NUM>) and width=<NUM>N according to <IMG>= cf[<IMG>]T with a scaling factor cf. The used elements of the Kaiser window begin with the (N+<NUM>)st element, which is used only once, and continue with subsequent elements which are used repeatedly: the (N+<NUM>)nd element is used three times, etc..

In one embodiment, the scaling factor is obtained from the smoothed decoding matrix. In particular, in one embodiment it is obtained according to <MAT>.

In the following, a full rendering system is described. A major focus of the invention is the initialization phase of the renderer, where a decode matrix D is generated as described above. Here, the main focus is a technology to derive the one or more decoding matrices, e.g. for a code book. For generating a decode matrix, it is known how many target loudspeakers are available, and where they are located (i.e. their positions).

<FIG> shows a flow-chart of a method for building the mix matrix G, according to one embodiment of the invention. In this embodiment, an initial mix matrix with only zeros is created <NUM>, and for every virtual source s with an angular direction Ωs = [θs,ϕs]T and radius rs, the following steps are performed. First, three loudspeakers l<NUM>,l<NUM>,l<NUM> are determined <NUM> that surround the position <MAT>, wherein unit radii are assumed, and a matrix R = [rl<NUM>, rl<NUM> , rl<NUM>] is built <NUM>, with <MAT>. The matrix R is converted <NUM> to Cartesian coordinates, according to Lt = spherical_to_cartesian(R). Then, a virtual source position is built <NUM> according to s = (sin Θs cos ϕs, sin Θs sin ϕs, cos Θs)T, and a gain g is calculated <NUM> according to g = Lt-<NUM> s, with g = (gl<NUM>, gl<NUM>, gl<NUM>)T. The gain is normalized <NUM> according to g = g/∥ g ∥<NUM>, and the corresponding elements Gl,s of G are replaced with the normalized gains: Gl<NUM>,s = gl<NUM>, Gl<NUM>,s = gl<NUM>, Gl<NUM>,s = gl<NUM>.

The following section gives a brief introduction to Higher Order Ambisonics (HOA) and defines the signals to be processed, i.e. rendered for loudspeakers.

Higher Order Ambisonics (HOA) is based on the description of a sound field within a compact area of interest, which is assumed to be free of sound sources. In that case the spatiotemporal behavior of the sound pressure p(t, x) at time t and position x = [r, θ, ϕ]T within the area of interest (in spherical coordinates: radius r, inclination θ, azimuth ϕ) is physically fully determined by the homogeneous wave equation. It can be shown that the Fourier transform of the sound pressure with respect to time, i.e., <MAT> where ω denotes the angular frequency (and <IMG> { } corresponds to <MAT>, may be expanded into the series of Spherical Harmonics (SHs) according to [<NUM>]: <MAT>.

(<NUM>), cs denotes the speed of sound and <MAT> the angular wave number. Further, jn(·) indicate the spherical Bessel functions of the first kind and order n and <MAT> denote the Spherical Harmonics (SH) of order n and degree m. The complete information about the sound field is actually contained within the sound field coefficients <MAT>.

It should be noted that the SHs are complex valued functions in general. However, by an appropriate linear combination of them, it is possible to obtain real valued functions and perform the expansion with respect to these functions.

Related to the pressure sound field description in eq. (<NUM>) a source field can be defined as: <MAT> with the source field or amplitude density [<NUM>] D(k cs,Ω) depending on angular wave number and angular direction Ω = [θ,ϕ]T. A source field can consist of far-field/ nearfield, discrete/continuous sources [<NUM>]. The source field coefficients <MAT> are related to the sound field coefficients <MAT> by, [<NUM>]: <MAT> where <MAT> is the spherical Hankel function of the second kind and rs is the source distance from the origin.

Signals in the HOA domain can be represented in frequency domain or in time domain as the inverse Fourier transform of the source field or sound field coefficients. The following description will assume the use of a time domain representation of source field coefficients: <MAT> of a finite number: The infinite series in eq. (<NUM>) is truncated at n = N. Truncation corresponds to a spatial bandwidth limitation. The number of coefficients (or HOA channels) is given by: <MAT> or by O<NUM>D = 2N + <NUM> for 2D only descriptions. The coefficients <MAT> comprise the Audio information of one time sample t for later reproduction by loudspeakers. They can be stored or transmitted and are thus subject of data rate compression. A single time sample t of coefficients can be represented by vector b(t) with O<NUM>D elements: <MAT> and a block of M time samples by matrix <MAT> <MAT>.

