Abstract: There is provided a method and corresponding system for generating harmonics based on an input signal.

Abstract: A method for determining a decoding L×K matrix for decoding incoming L-dimensional channel audio into outgoing K-dimensional channel audio where L?2 and K?1 is provided. The method comprising: determining a panning control parameter p and a sample component d that minimizes a first difference metric between an L-dimensional input sample x and an estimation of the input sample xest=d a, where a=A(p) and where A(p) is a first pre-set mapping function that returns an L-dimensional panning vector a for a given panning control parameter p; generating a K-dimensional raw output sample yraw=d s, where s=S(p) and where S(p) is a second pre-set mapping function that returns a K-dimensional panning vector s for a given panning control parameter p, and; determining the decoding L×K matrix M by solving an optimization problem that minimizes a second difference metric between the K-dimensional raw output sample yraw and the decoded input sample x M.

Abstract: There is provided a method and corresponding system for processing a set including Head-Related Transfer Functions, HRTFs, for at least two different directions of sound incidence to enable improved generation, modeling or simulation of ear signals corresponding to a diffuse sound field. For each direction of sound incidence to the head, a left-ear HRTF represents the transfer function from a sound source to the left ear, and a right-ear HRTF represents the corresponding transfer function to the right ear. The method comprises applying (S1) a phase adjustment to each HRTF, for each ear and direction, for reducing Interaural Time Differences, ITD, above a threshold frequency or in a frequency band above the threshold frequency; and adding (S2) a direction-dependent Interaural Phase Difference, IPD, to each HRTF, for each ear and direction, for reducing Interaural Coherence when modelling or simulating a diffuse sound field.

Abstract: There is provided a method and corresponding system for generating harmonics based on an input signal.

Abstract: There is provided a method for controlling bass reproduction properties of a multichannel audio system, wherein the audio system has inputs for at least two audio input signals and includes a set of loudspeakers, including at least one bass-capable loudspeaker and at least two high-range loudspeakers, each loudspeaker being associated with a loudspeaker channel. The method includes obtaining impulse responses or transfer functions that represent the sound reproduction properties of each loudspeaker channel at a number of measurement or control positions. The method also includes tuning, when the audio system includes more than one bass-capable loudspeaker, loudspeaker channels of at least two bass loudspeakers to each other so that their sum impulse response has minimum spatial variability, and/or controlling high-range loudspeaker speaker channels to be in-phase with each other and/or with bass-capable loudspeaker channel in a crossover frequency band.

Abstract: A method and corresponding system for generating harmonics from an input signal having an input frequency uses a harmonics generator having an associated gain controller generating adjustable gain control factors, controlling the amplitude of harmonic components. The method includes generating a series of harmonic components in a process including adjustment of relative balances between at least three harmonic components of different orders, the balances being independently adjustable by controlling at least two gain control factors. Each gain control factor is adjusted based on a frequency dependent measure of the input signal or signal derived therefrom, of the frequency dependent measure of the input signal and/or derived signal unique for the corresponding gain control factor. An output signal is created including the harmonic components according to the balances as controlled by the at least two gain control factors, wherein the balances between the harmonic components depend on the input frequency.

Abstract: Disclosed is a system configured to determine filter parameters of an audio precompensation filter for the compensation of an associated sound generating system having at least two loudspeakers. The system is configured to obtain sound field models describing the sound field in at least one region of space designated the bright zone and the sound field in at least one region of space designated the dark zone. The system is also configured to obtain a target sound field in the bright zone. Further, the system is configured to determine filter parameters of the audio precompensation filter so that a criterion function is optimized under the constraint of causality and stability of the dynamics of the audio precompensation filter to enable reproduction of a desired target sound field in the bright zone, while reproducing as little sound as possible in the dark zone.

Abstract: There is provided a method of determining filter coefficients for an audio filter system including a number, N?2, of filter paths for enabling processing of N audio channels, one filter path per channel, wherein each filter path includes at least one audio filter for performing the processing of the corresponding channel. The method includes providing a common set of filter design parameters for a pair of audio filters belonging to different filter paths, including phase difference information representing an inter-channel phase difference and frequency information representing a frequency value as filter design parameters; and determining filter coefficients for the pair of audio filters at least partly based on the common set of filter design parameters.

Abstract: Disclosed is a method for determining an audio filter. The method includes obtaining head-tracking data representing a head pose, also referred to as a head position, relative to a torso pose of an intended subject, and selecting a representation of a Head Related Transfer Function, HRTF, from a HRTF database, based on the head-tracking data and information representative of the position and/or direction of a sound source. The method also includes determining the audio filter based on the selected HRTF representation.

Abstract: There is provided a method for processing an audio input signal and a corresponding audio filter system. The method includes applying non-linear time-domain processing to the input signal to generate a processed copy of the input signal, transforming the input signal and the processed copy of the input signal to the frequency domain, and performing a comparison in the frequency-domain based on the transformed input signal and the transformed processed copy of the input signal. The method also includes determining at least one control parameter based on the comparison, performing frequency-domain processing of the transformed input signal based on the control parameter, and generating an output signal by transforming the frequency-domain processed signal to the time domain.

