The invention relates to a method and to an apparatus for generating 3D audio scene or object based content from two-channel stereo based content.
The invention is related to the creation of 3D audio scene/object based audio content from two-channel stereo channel based content. Some references related to up mixing two-channel stereo content to 2D surround channel based content include: [2] V. Pulkki, “Spatial sound reproduction with directional audio coding”, J. Audio Eng. Soc., vol. 55, no. 6, pp. 503-516, June 2007; [3] C. Avendano, J. M. Jot, “A frequency-domain approach to multichannel upmix”, J. Audio Eng. Soc., vol. 52, no. 7/8, pp. 740-749, July/August 2004; [4] M. M. Goodwin, J. M. Jot, “Spatial audio scene coding”, in Proc. 125th Audio Eng. Soc. Conv., 2008, San Francisco, Calif.; [5] V. Pulkki, “Virtual sound source positioning using vector base amplitude panning”, J. Audio Eng. Soc., vol. 45, no. 6, pp. 456-466, June 1997; [6] J. Thompson, B. Smith, A. Warner, J. M. Jot, “Direct-diffuse decomposition of multichannel signals using a system of pair-wise correlations”, Proc. 133rd Audio Eng. Soc. Conv., 2012, San Francisco, Calif.; [7] C. Faller, “Multiple-loudspeaker playback of stereo signals”, J. Audio Eng. Soc., vol. 54, no. 11, pp. 1051-1064, November 2006; [8] M. Briand, D. Virette, N. Martin, “Parametric representation of multichannel audio based on principal component analysis”, Proc. 120th Audio Eng. Soc. Conv., 2006, Paris; [9] A. Walther, C. Faller, “Direct-ambient decomposition and upmix of surround signals”, Proc. IWASPAA, pp. 277-280, October 2011, New Paltz, N.Y.; [10] E. G. Williams, “Fourier Acoustics”, Applied Mathematical Sciences, vol. 93, 1999, Academic Press; [11] B. Rafaely, “Plane-wave decomposition of the sound field on a sphere by spherical convolution”, J. Acoust. Soc. Am., 4(116), pages 2149-2157, October 2004.
Additional information is also included in [1] ISO/IEC IS 23008-3, “Information technology—High efficiency coding and media delivery in heterogeneous environments—Part 3: 3D audio”.
Loudspeaker setups that are not fixed to one loudspeaker may be addressed by special up/down-mix or re-rendering processing.
When an original spatial virtual position is altered, timbre and loudness artefacts can occur for encodings of two-channel stereo to Higher Order Ambisonics (denoted HOA) using the speaker positions as plane wave origins.
In the context of spatial audio, while both audio image sharpness and spaciousness may be desirable, the two may have contradictory requirements. Sharpness allows an audience to clearly identify directions of audio sources, while spaciousness enhances a listener's feeling of envelopment.
The present disclosure is directed to maintaining both sharpness and spaciousness after converting two-channel stereo channel based content to 3D audio scene/object based audio content.
A primary ambient decomposition (PAD) may separate directional and ambient components found in channel based audio. The directional component is an audio signal related to a source direction. This directional component may be manipulated to determine a new directional component. The new directional component may be encoded to HOA, except for the centre channel direction where the related signal is handled as a static object channel. Additional ambient representations are derived from the ambient components. The additional ambient representations are encoded to HOA.
The encoded HOA directional and ambient components may be combined and an output of the combined HOA representation and the centre channel signal may be provided.
In one example, this processing may be represented as:
A new format may utilize HOA for encoding spatial audio information plus a static object for encoding a centre channel. The new 3D audio scene/object content can be used when pimping up or upmixing legacy stereo content to 3D audio. The content may then be transmitted based on any MPEG-H compression and can be used for rendering to any loudspeaker setup.