Two dimensional representations of sound fields can be derived by an expansion with circular harmonics. This is a special case of the general description presented above using a fixed inclination of <MAT>, different weighting of coefficients and a reduced set to O<NUM>D coefficients (m = ±n). Thus, all of the following considerations also apply to 2D representations; the term "sphere" then needs to be substituted by the term "circle".

In one embodiment, metadata is sent along the coefficient data, allowing an unambiguous identification of the coefficient data. All necessary information for deriving the time sample coefficient vector b(t) is given, either through transmitted metadata or because of a given context. Furthermore, it is noted that at least one of the HOA order N or O3D, and in one embodiment additionally a special flag together with rs to indicate a nearfield recording are known at the decoder.

Next, rendering a HOA signal to loudspeakers is described. This section shows the basic principle of decoding and some mathematical properties.

Basic decoding assumes, first, plane wave loudspeaker signals and, second, that the distance from speakers to origin can be neglected. A time sample of HOA coefficients b rendered to L loudspeakers that are located at spherical directions <MAT> with l = <NUM>,. , L can be described by [<NUM>]: <MAT> where <MAT> represents a time sample of L speaker signals and decode matrix <MAT>. A decode matrix can be derived by <MAT> where Ψ+ is the pseudo inverse of the mode matrix Ψ. The mode-matrix Ψ is defined as <MAT> with <MAT> and <MAT> consisting of the Spherical Harmonics of the speaker directions <MAT> where H denotes conjugate complex transposed (also known as Hermitian).

Next, a pseudo inverse of a matrix by Singular Value Decomposition (SVD) is described. One universal way to derive a pseudo inverse is to first calculate the compact SVD: <MAT> where <MAT> are derived from rotation matrices and S = diag(S<NUM>,. , SK) ∈ <MAT> is a diagonal matrix of the singular values in descending order S<NUM> ≥ S<NUM> ≥ ··· ≥ SK with K > <NUM> and K ≤ min(O<NUM>D,L). The pseudo inverse is determined by <MAT> where <MAT>. For bad conditioned matrices with very small values of Sk , the corresponding inverse values <MAT> are replaced by zero. This is called Truncated Singular Value Decomposition. Usually a detection threshold with respect to the largest singular value S<NUM> is selected to identify the corresponding inverse values to be replaced by zero.

In the following, the energy preservation property is described. The signal energy in HOA domain is given by <MAT> and the corresponding energy in the spatial domain by <MAT>.

The ratio Ê / E for an energy preserving decoder matrix is (substantially) constant. This can only be achieved if DHD = cI, with identity matrix I and constant <MAT>. This requires D to have a norm-<NUM> condition number cond(D) = <NUM>. This again requires that the SVD (Singular Value Decomposition) of D produces identical singular values: D = U S VH with S = diag(SK,.

Generally, energy preserving renderer design is known in the art. An energy preserving decoder matrix design for L ≥ O<NUM>D is proposed in [<NUM>] by <MAT> where Ŝ from eq. (<NUM>) is forced to be Ŝ = I and thus can be dropped in eq. The product DHD = U VHV UH = I and the ratio Ê / E becomes one. A benefit of this design method is the energy preservation which guarantees a homogenous spatial sound impression where spatial pans have no fluctuations in perceived loudness. A drawback of this design is a loss in directivity precision and strong loudspeaker beam side lobes for asymmetric, non-regular speaker positions (see <FIG>). The present invention can overcome this drawback.

Also a renderer design for non-regular positioned speakers is known in the art: In [<NUM>], a decoder design method for L ≥ O<NUM>D and L < O<NUM>D is described which allows rendering with high precision in reproduced directivity. A drawback of this design method is that the derived renderers are not energy preserving (see <FIG>).

Spherical convolution can be used for spatial smoothing. This is a spatial filtering process, or a windowing in the coefficient domain (convolution). Its purpose is to minimize the side lobes, so-called panning lobes. A new coefficient <MAT> is given by the weighted product of the original HOA coefficient <MAT> and a zonal coefficient <MAT> [<NUM>]: <MAT>.

