Audio encoding device and method

Abstract

A method and a device encode N audio signals, from N microphones where N≥3. For each pair of the N audio signals an angle of incidence of direct sound is estimated. A-format direct sound signals are derived from the estimated angles of incidence by deriving from each estimated angle an A-format direct sound signal. Each A-format direct sound signal is a first-order virtual microphone signal, for example, a cardioids signal.

Description

FIELD

The present disclosure is related to audio recording and encoding, in particular for virtual reality applications, especially for virtual reality provided by a small portable device.

BACKGROUND

Virtual reality (VR) sound recording typically requires Ambisonic B-format with expensive directive microphones. Professional audio microphones exist to either record A-format to be encoded into Ambisonic B-format or directly Ambisonic B-format, for instance using Soundfield microphones. More generally speaking, it is technically difficult to arrange omnidirectional microphones on a mobile device to capture sound for VR.

A way to generate Ambisonic B-format signals, given a distribution of omnidirectional microphones, is based on differential microphone arrays, i.e. applying delay and adding beam-forming in order to derive first order virtual microphone (e.g. cardioids) signals as A-format.

The first limitation of this technique results from its spatial aliasing which, by design, reduces the bandwidth to frequencies f in the range:

$\begin{array}{cc}f<\frac{c}{4d_{\mathrm{mic}}},& \left(1\right)\end{array}$ where c stands for the sound celerity and d_mic the distance between a pair of two omnidirectional microphones. A second weakness results, for higher order Ambisonic B-format, from the microphone requirement. The required number of microphones and their required positions are not anymore suitable for mobile devices.

Another way of generating ambisonic B-format signals from omnidirectional microphones corresponds to sampling the sound field at the recording point in space using a sufficiently dense distribution of microphones. These sampled sound pressure signals are then converted to spherical harmonics, and can be linearly combined to eventually generate B-format signals.

The main limitation of such approaches is the required number of microphones. For consumer applications, with only few microphones (commonly up to 6), linear processing is too limited, leading to signal to noise ratio (SNR) issues at low frequencies, and aliasing at high frequencies.

Directional Audio Coding (DirAc) is a further method for spatial sound representation, but it does not generate B-format signals. Instead, it reads first order B-format signals and generates a number of related audio parameters (direction of arrival, diffuseness) and adds these to an omnidirectional audio channel. Later, the decoder takes the above information and converts it to a multi-channel audio signal using amplitude panning for direct sound and de-correlating for diffuse sound.

DirAc is thus a different technique, which takes B-format as input to render it to its own audio format.

SUMMARY

Therefore, the present inventors have recognized a need to provide an audio encoding device and method, which allow for generating ambisonic B-format sound signals, while requiring only a low number of microphones, and achieving a high output sound quality.

Embodiments of the present disclosure provide such audio encoding devices and methods that allow for generating ambisonic B-format sound signals, while requiring only a low number of microphones, and achieve a high output sound quality.

According to a first aspect of the present disclosure, an audio encoding device, for encoding N audio signals, from N microphones, where N≥3, is provided. The device comprises a delay estimator, configured to estimate angles of incidence of direct sound by estimating for each pair of the N audio signals an angle of incidence of direct sound, and a beam deriver, configured to derive A-format direct sound signals from the estimated angles of incidence by deriving from each estimated angle of incidence an A-format direct sound signal, each A-format direct sound signal being a first-order virtual microphone signal, especially a cardioids signal. This allows for determining the A-format direct sound signals with a low hardware effort.

According to an implementation form of the first aspect, the device additionally comprises an encoder, configured to encode the A-format direct sound signals in first-order ambisonic B-format direct sound signals by applying a transformation matrix to the A-format direct sound signals. This allows for generating ambisonic B-format signals using only a very low number of microphones, but still achieving a high output sound quality.

According to an implementation form of the first aspect, N=3. The audio encoding device moreover comprises a short time Fourier transformer, configured to perform a short time Fourier transformation on each of the N audio signals x₁, x₂, x₃, resulting in N short time Fourier transformed audio signals X₁[k,i], X₂[k,i], X₃[k,i]. The delay estimator is then configured to determine cross spectra of each pair of short time Fourier transformed audio signals according to:X₁₂[k,i]=α_XX₁[k,i]X*₂[k,i]+(1−α_X)X₁₂[k−1,i],X₁₃[k,i]=α_XX₁[k,i]X*₃[k,i]+(1−α_X)X₁₃[k−1,i],X₂₃[k,i]=α_XX₂[k,i]X*₃[k,i]+(1−α_X)X₂₃[k−1,i],determine an angle of the complex cross spectrum of each pair of short time Fourier transformed audio signals according to:

${\tilde{\psi }}_{12}\left[k,i\right]=\arctan j\frac{X_{12}\left[k,i\right]X_{12}^{*}\left[k,i\right]}{X_{12}\left[k,i\right]+X_{12}^{*}\left[k,i\right]},{\tilde{\psi }}_{13}\left[k,i\right]=\arctan j\frac{X_{13}\left[k,i\right]X_{13}^{*}\left[k,i\right]}{X_{13}\left[k,i\right]+X_{13}^{*}\left[k,i\right]},{\tilde{\psi }}_{23}\left[k,i\right]=\arctan j\frac{X_{23}\left[k,i\right]X_{23}^{*}\left[k,i\right]}{X_{23}\left[k,i\right]+X_{23}^{*}\left[k,i\right]},$ perform a phase unwrapping to {tilde over (ψ)}₁₂, {tilde over (ψ)}₁₃, {tilde over (ψ)}₂₃, resulting in Ψ₁₂, Ψ₁₃, Ψ₂₃ estimate the delay in number of samples according to:δ₁₂[k,i]=(N_STFT/2+1)/(iπ)ψ₁₂[k,i],δ₁₃[k,i]=(N_STFT/2+1)/(iπ)ψ₁₃[k,i],δ₂₃[k,i]=(N_STFT/2+1)/(iπ)ψ₂₃[k,i], if i≤i_alias orδ₁₂[k,i]=(N_STFT/2+1)/(iπ)Ψ₁₂[k,i],δ₁₃[k,i]=(N_STFT/2+1)/(iπ)Ψ₁₃[k,i],δ₂₃[k,i]=(N_STFT/2+1)/(iπ)Ψ₂₃[k,i], if i>i_alias estimate the delay in seconds according to:

${\tau }_{12}\left[k,i\right]=\frac{{\delta }_{12}\left[k,i\right]}{f_s}$ ${\tau }_{13}\left[k,i\right]=\frac{{\delta }_{13}\left[k,i\right]}{f_s}$ ${\tau }_{23}\left[k,i\right]=\frac{{\delta }_{23}\left[k,i\right]}{f_s}$ estimate the angles of incidence according to:

${\theta }_{12}\left[k,i\right]=\arcsin \left(\frac{c{\tau }_{12}\left[k,i\right]}{d_{\mathrm{mic}}}\right),{\theta }_{13}\left[k,i\right]=\arcsin \left(\frac{c{\tau }_{13}\left[k,i\right]}{d_{\mathrm{mic}}}\right),{\theta }_{23}\left[k,i\right]=\arcsin \left(\frac{c{\tau }_{23}\left[k,i\right]}{d_{\mathrm{mic}}}\right),$ whereinx₁ is a first audio signal of the N audio signals,x₂ is a second audio signal of the N audio signals,x₃ is a third audio signal of the N audio signals,X₁ is a first short time Fourier transformed audio signal,X₂ is a second short time Fourier transformed audio signal,X₃ is a third short time Fourier transformed audio signal,k is a frame of the short time Fourier transformed audio signal, andi is a frequency bin of the short time Fourier transformed audio signal,X₁₂ is a cross spectrum of a pair of X₁ and X₂,X₁₃ is a cross spectrum of a pair of X₁ and X₃,X₂₃ is a cross spectrum of a pair of X₂ and X₃,α_x is a forgetting factor,X* is the conjugate complex of X,j is the imaginary unit,{tilde over (ψ)}₁₂ is an angle of the complex cross spectrum of X₁₂,{tilde over (ψ)}₁₃ is an angle of the complex cross spectrum of X₁₃,{tilde over (ψ)}₂₃ is an angle of the complex cross spectrum of X₂₃,i_alias is a frequency bin corresponding to an aliasing frequency,f_s is a sampling frequency,d_mic is a distance of the microphones, andc is the speed of sound. This allows for a simple and efficient determining of the delays.

According to a further implementation form of the first aspect, the beam deriver is configured to determine cardioid directional responses according to:

$D_{12}\left[k,i\right]=\frac{1}{2}\left(1+\cos \left({\theta }_{12}\left[k,i\right]-\frac{\pi }{2}\right)\right),D_{13}\left[k,i\right]=\frac{1}{2}\left(1+\cos \left({\theta }_{13}\left[k,i\right]-\frac{\pi }{2}\right)\right),D_{23}\left[k,i\right]=\frac{1}{2}\left(1+\cos \left({\theta }_{23}\left[k,i\right]-\frac{\pi }{2}\right)\right),$ and derive the A-format direct sound signals according to:A₁₂[k,i]=D₁₂[k,i]X₁[k,i],A₁₃[k,i]=D₁₃[k,i]X₁[k,i],A₂₃[k,i]=D₂₃[k,i]X₁[k,i],whereinD is a cardioid directional response, andA is an A-format direct sound signal. This allows for a simple and efficient determining of the beam signals.

According to a further implementation form of the first aspect, the encoder is configured to encode the A-format direct sound signals to the first-order ambisonic B-format direct sound signals according to:

$\left[\begin{array}{c}{R}_W\\ R_X\\ R_Y\end{array}\right]={\Gamma }^{-1}\left[\begin{array}{c}{A}_{12}\\ A_{13}\\ A_{23}\end{array}\right],$ whereinR_W is a first, zero-order ambisonic B-format direct sound signal,R_x is a first, first-order ambisonic B-format direct sound signal,R_y is a second, first-order ambisonic B-format direct sound signal, andΓ⁻¹ is the transformation matrix. This allows for a simple and efficient determining of the beam signals.

According to a further implementation form of the first aspect, the device comprises a direction of arrival estimator, configured to estimate a direction of arrival from the first-order ambisonic B-format direct sound signals, and a higher order ambisonic encoder, configured to encode higher order ambisonic B-format direct sound signals, using the first-order ambisonic B-format direct sound signals and the estimated direction of arrival, wherein higher order ambisonic B-format direct sound signals have an order higher than one. Thereby, an efficient encoding of the ambisonic B-format direct sound signal is achieved.

According to a further implementation form of the first aspect, the direction of arrival estimator is configured to estimate the direction of arrival according to:

${\theta }_{\mathrm{XY}}\left[k,i\right]=\arctan \frac{R_Y\left[k,i\right]}{R_X\left[k,i\right]},$ whereinθ_XY [k,i] is a direction of arrival of a direct sound of frame k and frequency bin i. This allows for a simple and efficient determining of the directions of arrival.

According to a further implementation form of the first aspect, the higher order ambisonic B-format direct sound signals comprise second order ambisonic B-format direct sound signals limited to two dimensions, wherein the higher order ambisonic encoder is configured to encode the second order ambisonic B-format direct sound signals according to:

$R_R\stackrel{\Delta }{=}\left(3{\sin }^2\varphi -1\right)\text{/}2=-1\text{/}2,R_S\stackrel{\Delta }{=}\sqrt{3}\text{/}2\cos \theta \sin 2\varphi =0,R_T\stackrel{\Delta }{=}\sqrt{3}\text{/}2\sin \theta \sin 2\varphi =0,R_U\stackrel{\Delta }{=}\sqrt{3}\text{/}2\cos 2\theta {\cos }^2\varphi =\sqrt{3}\text{/}2\cos 2{\theta }_{\mathrm{XY}},R_V\stackrel{\Delta }{=}\sqrt{3}\text{/}2\sin 2\theta {\cos }^2\varphi =\sqrt{3}\text{/}2\sin 2{\theta }_{\mathrm{XY}},$ whereinR_R is a first, second-order ambisonic B-format direct sound signal,R_S is a second, second-order ambisonic B-format direct sound signal,R_T is a third, second-order ambisonic B-format direct sound signal,R_U is a fourth, second-order ambisonic B-format direct sound signal,R_V is a fifth, second-order ambisonic B-format direct sound signal,Δ denotes “defined as”,ϕ is an elevation angle, andθ is an azimuth angle. This allows for an efficient encoding of the higher order ambisonic B-format signals.

According to a further implementation form of the first aspect, the audio encoding device comprises a microphone matcher, configured to perform a matching of the N frequency domain audio signals, resulting in N matched frequency domain audio signals. This allows for further quality increase of the output signals.

According to a further implementation form of the first aspect, the audio encoding device comprises a diffuse sound estimator, configured to estimate a diffuse sound power, and a de-correlation filter bank, configured to perform a de-correlation of the diffuse sound power by generating three orthogonal diffuse sound components from the diffuse sound estimate power. This allows for implementing diffuse sound into the output signals.

According to a further implementation form of the first aspect, the diffuse sound estimator is configured to estimate the diffuse sound power according to:

$A=1-{\Phi }_{\mathrm{diff}}^2,V=2{\Phi }_{\mathrm{diff}}E\left\{X_1{X}_2^{*}\right\}-E\left\{X_1{X}_1^{*}\right\}-E\left\{X_2{X}_2^{*}\right\},C=E\left\{X_1{X}_1^{*}\right\}E\left\{X_2{X}_2^{*}\right\}-E{\left\{X_1{X}_2^{*}\right\}}^2,P_{\mathrm{diff}}\left[k,i\right]=\frac{-B-\sqrt{B^2-4\mathrm{AC}}}{2A},$ whereinP_diff is the diffuse sound power,E{ } is an expectation value,Φ_diff² is a normalized cross-correlation coefficient between N₁ and N₂,N₁ is diffuse sound in a first channel, andN₂ is diffuse sound in a second channel. This allows for an especially efficient estimation of the diffuse sound power.