In principle, the inventive method is adapted for generating 3D audio scene and object based content from two-channel stereo based content, and includes:
In principle the inventive apparatus is adapted for generating 3D audio scene and object based content from two-channel stereo based content, said apparatus including means adapted to:
In principle, the inventive method is adapted for generating 3D audio scene and object based content from two-channel stereo based content, and includes: receiving the two-channel stereo based content represented by a plurality of time/frequency (T/F) tiles; determining, for each tile, ambient power, direct power, source directions φs({circumflex over (t)},k) and mixing coefficients; determining, for each tile, a directional signal and two ambient T/F channels based on the corresponding ambient power, direct power, and mixing coefficients;
determining the 3D audio scene and object based content based on the directional signal and ambient T/F channels of the T/F tiles. The method may further include wherein, for each tile, a new source direction is determined based on the source direction φs({circumflex over (t)},k), and, based on a determination that the new source direction is within a predetermined interval, a directional centre channel object signal oc({circumflex over (t)},k) is determined based on the directional signal, the directional centre channel object signal oc({circumflex over (t)},k) corresponding to the object based content, and, based on a determination that the new source direction is outside the predetermined interval, a directional HOA signal bs({circumflex over (t)},k) is determined based on the new source direction. Moreover, for each tile, additional ambient signal channels ({circumflex over (t)},k) may be determined based on a de-correlation of the two ambient T/F channels, and ambient HOA signals ({circumflex over (t)},k) are determined based on the additional ambient signal channels. The 3D audio scene content is based on the directional HOA signals bs({circumflex over (t)},k) and the ambient HOA signals ({circumflex over (t)}, k).
Exemplary embodiments of the invention are described with reference to the accompanying drawings, which show in:
Even if not explicitly described, the following embodiments may be employed in any combination or sub-combination.
wherein c is the speed of sound waves in air.
The following definitions are used in this application (see also
x(t)
xϵ 2
b(t)
bϵ (N+1)
b(t) = [{dot over (b)}1(t), . . . , {dot over (b)}(N+1)
p
c
{circumflex over (Ω)}
Ω
x({circumflex over (t)}, k)
xϵ 2
b({circumflex over (t)}, k)
bϵ (N+1)
a({circumflex over (t)}, k)
aϵ 2
n({circumflex over (t)}, k)
nϵ 2
n = [n1, n2]T
C({circumflex over (t)}, k)
Cϵ 2×2
C({circumflex over (t)}, k) = E(x({circumflex over (t)}, k) x({circumflex over (t)}, k)H), with E( )
({circumflex over (t)}, k)
ϵ L
L ambience channels
y
s ({circumflex over (t)}, k)
y
s
Ψ
L ϵ (N+1)
=
b
s({circumflex over (t)}, k)
({circumflex over (t)}, k)
In one example, an initialisation may include providing to or receiving by a method or a device a channel stereo signal x(t) and control parameters pc (e.g., the two-channel stereo signal x(t) 10 and the input parameter set vector pc 12 illustrated in
The elements of parameter pc may be updated during operation of a system, for example by updating a smooth envelope of these elements or parameters.
A two channel stereo signal x(t) may be transformed by HOA upconverter 11 or 31 into the time/frequency (T/F) domain by a filter bank. In one embodiment a fast fourier transform (FFT) is used with 50% overlapping blocks of 4096 samples. Smaller frequency resolutions may be utilized, although there may be a trade-off between processing speed and separation performance. The transformed input signal may be denoted as x({circumflex over (t)},k) in T/F domain, where {circumflex over (t)} relates to the processed block and k denotes the frequency band or bin index.
In one example, for each T/F tile of the input two-channel stereo signal x(t), a correlation matrix may be determined. In one example, the correlation matrix may be determined based on:
wherein E( ) denotes the expectation operator. The expectation can be determined based on a mean value over tnum temporal T/F values (index {circumflex over (t)}) by using a ring buffer or an IIR smoothing filter.
The Eigenvalues of the correlation matrix may then be determined, such as for example based on:
λ1({circumflex over (t)},k)=½(c22+c11+√{square root over ((c11−c22)2+4|cr12|2)}) Equation No. 2a
λ2({circumflex over (t)},k)=½(c22+c11−√{square root over ((c11−c22)2+4|cr12|2)}) Equation No. 2b
wherein cr12=real(c12) denotes the real part of c12. The indices ({circumflex over (t)},k) may be omitted during certain notations, e.g., as within Equation Nos. 2a and 2b.
For each tile, based on the correlation matrix, the following may be determined: ambient power, directional power, elements of a gain vector that mixes the directional components, and an azimuth angle of the virtual source direction s({circumflex over (t)},k) to be extracted.
In one example, the ambient power may be determined based on the second eigenvalue, such as for example:
P
N({circumflex over (t)},k):PN({circumflex over (t)},k)=λ2({circumflex over (t)},k) Equation No. 3
In another example, the directional power may be determined based on the first eigenvalue and the ambient power, such as for example:
P
s({circumflex over (t)},k):Ps({circumflex over (t)},k)=λ1({circumflex over (t)},k)−PN({circumflex over (t)},k) Equation No. 4
In another example, elements of a gain vector a({circumflex over (t)},k)=[a1({circumflex over (t)},k),a2({circumflex over (t)},k)]T that mixes the directional components into x({circumflex over (t)},k) may be determined based on:
with
The azimuth angle of virtual source direction s({circumflex over (t)},k) to be extracted may be determined based on:
with φx giving the loudspeaker position azimuth angle related to signal x1 in radian (assuming that −φx is the position related to x2).