This is equivalent to a left convolution on S<NUM> in the spatial domain [<NUM>]. Conveniently this is used in [<NUM>] to smooth the directive properties of loudspeaker signals prior to rendering / decoding by weighting the HOA coefficients B by: <MAT> with vector <MAT> containing usually real valued weighting coefficients and a constant factor df. The idea of smoothing is to attenuate HOA coefficients with increasing order index n. A well-known example of smoothing weighting coefficients <IMG> are so called max rV, max rE and inphase coefficients [<NUM>]. The first offers the default amplitude beam (trivial, <IMG> = (<NUM>,<NUM>,. , <NUM>)T, a vector of length O3D with only ones), the second provides evenly distributed angular power and inphase features full side lobe suppression.

In the following, further details and embodiments of the disclosed solution are described. First, a renderer architecture is described in terms of its initialization, start-up behavior and processing.

Every time the loudspeaker setup, i.e. the number of loudspeakers or position of any loudspeaker relative to the listening position changes, the renderer needs to perform an initialization process to determine a set of decoding matrices for any HOA-order N that supported HOA input signals have. Also the individual speaker delays dl for the delay lines and speaker gains <IMG> are determined from the distance between a speaker and a listening position. This process is described below. In one embodiment, the derived decoding matrices are stored within a code book. Every time the HOA audio input characteristics change, a renderer control unit determines currently valid characteristics and selects a matching decode matrix from the code book. Code book key can be the HOA order N or, equivalently, O<NUM>D (see eq.

The schematic steps of data processing for rendering are explained with reference to <FIG>, which shows a block diagram of processing blocks of the renderer. These are a first buffer <NUM>, a Frequency Domain Filtering unit <NUM>, a rendering processing unit <NUM>, a second buffer <NUM>, a delay unit <NUM> for L channels, and a digital-to-analog converter and amplifier <NUM>.

The HOA time samples with time-index t and O3D HOA coefficient channels b(t) are first stored in the first buffer <NUM> to form blocks of M samples with block index µ. The coefficients of B(µ) are frequency filtered in the Frequency Domain Filtering unit <NUM> to obtain frequency filtered blocks B̂(µ). This technology is known (see [<NUM>]) for compensating for the distance of the spherical loudspeaker sources and enabling the handling of near field recordings. The frequency filtered block signals B̂(µ) are rendered to the spatial domain in the rendering processing unit <NUM> by: <MAT> with <MAT> representing a spatial signal in L channels with blocks of M time samples. The signal is buffered in the second buffer <NUM> and serialized to form single time samples with time index t in L channels, referred to as w(t) in <FIG>. This is a serial signal that is fed to L digital delay lines in the delay unit <NUM>. The delay lines compensate for different distances of listening position to individual speaker l with a delay of dl samples. In principle, each delay line is a FIFO (first-in-first-out memory). Then, the delay compensated signals <NUM> are D/A converted and amplified in the digital-to-analog converter and amplifier <NUM>, which provides signals <NUM> that can be fed to L loudspeakers. The speaker gain compensation <IMG> can be considered before D/A conversion or by adapting the speaker channel amplification in analog domain.

The renderer initialization works as follows.

First, speaker number and positions need to be known. The first step of the initialization is to make available the new speaker number L and related positions <IMG> = [r<NUM>, r<NUM>,. , rL], with <MAT> , where rl is the distance from a listening position to a speaker l , and where <MAT> are the related spherical angles. Various methods may apply, e.g. manual input of the speaker positions or automatic initialization using a test signal. Manual input of the speaker positions <IMG> may be done using an adequate interface, like a connected mobile device or an device-integrated user-interface for selection of predefined position sets. Automatic initialization may be done using a microphone array and dedicated speaker test signals with an evaluation unit to derive <IMG>. The maximum distance rmax is determined by rmax = max(r<NUM>,. ,rL), the minimal distance rmin by rmin = min(r<NUM> ,.

The L distances rl and rmax are input to the delay line and gain compensation <NUM>. The number of delay samples for each speaker channel dl are determined by <MAT> with sampling rate fs, speed of sound c (c ≅ <NUM>m/s at a temperature of <NUM>°celsius) and <MAT> indicating rounding to next integer. To compensate the speaker gains for different rl , loudspeaker gains <IMG> are determined by <MAT>, or are derived using an acoustical measurement.