According to a further implementation form of the first aspect, the de-correlation filter bank is configured to perform the de-correlation of the diffuse sound power by generating three orthogonal diffuse sound components from the diffuse sound estimate power:{tilde over (D)}_W[k,i]=DFR_Ww_uU₁P_2D-diff[k,i],{tilde over (D)}_X[k,i]=DFR_Xw_uU₂P_2D-diff[k,i],{tilde over (D)}_Y[k,i]=DFR_Yw_uU₃P_2D-diff[k,i],wherein

${\mathrm{DFR}}_a\stackrel{\Delta }{=}\frac{1}{4\pi }\underset{-\frac{\pi }{2}}{\overset{\frac{\pi }{2}}{\int }}\underset{-\pi }{\overset{\pi }{\int }}{R_a\left(\theta ,\varphi \right)}^2\cos \varphi d\theta d\varphi ,R_X\left(\theta ,\varphi \right)=\cos \varphi \cos \theta$ $R_Y\left(\theta ,\varphi \right)=\cos \varphi \sin \theta$ $R_W\left(\theta ,\varphi \right)=1$ $w_u\left[n\right]=\exp \left(-\frac{0.5\ln 1e6n}{f_s{\mathrm{RT}}_{60}}\right)\mathrm{with}-l_u<n<l_u$ wherein {tilde over (D)}_W[k,i] is a first channel diffuse sound component,wherein {tilde over (D)}_X[k,i] is second channel diffuse sound component,wherein {tilde over (D)}_Y[k,i] is third channel diffuse sound component,DFR_W is a diffuse-field response of the first channel,DFR_X is a diffuse-field response of the second channel,DFR_Y is a diffuse-field response of the third channel,w_u is an exponential window,RT₆₀ is a reverberation time,U₁,U₂,U₃ is the de-correlation filter bank,u is Gaussian noise sequence,l_u is a given length of the Gaussian noise sequence, andP_2D-diff is the diffuse noise power. Thereby, an efficient de-correlation of the diffuse sound power is calculated.

According to a further implementation form of the first aspect, the audio encoding device comprises an adder, configured to add channel-wise, the first-order ambisonic B-format direct sound signals and the higher order ambisonic B-format direct sound signals, and/or the diffuse sound signals, resulting in complete ambisonic B-format signals. Thereby, in a simple manner, a finished output signal is generated.

According to a second aspect of the present disclosure, an audio recording device comprising N microphones configured to record the N audio signals and an audio encoding device according to the first aspect or any of the implementation forms of the first aspect is provided. This allows for an audio recording and encoding in a single device.

According to a third aspect of the present disclosure, a method for encoding N audio signals, from N microphones, where N≥3 is provided. The method comprises estimating angles of incidence of direct sound by estimating for each pair of the N audio signals an angle of incidence of direct sound, and deriving A-format direct sound signals from the estimated angles of incidence by deriving from each estimated angle of incidence an A-format direct sound signal, each A-format direct sound signal being a first-order virtual microphone signal. This allows for determining the A-format direct sound signals with a low hardware effort.

According to an implementation form of the third aspect, the method additionally comprises encoding the ambisonic A-format direct sound signals in first-order ambisonic B-format direct sound signals by applying at least one transformation matrix to the A-format direct sound signals. This allows for a simple and efficient determining of the ambisonic B-format direct sound signals.

The method may further comprise extracting higher order ambisonic B-format direct sound signals by extracting direction of arrival from first order ambisonic B-format direct sound signals.

According to a fourth aspect of the present disclosure, a computer program with a program code for performing the method according to the third aspect is provided.

A method is provided for parametric encoding of multiple omnidirectional microphone signals into any order Ambisonic B-format by means of:

• • robust estimation of the angle of incidence of sound, based on microphone pair beam signals
  • and de-correlation of diffuse sound

The disclosed approach is based on at least three omnidirectional microphones on a mobile device. Successively, it estimates the angles of incidence of direct sound by means of delay estimation between the different microphone pairs. Given the incidences of direct sound, it derives beam signals, called the direct sound A-format signals. The direct sound A-format signals are then encoded into first order B-format using relevant transformation matrix.

For optional higher order B-format, a direction of arrival estimate is derived from the X and Y first order B-format signals. The diffuse, non-directive sound is optionally rendered as multiple orthogonal components, generated using de-correlation filters.

Generally, it has to be noted that all arrangements, devices, elements, units and means and so forth described in the present application could be implemented by software or hardware elements or any kind of combination thereof. Furthermore, the devices may be processors or may comprise processors, wherein the functions of the elements, units and means described in the present applications may be implemented in one or more processors. All steps which are performed by the various entities described in the present application as well as the functionality described to be performed by the various entities are intended to mean that the respective entity is adapted to or configured to perform the respective steps and functionalities. Even if in the following description or exemplary embodiments, a specific functionality or step to be performed by a general entity is not reflected in the description of a specific detailed element of that entity which performs that specific step or functionality, it should be clear for a skilled person that these methods and functionalities can be implemented in respect of software or hardware elements, or any kind of combination thereof.

BRIEF DESCRIPTION OF DRAWINGS

The present disclosure is in the following explained in detail in relation to embodiments of the present disclosure in reference to the enclosed drawings, in which:

FIG. 1 shows a first embodiment of the audio encoding device according to the first aspect of the present disclosure and the audio recording device according to the second aspect of the present disclosure;

FIG. 2 shows a second embodiment of the audio encoding device according to the first aspect of the present disclosure and the audio recording device according to the second aspect of the present disclosure;

FIG. 3 shows a pair of microphones in a diagram depicting the determining of an angle of incidence of a sound event;

FIG. 4 shows a third embodiment of the audio recording device according to the second aspect of the present disclosure;

FIG. 5 shows A-format direct sound signals in a two-dimensional diagram;

FIG. 6 shows B-format direct sound signals in a two-dimensional diagram;

FIG. 7 shows diffuse sound received by two microphones;

FIG. 8 shows direct sound and diffuse sound in a two-dimensional diagram;

FIG. 9 shows an example of a de-correlation filter, as used by an audio encoding device according to a fourth embodiment of the first aspect; and

FIG. 10 shows an embodiment of the third aspect of the present disclosure in a flow diagram.

DETAILED DESCRIPTION

First, we demonstrate the construction and general function of an embodiment of the first aspect and second aspect of the present disclosure along FIG. 1. With regard to FIG. 2-FIG. 9, further details of the construction and function of the first embodiment and the second embodiment are shown. With regard to FIG. 10, finally the function of an embodiment of the third aspect of the present disclosure is described in detail.

In FIG. 1, a first embodiment of the audio encoding device 3 is shown. Moreover, a first embodiment of the audio recording device 1 according to the second aspect of the present disclosure is shown.