In this sub section for better readability the indices ({circumflex over (t)},k) are omitted. Processing is performed for each T/F tile ({circumflex over (t)},k). For each T/F tile, a first directional intermediate signal is extracted based on a gain, such as, for example:
The intermediate signal may be scaled in order to derive the directional signal, such as for example, based on:
The two elements of an ambient signal n=[n1,n2]T are derived by first calculating intermediate values based on the ambient power, directional power, and the elements of the gain vector:
followed by scaling of these values:
A new source direction ϕs({circumflex over (t)},k) may be determined based on a stage_width W and, for example, the azimuth angle of the virtual source direction (e.g., as described in connection with Equation No. 6). The new source direction may be determined based on:
ϕs({circumflex over (t)},k)=Wφs({circumflex over (t)},k) Equation No. 11
A centre channel object signal oc({circumflex over (t)},k) and/or a directional HOA signal bs({circumflex over (t)},k) in the T/F domain may be determined based on the new source direction. In particular, the new source direction ϕs({circumflex over (t)},k) may be compared to a center_channel_capture_width cW. If |ϕs({circumflex over (t)},k)|<cW, then
o
c({circumflex over (t)},k)=s({circumflex over (t)},k) and bs({circumflex over (t)},k)=0 Equation No. 12a
else:
o
c({circumflex over (t)},k)=0 and bs({circumflex over (t)},k)=ys({circumflex over (t)},k)s({circumflex over (t)},k) Equation No. 12b
where ys({circumflex over (t)},k) is the spherical harmonic encoding vector derived from {circumflex over (φ)}s({circumflex over (t)},k) and a direct sound encoding elevation θS. In one example, the ys({circumflex over (t)},k) vector may be determined based on the following:
y
s({circumflex over (t)},k)=[Y00(θS,ϕs),Y1−1(θS,ϕs), . . . ,YNN(θS,ϕs)]T Equation No. 13
The ambient HOA signal ({circumflex over (t)},k) may be determined based on the additional ambient signal channels ({circumflex over (t)},k). For example, the ambient HOA signal ({circumflex over (t)},k) may be determined based on:
({circumflex over (t)},k)=diag(gL)({circumflex over (t)},k) Equation No. 14
where diag(gL) is a square diagonal matrix with ambient gains gL on its main diagonal, ({circumflex over (t)},k) is a vector of ambient signals derived from n and is a mode matrix for encoding ({circumflex over (t)},k) to HOA. The mode matrix may be determined based on:
=, . . . ,], =[Y00(θL,ϕL),Y1−1(θL,ϕL), . . . ,YNN,(θL,ϕL)]T Eq No. 15
wherein, L denotes the number of components in ({circumflex over (t)},k).
In one embodiment L=6 is selected with the following positions:
The vector of ambient signals is determined based on:
with weighting (filtering) factors Fi(k)ϵ1, wherein
di is a delay in samples, and ai(k) is a spectral weighting factor (e.g. in the range 0 to 1).
The combined HOA signal is determined based on the directional HOA signal bs({circumflex over (t)},k) and the ambient HOA signal ({circumflex over (t)},k). For example:
b({circumflex over (t)},k)=bs({circumflex over (t)},k)+({circumflex over (t)},k) Equation No. 18
The T/F signals b({circumflex over (t)},k) and oc({circumflex over (t)},k) are transformed back to time domain by an inverse filter bank to derive signals b(t) and oc(t). For example, the T/F signals may be transformed based on an inverse fast fourier transform (IFFT) and an overlap-add procedure using a sine window.
The signals b(t) and oc(t) and related metadata, the maximum HOA order index N and the direction
of signal oc(t) may be stored or transmitted based on any format, including a standardized format such as an MPEG-H 3D audio compression codec. These can then be rendered to individual loudspeaker setups on demand.
In this section the detailed deduction of the PAD algorithm is presented, including the assumptions about the nature of the signals. Because all considerations take place in T/F domain indices ({circumflex over (t)},k) are omitted.