Calculation of decoding matrices, e.g. for the code book, works as follows. Schematic steps of a method for generating the decode matrix, in one embodiment, are shown in <FIG> shows, in one embodiment, processing blocks of a corresponding device for generating the decode matrix. Inputs are speaker directions <IMG>, a spherical modeling grid <IMG> and the HOA-order N.

The speaker directions <IMG> = [Ω̂<NUM>,. , Ω̂L] can be expressed as spherical angles <MAT>, and the spherical modeling grid <IMG> = [Ω<NUM>,. , ΩS] by spherical angles Ωs = [θs, ϕs]T. The number of directions is selected larger than the number of speakers (S > L) and larger than the number of HOA coefficients (S > O3D). The directions of the grid should sample the unit sphere in a very regular manner. Suited grids are discussed in [<NUM>], [<NUM>] and can be found in [<NUM>], [<NUM>]. The grid <IMG> is selected once. As an example, a S = <NUM> grid from [<NUM>] is sufficient for decoding matrices up to HOA-order N = <NUM>. Other grids may be used for different HOA orders. The HOA-order N is selected incremental to fill the code book from N = <NUM>,. , Nmax, with Nmax as the maximum HOA-order of supported HOA input content.

The speaker directions <IMG> and the spherical modeling grid <IMG> are input to a Build Mix-Matrix block <NUM>, which generates a mix matrix G thereof. The a spherical modeling grid <IMG> and the HOA order N are input to a Build Mode-Matrix block <NUM>, which generates a mode matrix Ψ̃ thereof. The mix matrix G and the mode matrix Ψ̃ are input to a Build Decode Matrix block <NUM>, which generates a decode matrix D thereof. The decode matrix is input to a Smooth Decode Matrix block <NUM>, which smoothes and scales the decode matrix. Further details are provided below. Output of the Smooth Decode Matrix block <NUM> is the decode matrix D, which is stored in the code book with related key N (or alternatively O3D). In the Build Mode-Matrix block <NUM>, the spherical modeling grid <IMG> is used to build a mode matrix analogous to eq. (<NUM>): Ψ̃ = [y<NUM>,. ys] with <MAT>, <MAT>. It is noted that the mode matrix Ψ̃ is referred to as Ξ in [<NUM>].

In the Build Mix-Matrix block <NUM>, a mix matrix G is created with <MAT>. It is noted that the mix matrix G is referred to as W in [<NUM>]. An lth row of the mix matrix G consists of mixing gains to mix S virtual sources from directions <IMG> to speaker l. In one embodiment, Vector Base Amplitude Panning (VBAP) [<NUM>] is used to derive these mixing gains, as also in [<NUM>]. The algorithm to derive G is summarized in the following.

In the Build Decode Matrix block <NUM>, the compact singular value decomposition of the matrix product of the mode matrix and the transposed mixing matrix is calculated. This is an important aspect of the present invention, which can be performed in various manners. In one embodiment, the compact singular value decomposition S of the matrix product of the mode matrix Ψ̃ and the transposed mixing matrix GT is calculated according to: <MAT>.

In an alternative embodiment, the compact singular value decomposition S of the matrix product of the mode matrix Ψ̃ and the pseudo-inverse mixing matrix G+ is calculated according to: <MAT> where G+ is the pseudo-inverse of mixing matrix G.

In one embodiment, a diagonal matrix where Ŝ = diag( Ŝ<NUM>,. , ŜK) is created where.

A suitable threshold value a was found to be around <NUM>. Small deviations e.g. within a range of ±<NUM> or a range of ±<NUM>% are acceptable. The decode matrix is then calculated as follows: D̂ = V Ŝ UH.

In the Smooth Decode Matrix block <NUM>, the decode matrix is smoothed. Instead of applying smoothing coefficients to the HOA coefficients before decoding, as known in prior art, it can be combined directly with the decode matrix. This saves one processing step, or processing block respectively.

In order to obtain good energy preserving properties also for decoders for HOA content with more coefficients than loudspeakers (i.e. O3D > L), the applied smoothing coefficients <IMG> are selected depending on the HOA order N (O3D = (N + <NUM>)<NUM>):
For L ≥ O3D, <IMG> corresponds to max rE coefficients derived from the zeros of the Legendre polynomials of order N + <NUM> , as in [<NUM>].