The audio recording device 1 comprises a number of N≥3 microphones 2, which are connected to the audio encoding device 3. The audio encoding device 3 comprises a delay estimator 11, which is connected to the microphones 2. The audio encoding device 3 moreover comprises a beam deriver 12, which is connected to the delay estimator. Furthermore, the audio encoding device 3 comprises an encoder 13, which is connected to the beam deriver 12. Note that the encoder 13 is an optional feature with regard to the first aspect of the present disclosure.

In order to determine ambisonic B-format direct sound signals, the microphones 2 record N≥3 audio signals. These audio signals are preprocessed by components integrated into the microphones 2, in this diagram. For example, a transformation into the frequency domain is performed. This will be shown in more detail along FIG. 2. The preprocessed audio signals are handed to the delay estimator 11, which estimates angles of incidence of direct sound by estimating for each pair of the N audio signals and angle of incidence of direct sound. These angles of incidence of direct sound are handed to the beam deriver 12, which derives A-format direct sound signals therefrom. Each A-format direct sound signal is a first-order virtual microphone signal, especially a cardioid signal. These signals are handed on to the encoder 13, which encodes the A-format direct sound signals to first-order ambisonic B-format direct sound signals by applying a transformation matrix to the A-format direct sound signals. The encoder outputs the first-order ambisonic B-format direct sound signals.

In FIG. 2, a second embodiment of the audio encoding device 3 and the audio recording device 1 are shown. Here, the individual microphones 2a, 2b, 2c, which correspond to the microphones 2 of FIG. 1, are shown. Each of the microphones 2a, 2b, 2c is connected to a short-time Fourier transformer 10a, 10b, 10c, which each performs a short-time Fourier transformation of the N audio signals resulting in N short-time Fourier transformed audio signals. These are handed on to the delay estimator 11, which performs the delay estimation and hands the angles of incidence to the beam deriver 12. The beam deriver 12 determines the A-format direct sound signals and hands them to the encoder 13, which performs the encoding to B-format direct sound signals. In FIG. 2, further components of the audio encoding device 3 are shown. Here, the audio encoding device 3 moreover comprises a direction-of-arrival estimator 20, which is connected to the encoder 13. Moreover, it comprises a higher order ambisonic encoder 21, which is connected to the direction-of-arrival estimator 20.

The direction-of-arrival estimator 20 estimates a direction of arrival from the first-order ambisonic B-format direct sound signals and hands it to the higher order ambisonic encoder 21. The higher order ambisonic encoder 21 encodes higher order ambisonic B-format direct sound signals, using the first-order ambisonic B-format direct sound signals and the estimated direction of arrival as an input. The higher order ambisonic B-format direct sound signals have a higher order than 1.

Moreover, the audio encoding device 3 comprises a microphone matcher 30, which performs a matching of the N frequency domain audio signals output by the short-time Fourier transformers 10a, 10b, 10c resulting in N match frequency domain audio signals. Connected to the microphone matcher 30, the audio encoding device 3 moreover comprises a diffuse sound estimator 31, which is configured to estimate a diffuse sound power based upon the N match frequency domain audio signals. Furthermore, the audio encoding device 3 comprises a de-correlation filter bank 32, which is connected to the diffuse sound estimator 31 and configured to perform a de-correlation of the diffuse sound power by generating three orthogonal diffuse sound components from the diffuse sound estimate power.

Finally, the audio encoding device 3 comprises an adder 40, which adds the first-order B-format direct sound signals provided by the encoder 13, the higher order ambisonic B-format signals provided by the higher order encoder 21 and the diffuse sound components provided by the de-correlation filter bank 32. The sum signal is handed to an inverse short-time Fourier transformer 41, which performs an inverse short-time Fourier transformation to achieve the final ambisonic B-format signals in the time domain.

In the following, along FIG. 3-9, further details regarding the function of the individual components, shown in FIG. 2 are described.

In FIG. 3, an angle of incidence, as it is determined by the delay estimator 11 is shown.

Especially, the propagation of direct sound following a ray from a sound source to a pair of microphones in the free-field is considered in FIG. 3.

In FIG. 4, an example of an audio recording device 1 is shown in a two-dimensional diagram. The three microphones 2a, 2b, 2c are depicted in their actual physical location.

The following algorithm aims at estimating the angle of incidence of direct sound based on cross-correlation between both recorded microphone signals x₁ and x₂, and derives parametrically gain filters to generate beams focusing in specific directions.

A phase estimation, between both recording microphones, is carried out at each time-frequency tile. The microphone time-frequency representations, X₁ and X₂, of the microphone signals, are obtained using a N_STFT points short-time Fourier transform (STFT). The delay relation between the two microphones can be derived from the cross-spectrum:X₁₂[k,i]=α_XX₁[k,i]X*₂[k,i]+(1−α_X)X₁₂[k−1,i],  (2)where * denotes the complex conjugate operator. And a_x is determined by:

$\begin{array}{cc}{\alpha }_X=\frac{N_{\mathrm{STFT}}}{T_X{f}_s},& \left(3\right)\end{array}$ where T_X is an time-constant in seconds and f_s is the sampling frequency. The phase response is defined as the angle of the complex cross-spectrum X₁₂, derived as the ratio between the imaginary and the real part of it:

$\begin{array}{cc}{\tilde{\psi }}_{12}\left[k,i\right]=\arctan j\frac{X_{12}\left[k,i\right]X_{12}^{*}\left[k,i\right]}{X_{12}\left[k,i\right]+X_{12}^{*}\left[k,i\right]},& \left(4\right)\end{array}$ where j is the imaginary unit, that satisfies j²=−1.

Unfortunately, analogous to the Nyquist frequency in temporal sampling, a microphone array has a restriction on the minimum spatial sampling rate. Using two microphones, the smallest wavelength of interest is given by:λ_alias=2d_mic  (5)corresponding to a maximum frequency,

$\begin{array}{cc}{f}_{\mathrm{alias}}=\frac{c}{{\lambda }_{\mathrm{alias}}},& \left(6\right)\end{array}$ up to which the phase estimation is unambiguous. Above this frequency, the measured phase is still obtained following (4) but with an uncertainty term related to an integer l modulo of 2π:{tilde over (ψ)}₁₂[k,i]=ψ₁₂[k,i]+2π·l[i].  (7)

Because the maximum travelling time between the two microphones of the array is given by d_mic/c, the bounds of integer l is defined by:

$\begin{array}{cc}l\left[i\right]\le L\left[i\right]=\frac{{\mathrm{id}}_{\mathrm{mic}}{f}_s}{c\left(\frac{N_{\mathrm{STFT}}}{2}+1\right)},& \left(8\right)\end{array}$

A high frequency extension is provided based in equation (8) to constrain an unwrapping algorithm. The unwrapping aims at correcting the phase angle {tilde over (ψ)}₁₂[k,i] by adding a multiple l[k,i] of 2π when absolute jump between the two consecutive elements, |{tilde over (ψ)}₁₂[k,i]−{tilde over (ψ)}₁₂[k,i−1]|, are greater than or equal to the jump tolerance of π. The estimated unwrapped phase ψ₁₂ is obtained by limiting the multiples l to their physical possible values. Eventually, even if the phase is aliased at high-frequency, its slope still follows the same principles as the delay estimation at low frequency. For the purpose of delay estimation, it is then sufficient to integrate the unwrapped phase ψ₁₂ over a number of frequency bins in order to derive its slope for later delay

$\begin{array}{cc}{\Psi }_{12}\left[k,i\right]=\frac{1}{2N_{\mathrm{hf}}}\sum _{j=-N_{\mathrm{hf}}}^{N_{\mathrm{hf}}}{\psi }_{12}\left[k,i+j\right],& \left(9\right)\end{array}$ where N_hf stands for the frequency bandwidth on which the phase is integrated.