The following signal model in time frequency domain (T/F) is assumed:
x=as+n, Equation No. 19a
x
1
=a
1
s+n
1, Equation No. 19b
x
2
=a
2
s+n
2, Equation No. 19c
√{square root over (a12+a22)}=1 Equation No. 19d
The covariance matrix becomes the correlation matrix if signals with zero mean are assumed, which is a common assumption related to audio signals:
wherein E( ) is the expectation operator which can be approximated by deriving the mean value over T/F tiles.
Next the Eigenvalues of the covariance matrix are derived. They are defined by
λ1,2(C)={x:det(C−xI)=0}. Equation No. 21
Applied to the covariance matrix:
The solution of λ1,2 is:
λ1,2=½(c22+c11±√{square root over ((c11−c22)2+4|c12|2)}) Equation No. 23
The model assumptions and the covariance matrix are given by:
The model covariance becomes
In the following real positive-valued mixing coefficients a1, a2 and √{square root over (a12+a22)}=1 are assumed, and consequently cr12=real(c12). The Eigenvalues become:
The ambient power estimate becomes:
P
N=λ2=½(c22+c11−√{square root over ((c11−c22)2+4|cr12|2)}) Equation No. 26
The direct sound power estimate becomes:
P
S=λ1−PN=√{square root over ((c11−c22)2+4|cr12|2)} Equation No. 27
The ratio A of the mixing gains can be derived as:
with a12=1−a22, and a22=1−a12 it follows:
The principal component approach includes:
The first and second Eigenvalues are related to Eigenvectors v1,v2 which are given in mathematical literature and in [8] by
Here the signal x1 would relate to the x-axis and the signal x2 would relate to the y-axis of a Cartesian coordinate system. This would map the two channels to be 90° apart with relations: cos({circumflex over (φ)})=a1s/s, sin({circumflex over (φ)})=a2s/s. Thus the ratio of the mixing gains can be used to derive {circumflex over (φ)}, with:
The preferred azimuth measure φ would refer to an azimuth of zero placed half angle between related virtual speaker channels, positive angle direction in mathematical sense counter clock wise. To translate from the above-mentioned system:
The tangent law of energy panning is defined as
where φo is the half loudspeaker spacing angle. In the model used here,
It can be shown that
Based on
Mapping the angle φ to a real loudspeaker spacing includes: Other speaker φx spacings than the 90°
addressed in the model can be addressed based on either:
or more accurate
To encode the directional signal to HOA with limited order, the accuracy of the first method
is regarded as being sufficient.
The directional signal is extracted as a linear combination with gains gT=[g1,g2] of the input signals:
ŝ:=g
T
x=g
T(as+n) Equation No. 35a
The error signal is
err=s−gT(as+n) Equation No. 35b
and becomes minimal if fully orthogonal to the input signals x with ŝ=s:
E(xerr*)=0 Equation No. 36
aP
ŝ
ag
T
aP
ŝ
+gP
n=0 Equation No. 37
taking in mind the model assumptions that the ambient components are not correlated:
(E(n1n2*)=0) Equation No. 38
Because the order of calculation of a vector product of the form gTa is interchangeable, gTa=agT:
(aaTpŝ+IPN)g=aPŝ Equation No. 39
The term in brackets is a quadratic matrix and a solution exists if this matrix is invertible, and by first setting Pŝ=Ps the mixing gains become:
Solving this System Leads to:
The solution is scaled such that the power of the estimate ŝ becomes Ps, with
The unscaled first ambient signal can be derived by subtracting the unscaled directional signal component from the first input channel signal:
{circumflex over (n)}
1
=x
1
−a
1
ŝ=x
1
−a
1
g
T
x:=h
T
x Equation No. 43
Solving this for {circumflex over (n)}1=hTx leads to
The solution is scaled such that the power of the estimate {circumflex over (n)}1 becomes PN, with
The unscaled second ambient signal can be derived by subtracting the rated directional signal component from the second input channel signal
{circumflex over (n)}
2
=x
2
−a
2
ŝ=x
2
−a
2
g
T
x:=w
T
X Equation No. 46
Solving this for {circumflex over (n)}2=wTX leads to
The solution is scaled such that the power P{circumflex over (n)} of the estimate {circumflex over (n)}2 becomes PN, with
Using the covariance matrix, the channel power estimate of x can be expressed by:
P
x
=tr(C)=tr(E(xxH))=E(tr(xxH))=E(tr(xHx))=E(xHx) Eq No. 49
with E( ) representing the expectation and tr( ) representing the trace operators.