For L < O<NUM>D, the coefficients of <IMG> constructed from a Kaiser window as follows: <MAT> with len = 2N + <NUM>, width = <NUM>N, where <IMG> is a vector with <NUM>N + <NUM> real valued elements. The elements are created by the Kaiser window formula <MAT> where I<NUM>() denotes the zero-order Modified Bessel function of first kind. The vector <IMG> is constructed from the elements of : <MAT> where every element <IMG> gets <NUM>n + <NUM> repetitions for HOA order index n = <NUM>. N, and cf is a constant scaling factor for keeping equal loudness between different HOA-order programs. That is, the used elements of the Kaiser window begin with the (N+<NUM>)st element, which is used only once, and continue with subsequent elements which are used repeatedly: the (N+<NUM>)nd element is used three times, etc..

In one embodiment, the smoothed decode matrix is scaled. In one embodiment, the scaling is performed in the Smooth Decode Matrix block <NUM>, as shown in <FIG>. In a different embodiment, the scaling is performed as a separate step in a Scale Matrix block <NUM>, as shown in <FIG>.

In one embodiment, the constant scaling factor is obtained from the decoding matrix. In particular, it can be obtained according to the so-called Frobenius norm of the decoding matrix: <MAT> where d̃l,q is a matrix element in line l and column q of the matrix D̃ (after smoothing). The normalized matrix is D = cf D̃.

<FIG> shows, according to one aspect of the invention, a device for decoding an audio sound field representation for audio playback. It comprises a rendering processing unit <NUM> having a decode matrix calculating unit <NUM> for obtaining the decode matrix D, the decode matrix calculating unit <NUM> comprising means 1x for obtaining a number L of target speakers and means for obtaining positions <IMG> of the speakers, means 1y for determining positions a spherical modeling grid <IMG> and means 1z for obtaining a HOA order N, and first processing unit <NUM> for generating a mix matrix G from the positions of the spherical modeling grid <IMG> and the positions of the speakers, second processing unit <NUM> for generating a mode matrix Ψ̃ from the spherical modeling grid <IMG> and the HOA order N, third processing unit <NUM> for performing a compact singular value decomposition of the product of the mode matrix Ψ̃ with the Hermitian transposed mix matrix G according to U S VH = Ψ̃GH , where U,V are derived from Unitary matrices and S is a diagonal matrix with singular value elements, calculating means <NUM> for calculating a first decode matrix D̂ from the matrices U,V according to D̂ = V UH, and a smoothing and scaling unit <NUM> for smoothing and scaling the first decode matrix D̂ with smoothing coefficients <IMG>, wherein the decode matrix D is obtained. In one embodiment, the smoothing and scaling unit <NUM> as a smoothing unit <NUM> for smoothing the first decode matrix D̂, wherein a smoothed decode matrix D̃ is obtained, and a scaling unit <NUM> for scaling smoothed decode matrix D̃, wherein the decode matrix D is obtained.

<FIG> shows speaker positions in an exemplary <NUM>-speaker setup in a node schematic, where speakers are shown as connected nodes. Foreground connections are shown as solid lines, background connections as dashed lines. <FIG> shows the same speaker setup with <NUM> speakers in a foreshortening view.

In the following, obtained example results with the speaker setup as in <FIG> and <FIG> are described. The energy distribution of the sound signal, and in particular the ratio Ê / E is shown in dB on the <NUM> sphere (all test directions). As an example for a loud speaker panning beam, the center speaker beam (speaker <NUM> in <FIG>) is shown. For example, a decoder matrix that is designed as in [<NUM>], with N=<NUM>, produces a ratio Ê / E as shown in <FIG>. It provides almost perfect energy preserving characteristics, since the ratio Ê / E is almost constant: differences between dark areas (corresponding to lower volumes) and light areas (corresponding to higher volumes) are less than <NUM>. However, as shown in <FIG>, the corresponding panning beam of the center speaker has strong side lobes. This disturbs spatial perception, especially for off-center listeners.