For each frequency bin i, dividing by the corresponding physical frequency, the delay δ₁₂[k,i], expressed in number of samples, is obtained from the previously derived phase:δ₁₂[k,i]=(N_STFT/2+1)/(iπ)ψ₁₂[k,i] if i≤i_alias

otherwise:δ₁₂[k,i]=(N_STFT/2+1)/(iπ)Ψ₁₂[k,i],  (10)where i_alias is the frequency bin corresponding to the aliasing frequency (1). The delay in second is:

$\begin{array}{cc}{\tau }_{12}\left[k,i\right]=\frac{{\delta }_{12}\left[k,i\right]}{f_s}.& \left(11\right)\end{array}$

The derived delay relates directly to the angle of incidence of sound emitted by a sound source, as illustrated in FIG. 2. Given the travelling time delay between both microphones, the resulting angle of incidence θ₁₂[k,i] is:

$\begin{array}{cc}{\theta }_{12}\left[k,i\right]=\arcsin \left(\frac{c{\tau }_{12}\left[k,i\right]}{d_{\mathrm{mic}}}\right),& \left(12\right)\end{array}$ with d_mic the distance between both microphones and c the celerity of sound in the air.

In free-field, for direct sound, the directional response of a cardioid microphone pointing on the side of the array, is built as a function of the estimated angle of incidence:

$\begin{array}{cc}D\left[k,i\right]=\frac{1}{2}\left(1+\cos \left({\theta }_{12}\left[k,i\right]-\frac{\pi }{2}\right)\right).& \left(13\right)\end{array}$

By applying the gain D to the input spectrum X₁, a virtual cardioid signal can be retrieved from the direct sound of the input microphone signals. This corresponds to the function of the beam estimator 12.

In FIG. 5, three cardioid signals based upon three microphone pairs are depicted in a two-dimensional diagram, showing the respective gains.

In FIG. 6, the gains of B-format ambisonic direct sound signals are shown in a two-dimensional diagram.

In the following, the conversion from A-format direct sound signals to B-format direct sound signals is shown. This corresponds to the function of the encoder 13.

In the following Table are listed the Ambisonic B-format channels and their spherical representation D(θ,ϕ) up to third-order, normalized with the Schmidt semi-normalization (SN3D), where θ and ϕ are, respectively, the azimuth and elevation angles:

	Order	Channel	SN3D Definition: D(θ, ϕ) =
	0	W	1
	1	X	cos θcos ϕ
		Y	sin θcos ϕ
		Z	sin ϕ
	2	R	(3sin2 ϕ − 1)/2
		S	{square root over (3/2)} cosθsin2ϕ
		T	{square root over (3/2)} sinθsin2ϕ
		U	{square root over (3/2)} cos2θcos2 ϕ
		V	{square root over (3/2)} sin2θcos2 ϕ
	3	K	sinϕ(5sin2 ϕ − 3)/2
		L	{square root over (3/8)} cosθcosϕ(5sin2 ϕ − 1)
		M	{square root over (3/8)} sinθcosϕ(5sin2 ϕ − 1)
		N	{square root over (15/2)} cos2θsinϕcos2 ϕ
		O	{square root over (15/2)} sin2θsinϕcos2 ϕ
		P	{square root over (5/8)} cos3θcos3 ϕ
		Q	{square root over (5/8)} sin3θcos3 ϕ

These spherical harmonics form a set of orthogonal basis functions and can be used to describe any function on the surface of a sphere.

Without loss of generality, three, the minimum number of, microphones are considered and placed in the horizontal XY-plane, for instance disposed at the edges of a mobile device as illustrated in FIG. 3, having the coordinates (x_m1, y_m1), (x_m2, y_m2), and (x_m3, y_m3).

The three possible unordered microphone pairs are defined as:pair 1Δ=mic2→mic1pair 2Δ=mic3→mic2pair 3Δ=mic1→mic3

The look direction (Θ=0) being defined by the X-axis, their direction vectors are:

$\begin{array}{cc}{v}_{p_1}=\left(\begin{array}{c}{x}_{m_1}\\ y_{m_1}\end{array}\right)-\left(\begin{array}{c}{x}_{m_2}\\ y_{m_2}\end{array}\right),v_{p_2}=\left(x_{m_2},y_{m_2}\right)-\left(\begin{array}{c}{x}_{m_3}\\ y_{m_3}\end{array}\right),\mathrm{and}{v}_{p_3}=\left(\begin{array}{c}{x}_{m_3}\\ y_{m_3}\end{array}\right)-\left(x_{m_1},y_{m_1}\right).& \left(14\right)\end{array}$

The direction for each of the pair in the horizontal plane are:

$\begin{array}{cc}\forall n\in \left[1.\mathrm{.3}\right],{\theta }_{p_n}=\arctan \left(\frac{y_{v_{p_n}}}{x_{v_{p_n}}}\right).& \left(15\right)\end{array}$

And the microphone spacing:

$\begin{array}{cc}\forall n\in \left[1.\mathrm{.3}\right],{\partial }_{p_n}=\sqrt{x_{v_{p_n}}^2+y_{v_{p_n}}^2}.& \left(16\right)\end{array}$

The gain (13) resulting from the angle of incidence estimation is applied to each pair leading to cardioid directional responses:∀n∈[1 . . . 3],A_pn[k,i]=D_pn[k,i]X₁[k,i].  (17)

The three resulting cardioids are pointing in the three directions θ_p1, θ_p2, and θ_p3, defining the corresponding A-format representation, as illustrated in FIG. 4.

Assuming that the obtained cardioids are coincident, the corresponding first order Ambisonic B-format signals can be computed by means of linear combination of the spectra A_pn, The conversion from Ambisonic B-format to A-format is implemented as:

$\begin{array}{cc}\left[\begin{array}{c}{A}_{p_1}\\ A_{p_2}\\ A_{p_3}\end{array}\right]=\Gamma \left[\begin{array}{c}{R}_W\\ R_X\\ R_Y\end{array}\right]=\frac{1}{2}\left[\begin{array}{ccc}1& \cos {\theta }_{p_1}& \sin {\theta }_{p_1}\\ 1& \cos {\theta }_{p_2}& \sin {\theta }_{p_2}\\ 1& \cos {\theta }_{p_3}& \sin {\theta }_{p_3}\end{array}\right]\left[\begin{array}{c}{R}_W\\ R_X\\ R_Y\end{array}\right]& \left(18\right)\end{array}$

The inverse matrix of (18) enables to convert the cardioids to Ambisonic B-format,

$\begin{array}{cc}\left[\begin{array}{c}{R}_W\\ R_X\\ R_Y\end{array}\right]={\Gamma }^{-1}\left[\begin{array}{c}{A}_{p_1}\\ A_{p_2}\\ A_{p_3}\end{array}\right]& \left(19\right)\end{array}$

The first order Ambisonic B-format normalized directional responses R_W, R_X, and R_Y, are shown in FIG. 5, where R_W corresponds to a monopole. while the signals R_X and R_Y correspond to two orthogonal dipoles.