When returning to the signal model from section Primary ambient decomposition in T/F domain and the related model assumptions in T/F domain:
x=as+n, Equation No. 50a
x
1
=a
1
s+n
1, Equation No. 50b
x
2
=a
2
s+n
2, Equation No. 50c
√{square root over (a12+a22)}=1, Equation No. 50d
the channel power estimate of x can be expressed by:
P
x
=E(xHx)=Ps+2PN Equation No. 51
The value of Px may be proportional to the perceived signal loudness. A perfect remix of x should preserve loudness and lead to the same estimate.
During HOA encoding, e.g., by a mode-matrix Y(Ωx), the spherical harmonics values may be determined from directions Ωx of the virtual speaker positions:
b
x1
=Y(Ωx)x Equation No. 52
HOA rendering with rendering matrix D with near energy preserving features (e.g., see section 12.4.3 of Reference [1]) may be determined based on:
where I is the unity matrix and (N+1)2 is a scaling factor depending on HOA order N:
{hacek over (x)}=DY(Ωx)x Equation No. 54
The signal power estimate of the rendered encoded HOA signal becomes:
The following may be determined then:
P
{hacek over (x)}
P
x, Equation No. 55c
This may lead to:
Y(Ωx)HY(Ωx):=(N+1)2I, Equation No. 56
which usually cannot be fulfilled for mode matrices related to arbitrary positions. The consequences of Y(Ωx)HY(Ωx) not becoming diagonal are timbre colorations and loudness fluctuations. Y(Ωid) becomes a un-normalised unitary matrix only for special positions (directions) Ωid where the number of positions (directions) is equal or bigger than (N+1)2 and at the same time where the angular distance to next neighbour positions is constant for every position (i.e. a regular sampling on a sphere).
Regarding the impact of maintaining the intended signal directions when encoding channels based content to HOA and decoding:
Let x=as, where the ambient parts are zero. Encoding to HOA and rendering leads to {circumflex over (x)}=D Y(Ωx)a s.
Only rendering matrices satisfying D Y(Ωx)=I would lead to the same spatial impression as replaying the original. Generally, D=Y(Ωx)−1 does not exist and using the pseudo inverse will in general not lead to D Y(Ωx)=I.
Generally, when receiving HOA content, the encoding matrix is unknown and rendering matrices D should be independent from the content.
sumEn=gnl2+gnr2 Equation No. 57
The top part shows VBAP or tangent law amplitude panning gains. The mid and bottom parts show naive HOA encoding and 2-channel rendering of a VBAP panned signal, for N=2 in the mid and for N=6 at the bottom. Perceptually the signal gets louder when the signal source is at mid position, and all directions except the extreme side positions will be warped towards the mid position. Section 6a of
a naive HOA encoding and 2-channel rendering of VBAP panned signal for N=2. Section 6c relates to naive HOA encoding and 2-channel rendering of VBAP panned signal for N=6.
x=as+n Equation No. 58a
after performing PAD and HOA upconversion leads to
b
x2
=y
s
s+
{circumflex over (n)}, Equation No. 58b
with
{circumflex over (n)}=diag(gL) Equation No. 58c
The power estimate of the rendered HOA signal becomes:
For N3D normalised SH:
y
s
H
y
s=(N+1)2 Equation No. 60
and, taking into account that all signals of ii are uncorrelated, the same applies to the noise part:
P
{tilde over (x)}
≈P
s+Σl=1LPn
and ambient gains gL=[1,1,0,0,0,0] can be used for scaling the ambient signal power
Σl=1LPn
and
P
{tilde over (x)}
=P
x. Equation No. 62b
The intended directionality of s now is given by Dys which leads to a classical HOA panning vector which for stage_width W=1 captures the intended directivity.
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, see [1]. In that case the spatio-temporal behaviour of the sound pressure p(t,x) at time t and position {circumflex over (Ω)} within the area of interest is physically fully determined by the homogeneous wave equation. Assumed is a spherical coordinate system of
A Fourier transform (e.g., see Reference [10]) of the sound pressure with respect to time denoted by t(⋅), i.e.
P(ω,{circumflex over (Ω)})=t(p(t,{circumflex over (Ω)}))=∫−∞∞p(t,{circumflex over (Ω)})e−iωtdt, Equation No. 63
with ω denoting the angular frequency and i indicating the imaginary unit, can be expanded into a series of Spherical Harmonics according to
P(ω=kcs,r,δ,ϕ)=Σn=0NΣm=−nn(k)jn(kr)Ynm(θ,ϕ) Equation No. 64
Here cs denotes the speed of sound and k denotes the angular wave number, which is related to the angular frequency ω by
Further, jn(⋅) denote the spherical Bessel functions of the first kind and Ynm(θ,ϕ) denote the real valued Spherical Harmonics of order n and degree m, which are defined below. The expansion coefficients Anm(k) only depend on the angular wave number k. It has been implicitly assumed that sound pressure is spatially band-limited. Thus, the series is truncated with respect to the order index n at an upper limit N, which is called the order of the HOA representation.