On the other hand, a decoder matrix that is designed as in [<NUM>], with N=<NUM>, produces a ratio Ê / E as shown in <FIG>. In the scale used in <FIG>, dark areas correspond to lower volumes down to -2dB and light areas to higher volumes up to +2dB. Thus, the ratio Ê / E shows fluctuations larger than 4dB, which is disadvantageous because spatial pans e.g. from top to center speaker position with constant amplitude cannot be perceived with equal loudness. However, as shown in <FIG>, the corresponding panning beam of the center speaker has very small side lobes, which is beneficial for off-center listening positions.

<FIG> shows the energy distribution of a sound signal that is obtained with a decoder matrix according to the present invention, exemplarily for N=<NUM> for easy comparison. The scale (shown on the right-hand side of <FIG>) of the ratio Ê / E ranges from <NUM> - <NUM>. Thus, fluctuations in the ratio are smaller than <NUM>. 31dB, and the energy distribution in the sound field is very even. Consequently, any spatial pans with constant amplitude are perceived with equal loudness. The panning beam of the center speaker has very small side lobes, as shown in <FIG>. This is beneficial for off center listening positions, where side lobes may be audible and thus would be disturbing. Thus, the present invention provides combined advantages achievable with the prior art in [<NUM>] and [<NUM>], without suffering from their respective disadvantages.

It is noted that whenever a speaker is mentioned herein, a sound emitting device such as a loudspeaker is meant.

The flowchart and/or block diagrams in the figures illustrate the configuration, operation and functionality of possible implementations of systems, methods and computer program products according to various embodiments of the present invention. In this regard, each block in the flowchart or block diagrams may represent a module, segment, or portion of code, which comprises one or more executable instructions for implementing the specified logical functions.

It will also be noted that each block of the block diagrams and/or flowchart illustration, and combinations of the blocks in the block diagrams and/or flowchart illustration, can be implemented by special purpose hardware-based systems that perform the specified functions or acts, or combinations of special purpose hardware and computer instructions.

Further, as will be appreciated by one skilled in the art, aspects of the present principles can be embodied as a system, method or computer readable medium. Accordingly, aspects of the present principles can take the form of an entirely hardware embodiment, an entirely software embodiment (including firmware, resident software, micro-code, and so forth), or an embodiment combining software and hardware aspects that can all generally be referred to herein as a "circuit," "module", or "system. " Furthermore, aspects of the present principles can take the form of a computer readable storage medium. Any combination of one or more computer readable storage medium(s) may be utilized. A computer readable storage medium as used herein is considered a non-transitory storage medium given the inherent capability to store the information therein as well as the inherent capability to provide retrieval of the information therefrom.

Claim 1:
A computer-implemented method for generating a decode matrix (D) used for rendering a Higher-Order Ambisonics sound field representation for audio playback, comprising:
- obtaining (<NUM>) a number (L) of target speakers and positions (<IMG>) of the speakers;
- determining (<NUM>) positions of a spherical modelling grid (<IMG>) and obtaining a HOA order (N);
- generating (<NUM>) a mix matrix (G) from the positions of the spherical modelling grid (<IMG>) and the positions of the speakers (<IMG>);
- generating (<NUM>) a mode matrix (Ψ̃) from the spherical modeling grid (<IMG>) and the HOA order (N);
- performing (<NUM>) a compact singular value decomposition of the product of the mode matrix (Ψ̃) with the Hermitian transposed mix matrix (G) according to VSUH = GΨ̃H , where U,V are derived from Unitary matrices and S is a diagonal matrix with singular value elements, and calculating a first decode matrix (D̂) from the matrices U,V according to D̂ = V Ŝ UH, wherein Ŝ is a truncated compact singular value decomposition matrix that is either an identity matrix or a modified diagonal matrix, the modified diagonal matrix being derived from said diagonal matrix with singular value elements by replacing singular value elements larger or equal than a threshold by ones, and replacing singular value elements that are smaller than the threshold by zeros, wherein the threshold depends on the actual values of the diagonal matrix with singular value elements; and
- smoothing and scaling (<NUM>,<NUM>) the first decode matrix (D̂) with smoothing coefficients that are determined based on elements of a Kaiser window determined based on <IMG> = KaiserWindow(len, width), wherein len = 2N + <NUM>, width = 2N, wherein <IMG> is a vector with 2N + <NUM> real valued elements based on <MAT> wherein I<NUM>( ) denotes a zero-order Modified Bessel function of a first kind and i = <NUM>, ..., 2N, wherein the decode matrix (D) is obtained.