In the following, the determining of higher order ambisonic B-format signals is shown. This corresponds to the function of the direction-of-arrival estimator 20 and the higher order ambisonic encoder 21.

Deriving previously, the first order ambisonic B-format signals R_W, R_X, and R_Y for the direct sound, no explicit direction of arrival (DOA) of sound was computed. Instead the directional responses of the three signals R_W, R_X, and R_Y have been obtained from the A-format cardioid signals A_pn in (17).

In order to obtain the higher order (e.g. second and third) ambisonic B-format signals, an explicit DOA is derived based on the two first order ambisonic B-format signals R_X and R_Y as:

$\begin{array}{cc}{\theta }_{\mathrm{XY}}\left[k,i\right]=\arctan \frac{R_Y\left[k,i\right]}{R_X\left[k,i\right]}.& \left(20\right)\end{array}$

Again, assuming three omnidirectional microphones in the horizontal plane (φ=0), the channels of interest as defined in the ambisonic definition in the Table are limited to:

• • order 0: W
  • order 1: X, Y
  • order 2: R, U, V
  • order 3: L, M, P, Q

The other channels are null since they are modulated by sinφ, with φ=0. For each of the above listed channels the directional responses are thus derived by substituting the azimuth angle Θ by the estimated DOA Θ_XY. For instance, considering second order (assuming no elevation, i.e. φ=0):

$\begin{array}{cc}{R}_R\stackrel{\Delta }{=}\left(3{\sin }^2\varphi -1\right)\text{/}2=-1\text{/}2R_S\stackrel{\Delta }{=}\sqrt{3}\text{/}2\cos \theta \sin 2\varphi =0R_T\stackrel{\Delta }{=}\sqrt{3}\text{/}2\sin \theta \sin 2\varphi =0R_U\stackrel{\Delta }{=}\sqrt{3}\text{/}2\cos 2{\mathrm{\theta cos}}^2\varphi =\sqrt{3}\text{/}2\cos 2{\theta }_{\mathrm{XY}}{R}_V\stackrel{\Delta }{=}\sqrt{3}\text{/}2\sin 2{\mathrm{\theta cos}}^2\varphi =\sqrt{3}\text{/}2\sin 2{\theta }_{\mathrm{XY}}& \left(21\right)\end{array}$

The resulting ambisonic channels, R_R, R_U, R_V, R_L, R_M, R_P, and R_Q, contain only the direct sound components of the sound field.

Now, the handling of diffuse sound is shown. This corresponds to the diffuse sound estimator 31 and the de-correlation filter bank 32 of FIG. 2.

In FIG. 7, the occurrence of direct sound from a sound source and omnidirectional diffuse sound is shown in a diagram depicting the locations of two microphones.

In FIG. 8, the directional responses to a sound source of direct sound is shown. Additionally, omnidirectional diffuse sound is depicted.

The previous derivation of the ambisonic B-format signals is only valid under the assumption of direct sound. It does not hold for diffuse sound. In the following a method for obtaining an equivalent diffuse sound for Ambisonic B-format signals is given. Considering enough time after the direct sound and a number of early reflections, numerous reflections are themselves reflected in the space creating a diffuse sound field. By diffuse sound field is mathematically understood as independent sounds having the same energy and coming from all directions, as illustrated in FIG. 7.

It is assumed that X₁ and X₂ can be modelled as:X₁[k,i]=S[k,i]+N₁[k,i],X₂[k,i]=a[k,i]S[k,i]+N₂[k,i],  (22)where a[k,i] is a gain factor, S[k,i] is the direct sound in the left channel, and N₁[k,i] and N₂[k,i] represent diffuse sound. From (22) it follows that:E{X₁X*₁}=E{SS*}+E{N₁N*₁}E{X₂X*₂}=a²E{SS*}+E{N₂N*₂}E{X₁X*₂}=aE{SS*}+E{N₁N*₂}.  (23)

It is reasonable to assume that the amount of diffuse sound in both microphone signals is the same, i.e. E{N₁N*₁}=E{N₂N*₂}=E{NN*}. Furthermore, the normalized cross-correlation coefficient between N₁ and N₂ is denoted Φ_diff and can be obtained from the Cook's,

$\begin{array}{cc}{\Phi }_{\mathrm{diff}}\left[i\right]=\frac{\sin D}{D}\mathrm{with}D=\frac{2\pi {\mathrm{if}}_s{d}_{\mathrm{mic}}}{{\mathrm{cN}}_{\mathrm{STFT}}}.& \left(24\right)\end{array}$ Eventually (23) can be re-written asE{X₁X*₁}=E{SS*}+E{NN*}E{X₂X*₂}=a²E{SS*}+E{NN*}E{X₁X*₂}=aE{SS*}+Φ_diffE{NN*}.  (25)

Elimination of E{SS*} and a in (25) yields the quadratic equation:AE{NN*}²+BE{NN*}+C=0  (26)withA=1−Φ_diff²,B=2Φ_diffE{X₁X*₂}−E{X₁X*₁}−E{X₂X*₂},C=E{X₁X*₁}E{X₂X*₂}−E{X₁X*₂}².  (27)

The power estimate of diffuse sound, denoted P_diff, is then one of the two solutions of (26), the physically possible one (the other solution of (26), yielding a diffuse sound power larger than the microphone signal power, is discarded, as it is physically impossible), i.e.:

$\begin{array}{cc}{P}_{\mathrm{diff}}\left[k,i\right]=E\left\{{\mathrm{NN}}^{*}\right\}=\frac{-B-\sqrt{B^2-4\mathrm{AC}}}{2A}.& \left(28\right)\end{array}$

Note that straightforwardly the contribution of the direct sound can be computed as:P_dir[k,i]=P_X1[k,i]−P_diff[k,i].  (29)

This corresponds to the function of the diffuse sound estimator 31.

By definition the Ambisonic B-format signals are obtained by projecting the sound field unto the spherical harmonics basis defined in the previous table. Mathematically, the projection corresponds to the integration of the sound field signal over the spherical harmonics.

