Automatic loudspeaker directivity adaptation
US-2024236597-A1 · Jul 11, 2024 · US
US9666195B2 · US · B2
| Field | Value |
|---|---|
| Publication number | US-9666195-B2 |
| Application number | US-201314386784-A |
| Country | US |
| Kind code | B2 |
| Filing date | Mar 20, 2013 |
| Priority date | Mar 28, 2012 |
| Publication date | May 30, 2017 |
| Grant date | May 30, 2017 |
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Decoding of Ambisonics representations for a stereo loudspeaker setup is known for first-order Ambisonics audio signals. But such first-order Ambisonics approaches have either high negative side lobes or poor localization in the frontal region. The invention deals with the processing for stereo decoders for higher-order Ambisonics HOA. The desired panning functions can be derived from a panning law for placement of virtual sources between the loudspeakers. For each loudspeaker a desired panning function for all possible input directions at sampling points is defined. The panning functions are approximated by circular harmonic functions, and with increasing Ambisonics order the desired panning functions are matched with decreasing error. For the frontal region between the loudspeakers, a panning law like the tangent law or vector base amplitude panning (VBAP) are used. For the rear directions panning functions with a slight attenuation of sounds from these directions are defined.
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The invention claimed is: 1. Method for decoding stereo loudspeaker signals l(t) from a three-dimensional spatial higher-order Ambisonics audio signal a(t), with t designating time, from azimuth angle values φ L and φ R of left and right loudspeakers, and from S sampling points on a circle, said method including the steps: receiving said audio signal a(t), calculating by at least one processor, from azimuth angle values Φ of left and right loudspeakers and from the number S of virtual sampling points on a circle, a matrix G containing desired panning function values for all virtual sampling points, wherein G = [ g L ( ϕ 1 ) … g L ( ϕ S ) g R ( ϕ 1 ) … g R ( ϕ S ) ] and the g L (φ) and g R (φ) elements are the panning functions and g L (φ S ) and g R (φ S ) are the values at the S different sampling points corresponding respectively to values Φ 1 , Φ 2 . . . Φ S of said azimuth angle value Φ, determining by said at least one processor the order N of said Ambisonics audio signal a(t); calculating by said at least one processor from said number S and from said order N a mode matrix Ξ and the corresponding pseudo-inverse Ξ + of said mode matrix Ξ, wherein Ξ=[y*(φ 1 ), y*(φ 2 ), . . . , y*(φ S )] and y*(φ)=[Y −N *(φ), . . . , Y 0 *(φ), . . . , Y N *(φ)] T is the complex conjugation of the circular harmonics vector y(φ)=[Y −N (φ), . . . , Y 0 (φ), . . . , Y N (φ)] T of said Ambisonics audio signal a(t) and Y m (φ) are the circular harmonic functions, with m being an integer comprises between −N and N; calculating by said from at least one processor from said matrices G and Ξ + a decoding matrix D=G Ξ + ; calculating by said at least one processor the loudspeaker signals l(t)=Da(t), wherein a 3D-to-2D conversion of a(t) is carried out for this calculating, outputting said loudspeaker signals l(t). 2. Method for determining a decoding matrix D that can be used for decoding stereo loudspeaker signals l(t)=Da(t) from a 2-D higher-order Ambisonics audio signal a(t), with t designating time said method including the steps: receiving said audio signal a(t), receiving the order N of said Ambisonics audio signal a(t); calculating by at least one processor, from desired azimuth angle values Φ of left and right loudspeakers and from the number S of virtual sampling points on a circle, a matrix G containing desired panning function values for all virtual sampling points, wherein G = [ g L ( ϕ 1 ) … g L ( ϕ S ) g R ( ϕ 1 ) … g R ( ϕ S ) ] and the g L (φ) and g R (φ) elements are the panning functions and g L (φ S ) and g R (φ S ) are the values at the S different sampling points corresponding respectively to values Φ 1 , Φ 2 , . . . Φ S of said azimuth value Φ, calculating by said at least one processor from said number S and from said order N a mode matrix Ξ and the corresponding pseudo-inverse Ξ + of said mode matrix Ξ, wherein Ξ=[y*(φ 1 ), y*(φ 2 ), y*(φ S )] and =[Y −N *(φ), . . . , Y 0 *(φ), . . . , Y N *(φ)] T is the complex conjugation of the circular harmonics vector y(φ)=[Y −N (φ), . . . , Y 0 (φ), . . . , Y N (φ)] T of said Ambisonics audio signal a(t) and Y m (φ) are the circular harmonic functions, with m being an integer comprises between −N and N; calculating by said at least one processor from said matrices G and Ξ + a decoding matrix D=G Ξ + , calculating by said at lease one processor the loudspeaker signals l(t)=Da(t), wherein a 3D-to-2D conv
of the matrix type, i.e. in which input signals are combined algebraically, e.g. after having been phase shifted with respect to each other · CPC title
in which the audio signals are in digital form, i.e. employing more than two discrete digital channels (data reduction aspects thereof based on psychoacoustics G10L19/02) · CPC title
Application of ambisonics in stereophonic audio systems · CPC title
Non-adaptive circuits, e.g. manually adjustable or static, for enhancing the sound image or the spatial distribution (control circuits for electronic adaptation of the sound field H04S7/30) · CPC title
Multi-channel, i.e. more than two input channels, sound reproduction with two speakers wherein the multi-channel information is substantially preserved · CPC title
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