If the sound field is represented by a superposition of an infinite number of harmonic plane waves of different angular frequencies ω and arriving from all possible directions specified by the angle tuple (θ,ϕ), the respective plane wave complex amplitude function B(ω,θ,ϕ) can be expressed by the following Spherical Harmonics expansion
B(ω=kcs,θ,ϕ)=Σn=0NΣm=−nnBnm(k)Ynm(θ,ϕ) Equation No. 65
where the expansion coefficients Bnm(k) are related to the expansion coefficients Anm(k) by
A
n
m(k)=inBnm(k) Equation No. 66
Assuming the individual coefficients Bnm(ω=kcs) to be functions of the angular frequency ω, the application of the inverse Fourier transform (denoted by −1(⋅) provides time domain functions
for each order n and degree m, which can be collected in a single vector b(t) by
The position index of a time domain function bnm(t) within the vector b(t) is given by n(n+1)+1+m. The overall number of elements in the vector b(t) is given by 0=(N+1)2.
The final Ambisonics format provides the sampled version b(t) using a sampling frequency fS as
{b(lTS)}={b(TS),b(2TS),b(3TS),b(4TS), . . . }, Equation No. 69
where TS=1/fS denotes the sampling period. The elements of b(lTS) are here referred to as Ambisonics coefficients. The time domain signals bnm(t) and hence the Ambisonics coefficients are real-valued.
The real-valued spherical harmonics Ynm(θ,ϕ) (assuming N3D normalisation) are given by
with
The associated Legendre functions Pn,m(x) are defined as
with the Legendre polynomial Pn(x) and without the Condon-Shortley phase term (−1)m.
The mode matrix Ψ(N
Ωq(N
related to order N2 is defined by
Ψ(N
with yq(N
=[Y00(Ωq(N
denoting the mode vector of order N1 with respect to the directions Ωq(N
A digital audio signal generated as described above can be related to a video signal, with subsequent rendering.
At 720, direct and ambient components are determined. For example, the direct and ambient components may be determined in the T/F domain. At 730, audio scene (e.g., HOA) and object based audio (e.g., a centre channel direction handled as a static object channel) may be determined. The processing at 720 and 730 may be performed in accordance with the principles described in connection with A-E and Equation Nos. 1-72.
It should be noted that the description and drawings merely illustrate the principles of the proposed methods and apparatus. It will thus be appreciated that those skilled in the art will be able to devise various arrangements that, although not explicitly described or shown herein, embody the principles of the invention and are included within its spirit and scope. Furthermore, all examples recited herein are principally intended expressly to be only for pedagogical purposes to aid the reader in understanding the principles of the proposed methods and apparatus and the concepts contributed by the inventors to furthering the art, and are to be construed as being without limitation to such specifically recited examples and conditions. Moreover, all statements herein reciting principles, aspects, and embodiments of the invention, as well as specific examples thereof, are intended to encompass equivalents thereof.
The methods and apparatus described in the present document may be implemented as software, firmware and/or hardware. Certain components may e.g. be implemented as software running on a digital signal processor or microprocessor. Other components may e.g. be implemented as hardware and or as application specific integrated circuits. The signals encountered in the described methods and apparatus may be stored on media such as random access memory or optical storage media. They may be transferred via networks, such as radio networks, satellite networks, wireless networks or wireline networks, e.g. the Internet.
The described processing can be carried out by a single processor or electronic circuit, or by several processors or electronic circuits operating in parallel and/or operating on different parts of the complete processing.
The instructions for operating the processor or the processors according to the described processing can be stored in one or more memories. The at least one processor is configured to carry out these instructions.
Number | Date | Country | Kind |
---|---|---|---|
15306544.6 | Sep 2015 | EP | regional |
This application is division of U.S. patent application Ser. No. 15/761,351, filed Mar. 19, 2018, which claims priority to European Patent Application No. 15306544.6, filed on Sep. 30, 2015, which is incorporated herein by reference in its entirety.
Number | Date | Country | |
---|---|---|---|
Parent | 15761351 | Mar 2018 | US |
Child | 16560733 | US |