As illustrated in FIG. 7, due to the orthogonality property of the spherical harmonics basis: projecting mathematically independent sounds from all directions unto this basis will result in three orthogonal components:D_W⊥D_X⊥D_Y.  (30)

Note that this property does not hold anymore for direct sound, since a sound source emitting from only ne direction projected unto the same basis will result in a single gain equal to the directional responses at the incidence angle of the sound source, leading to non-orthogonal, or in other terms, correlated components R_W, R_X, and R_Y.

However, here, considering a distribution of three omnidirectional microphones, the single diffuse sound estimate (28) is equivalent for all three microphones (or all three microphone pairs). Therefore there is no possibility to retrieve the native diffuse sound components of the Ambisonic B-format signals, i.e. D_W, D_X, and D_Y as they would be obtained separately by projection of the diffuse sound field unto the spherical harmonics basis.

Instead of getting the exact diffuse sound Ambisonic B-format signals, an alternative is to generate three orthogonal diffuse sound components from the single known diffuse sound estimate P_diff. This way, even if the diffuse sound components do not correspond to the native Ambisonic B-format obtained by projection, the most perceptually important property of orthogonality (enabling localization and spatialization) is preserved. This can be achieved by using de-correlation filters.

The de-correlation filters are derived from a Gaussian noise sequence u of given length l_u. A Gram-Schmidt process applied to this sequence leads to N_u orthogonal sequences U₁, U₂, Λ, U_Nu which serve as filters to generate N_u orthogonal diffuse sounds. In the three microphones case described previously (N_u=3):

Given the length l_u of the noise Gaussian noise sequence u, the de-correlation filters are shaped such that they have an exponential decay over time, similarly as reverberation is a room. To do so, the sequences U₁, U₂, Λ, U_N are multiplied with an exponential window w_u with a time constant corresponding to the reverberation time RT₆₀:

$\begin{array}{cc}{w}_u\left[n\right]=\exp \left(-\frac{0.5\ln 1e6n}{f_s{\mathrm{RT}}_{60}}\right)\mathrm{with}-l_u<n<l_u.& \left(31\right)\end{array}$

In FIG. 9, the filter response of a filter of the de-correlation filter bank 32 of FIG. 2 is shown. Especially the time constant of such a filter is depicted.

The exponential decay of the de-correlation filters, illustrated in FIG. 9, will directly have an influence on the diffuse sound components in the B-format signals. A long decay will over emphasize the diffuse sound contribution in the final B-format but will ensure better separation between the three diffuse sound components.

Eventually, the resulting de-correlation filters are modulated by the diffuse-field responses of the ambisonic B-format channels they correspond to. This way the amount of diffuse sound in each ambisonic B-format channel matches the amount of diffuse sound of a natural B-format recording. The diffuse-field response DFR is the average of the corresponding spherical harmonic directional-response-squared contributions considering all directions, i.e.:

$\begin{array}{cc}\mathrm{DFR}=\frac{1}{4\pi }\underset{-\frac{\pi }{2}}{\overset{\frac{\pi }{2}}{\int }}\underset{-\pi }{\overset{\pi }{\int }}{D\left(\theta ,\varphi \right)}^2\cos \varphi d\theta d\varphi .& \left(32\right)\end{array}$

In the three microphones case (N_u=3), the resulting de-correlations filters are:{tilde over (D)}_W[k,i]=DFR_Ww_uU₁P_2D-diff[k,i],{tilde over (D)}_X[k,i]=DFR_Xw_uU₂P_2D-diff[k,i],{tilde over (D)}_Y[k,i]=DFR_Yw_uU₃P_2D-diff[k,i].  (33)

This way, the orthogonality property between all three diffuse sounds being preserved any further processing using the generated B-format will work on diffuse sound too, i.e., using conventional ambisonic decoding.

Eventually both direct and diffuse sound contributions have to be mixed together in order to generate the full Ambisonic B-format. Given the assumed signal model, the direct and diffuse sounds are, by definition, orthogonal, too. Thus the complete Ambisonic B-format signal are obtained using a straightforward addition:B_W[k,i]=R_W[k,i]+{tilde over (D)}_W[k,i],B_X[k,i]=R_X[k,i]+{tilde over (D)}_X[k,i],B_Y[k,i]=R_Y[k,i]+{tilde over (D)}_Y[k,i].  (34)This addition is performed by the adder 40 of FIG. 2.

After this addition, only the inverse short-time Fourier transformation by the inverse short-time Fourier transformer 41 is performed in order to achieve the output B-format ambisonic signals.

Finally, in FIG. 10, an embodiment of the audio encoding method according to the third aspect of the present disclosure is shown. In a first optional step 100 at least 3 audio signals are recorded. In a second step 101, angles of incidence of direct sound are estimated, by estimating for each pair of the N audio signals an angle of incidence of direct sound. In a third step 102, A-format direct sound signals are derived from the estimated angles of incidence, by deriving from each estimated angle of incidence an A-format direct sound signal, each A-format direct sound signal being a first-order virtual microphone signal. In a fourth step 103, the ambisonic A-format direct sound signals are encoded to first-order ambisonic B-format direct sound signals by applying at least one transformation matrix to the A-format direct sound signals. Note that the fourth step of performing the encoding is an optional step with regard to the third aspect of the present disclosure. In a further optional fifth step 104, a higher order ambisonic B-Format signal is generated based on direction of arrival derived from first order B-Format.

Note that the audio encoding device according to the first aspect of the present disclosure as well as the audio recording device according to the second aspect of the present disclosure relate very closely to the audio encoding method according to the third aspect of the present disclosure. Therefore, the elaborations along FIG. 1-9 are also valid with regard to the audio encoding method shown in FIG. 10.

These encoded signals are fully compatible with conventional Ambisonic B-format signals, and thus, can be used as input for Ambisonic B-format decoding or any other processing. The same principle can be applied to retrieve full higher order Ambisonic B-format signals with both direct and diffuse sounds contributions.

Abbreviations and Notations

Abbreviation Definition
VR           Virtual Reality
DirAc        Directional Audio Coding
DOA          Direction Of Arrival
STFT         short-Time Fourier Transform
SN3D         Schmidt semi-Normalization 3D
DFR          Diffuse-Field Response
SNR          Signal to Noise Ratio
HOA          High Order Ambisonic

Notation                   Definition
x1, x2                     Both recorded microphone signals
X1[k, i]                   STFT of x1 in frame k and frequency bin i
S[k, i]                    STFT of source signal
N1[k, i]                   Diffuse noise in microphone 1
αX                         Forgeting factor
TX                         averaging time-constant
X12 [k, i]                 cross-spectrum two microphone signal 1 and 2
fs                         sampling frequency
falias                     Frequency aliasing
dmic                       Distance between both microphones
E { }                      Expectation oparator
θ and ϕ                    azimuth and elevation angles
Pdiff                      power estimate of diffuse noise
RW, RX, RY                 First order Ambisonic components
RR, RU, RV, RL, RM,        Higher order Ambisonic components
RP, and RQ
P2D-diff                   power estimate of diffuse noise in 2D
U1, U2, Λ, UNu             Orthogonal sequences
{tilde over (ψ)}12         Angle of the complex cross-spectrum X12
Ψ12                        The mean of unwrapped phase ψ12 over
                           frequency aliasing
l[i]                       An uncertainty integer which depends on
                           frequency i
L[i]                       Upper bound function for l[i] which depends
                           on frequency i
D(θ, ϕ)                    Spherical representation of the Ambisonic
                           channels
Ap1, Ap2, Ap3, . . . , Apn The cardioids that each of them generated
                           with pair of microphones
RT60                       Reverberation time
lu                         Length of Gaussian noise sequence u
wu                         Exponential window
DFRW, DFRX, DFRY           Diffuse-Field Responses for W, X, Y
                           components

The present disclosure is not limited to the examples and especially not to a specific number of microphones. The characteristics of the exemplary embodiments can be used in any advantageous combination.

The present disclosure has been described in conjunction with various embodiments herein. However, other variations to the disclosed embodiments can be understood and effected by those skilled in the art in practicing the claimed invention, from a study of the drawings, the disclosure and the appended claims. In the claims, the word “comprising” does not exclude other elements or steps and the indefinite article “a” or “an” does not exclude a plurality. A single processor or other unit may fulfill the functions of several items recited in the claims. The mere fact that certain measures are recited in usually different dependent claims does not indicate that a combination of these measures cannot be used to advantage. A computer program may be stored/distributed on a suitable medium, such as an optical storage medium or a solid-state medium supplied together with or as part of other hardware, but may also be distributed in other forms, such as via the internet or other wired or wireless communication systems.

Claims

1. An audio encoding device, for encoding N audio signals, from N microphones where N≥3, the audio encoding device comprising: a delay estimator configured to estimate angles of incidence of direct sound by estimating, for each pair of the N audio signals, an angle of incidence of the direct sound, and a beam deriver configured to derive A-format direct sound signals from the estimated angles of incidence by deriving, from each of the estimated angles of incidence, a respective one of the A-format direct sound signals, each of the A-format direct sound signals being a first-order virtual microphone signal; andan encoder configured to encode the A-format direct sound signals in first-order ambisonic B-format direct sound signals by applying a transformation matrix to the A-format direct sound signals,wherein N=3,wherein the audio encoding device comprises a short time Fourier transformer configured to perform a short time Fourier transformation on each of the N audio signals x1, x2, x3, resulting in N short time Fourier transformed audio signals X1[k,i], X2[k,i], X3[k,i],wherein the delay estimator is configured to: determine cross spectra of each pair of the short time Fourier transformed audio signals according to: X12[k,i]=αXX1[k,i]X*2[k,i]+(1−αX)X12[k−1,i],X13[k,i]=αXX1[k,i]X*3[k,i]+(1−αX)X13[k−1,i], andX23[k,i]=αXX2[k,i]X*3[k,i]+(1−αX)X23[k−1,i],determine an angle of the complex cross spectrum of each pair of the short time Fourier transformed audio signals according to:

2. The audio encoding device according to claim 1, wherein the beam deriver is configured to:determine cardioid directional responses according to:

3. The audio encoding device according to claim 2, wherein the encoder is configured to encode the A-format direct sound signals to the first-order ambisonic B-format direct sound signals according to:

4. The audio encoding device according to claim 1, comprising a direction of arrival estimator configured to estimate a direction of arrival from the first-order ambisonic B-format direct sound signals, anda higher order ambisonic encoder configured to encode higher order ambisonic B-format direct sound signals using the first-order ambisonic B-format direct sound signals and the estimated direction of arrival, wherein higher order ambisonic B-format direct sound signals have an order higher than one.

5. The audio encoding device according to claim 4, wherein the direction of arrival estimator is configured to estimate the direction of arrival according to:

6. The audio encoding device according to claim 5, wherein the higher order ambisonic B-format direct sound signals comprise second order ambisonic B-format direct sound signals limited to two dimensions,wherein the higher order ambisonic encoder is configured to encode the second order ambisonic B-format direct sound signals according to:

7. The audio encoding device according to claim 1, comprising a microphone matcher configured to perform a matching of the N frequency domain audio signals, resulting in N matched frequency domain audio signals.

8. The audio encoding device according to claim 7, comprising a diffuse sound estimator configured to estimate a diffuse sound power, anda de-correlation filter bank configured to perform a de-correlation of the diffuse sound power by generating three orthogonal diffuse sound components from the diffuse sound estimate power.

9. The audio encoding device according to claim 8, wherein the diffuse sound estimator is configured to estimate the diffuse sound power according to:

10. The audio encoding device according to claim 9, wherein the de-correlation filter bank is configured to perform the de-correlation of the diffuse sound power by generating three orthogonal diffuse sound components from the diffuse sound estimate power: {tilde over (D)}W[k,i]=DFRWwuU1P2D-diff[k,i],{tilde over (D)}X[k,i]=DFRXwuU2P2D-diff[k,i], and{tilde over (D)}Y[k,i]=DFRYwuU3P2D-diff[k,i],wherein:

11. The audio encoding device according to claim 1, comprising an adder, which is configured to add channel-wise, the first-order ambisonic B-format direct sound signals and the higher order ambisonic B-format direct sound signals, and/or the diffuse sound signals, resulting in complete ambisonic B-format signals.

12. The audio encoding device according to claim 1, wherein delay estimator configured to estimate the angle of incidence for each pair of the N audio signal based on a travelling time delay between the pair of audio signals.

13. The audio encoding device according to claim 1, wherein delay estimator configured to estimate the angle of incidence for each pair of the N audio signal based on a delay in second and a delay in samples between the pair of audio signals.

14. An audio recording device comprising the N microphones configured to record the N audio signals, and the audio encoding device according to claim 1.

15. A method for encoding N audio signals, from N microphones where N≤3, the method comprising: estimating angles of incidence of direct sound by estimating for each pair of the N audio signals an angle of incidence of the direct sound,deriving A-format direct sound signals from the estimated angles of incidence by deriving, from each of the estimated angles of incidence, a respective one of the A-format direct sound signals, each of the A-format direct sound signals being a first-order virtual microphone signal, andencoding the A-format direct sound signals in first-order ambisonic B-format direct sound signals by applying a transformation matrix to the A-format direct sound signals,wherein N=3,wherein the encoding further comprises performing a short time Fourier transformation on each of the N audio signals x1, x2, x3, resulting in N short time Fourier transformed audio signals X1[k,j], X2[k,j], X3[k,j],wherein the method further comprises:determining cross spectra of each pair of the short time Fourier transformed audio signals according to: X12[k,i]=αXX1[k,i]X2*[k,i]+(1−αX)X12[k−1,i], X13[k,i]=αXX1[k,i]X3*[k,i]+(1−αX)X13[k−1,i], and X23[k,i]=αXX2[k,i]X3*[k,i]+(1−αX)X23[k−1,i], determining an angle of the complex cross spectrum of each pair of the short time Fourier transformed audio signals according to:

16. A non-transitory computer readable storage medium comprising a computer program with a program code, which is configured to be executed by a computer to cause the computer to perform the method according to claim 15.