Abstract: There is provided a method and corresponding system for determining phase adjustment filters for an associated sound generating system including at least two audio reproduction channels C1 and C2 where each of the audio reproduction channels C1 and C2 has an input signal and at least one loudspeaker located in a listening environment. The method includes estimating, for each of the audio reproduction channels C1 and C2, an acoustic transfer function at each of M?1 spatial positions in the listening environment, based on sound measurements at the spatial positions; and determining, based on the acoustic transfer functions, phase adjustment filters F1 and to be applied, respectively, to the audio reproduction channels C1 and C2, to reduce the inter-loudspeaker differential phase, IDP, between the audio reproduction channels C1 and C2 in p listener positions.

Abstract: Disclosed is a method for determining filter coefficients of an audio precompensation controller for the compensation of an associated sound system, including N?2 loudspeakers, including estimating, for each one of at least a pair of the loudspeakers, a model transfer function at each of M control points distributed in Z?2 spatially separated listening zones in a listening environment of the sound system. The method also includes determining, for each of the M control points, a zone-dependent target transfer function at least based on the zone affiliation of the control point; and determining the filter coefficients of the audio precompensation controller at least based on the model transfer functions and the target transfer functions of the M control points. Consequently, an audio precompensation controller for an associated sound system can be obtained that enables improved and/or customized sound reproduction in two or more listening zones simultaneously.

Abstract: Disclosed is a method and a system to determine an audio precompensation controller for an associated sound generating system including a total of N?2 loudspeakers, each having a loudspeaker input. The audio precompensation controller has a number L?1 inputs for L input signals and N outputs for N controller output signals, one to each loudspeaker. For each one of at least a subset of the N loudspeaker inputs, an impulse response is estimated at each measurement position. For each one of the L input signal(s), a selected one of the N loudspeakers is specified as a primary loudspeaker and a selected subset S including at least one of the N loudspeakers as support loudspeaker(s).

Abstract: Disclosed is a method for determining filter coefficients of an audio precompensation controller for the compensation of an associated sound system, including N?2 loudspeakers, including estimating, for each one of at least a pair of the loudspeakers, a model transfer function at each of M control points distributed in Z?2 spatially separated listening zones in a listening environment of the sound system. The method also includes determining, for each of the M control points, a zone-dependent target transfer function at least based on the zone affiliation of the control point; and determining the filter coefficients of the audio precompensation controller at least based on the model transfer functions and the target transfer functions of the M control points. Consequently, an audio precompensation controller for an associated sound system can be obtained that enables improved and/or customized sound reproduction in two or more listening zones simultaneously.

Abstract: The proposed technology provides an audio decoder (100) configured to receive input signals representative of at least two audio input channels. The audio decoder is configured to provide direct signal paths and cross-feed signal paths (10) for the input signals. The audio decoder is configured to apply head shadowing filters (20) in the direct signal paths and cross-feed signal paths for simulating head shadowing of loudspeakers placed at different angles to an intended listener. The audio decoder is also configured to apply phase shift filters (30) in the direct signal paths and cross-feed signal paths for introducing a phase difference between the direct signal paths and the cross-feed signal paths representing a phase difference occurring between the ears of the intended listener. The audio decoder is further configured to sum (40) the direct and cross-feed signal paths to provide output signals.

Abstract: A method for determining an audio precompensation controller for an associated sound generating system comprising a total of N?2 loudspeakers, each having a loudspeaker input. The audio precompensation controller has a number L?2 inputs for L input signal(s) and N outputs for N controller output signals, one to each loudspeaker. It is relevant to: estimate (S1), for each one of at least a subset of the N loudspeaker inputs, an impulse response at each measurement position; specify (S2), for each one of the L input signal(s), a selected one of the N loudspeakers as a primary loudspeaker and optionally a selected subset S including at least one of the N loudspeakers as support loudspeaker(s); select (S2) at least one loudspeaker pair, that is required to be symmetrical with respect to the listing position; and specify (S3), for each primary loudspeaker, a target impulse response at each measurement position.

Abstract: A method for determining an audio precompensation controller for an associated sound generating system comprising a total of N?2 loudspeakers, each having a loudspeaker input. The audio precompensation controller has a number L?2 inputs for L input signal(s) and N outputs for N controller output signals, one to each loudspeaker. It is relevant to: estimate (S1), for each one of at least a subset of the N loudspeaker inputs, an impulse response at each measurement position; specify (S2), for each one of the L input signal(s), a selected one of the N loudspeakers as a primary loudspeaker and optionally a selected subset S including at least one of the N loudspeakers as support loudspeaker(s); select (S2) at least one loudspeaker pair, that is required to be symmetrical with respect to the listing position; and specify (S3), for each primary loudspeaker, a target impulse response at each measurement position.

Abstract: Disclosed is a method and a system to determine an audio precompensation controller for an associated sound generating system including a total of N?2 loudspeakers, each having a loudspeaker input. The audio precompensation controller has a number L?1 inputs for L input signals and N outputs for N controller output signals, one to each loudspeaker. For each one of at least a subset of the N loudspeaker inputs, an impulse response is estimated at each measurement position. For each one of the L input signal(s), a selected one of the N loudspeakers is specified as a primary loudspeaker and a selected subset S including at least one of the N loudspeakers as support loudspeaker(s).

Abstract: A scheme to design an audio precompensation controller for a multichannel audio system, with a prescribed number N of loudspeakers in prescribed positions so that listeners positioned in any of P>1 spatially extended listening regions should be given the illusion of being in another acoustic environment that has L sound sources located at prescribed positions in a prescribed room acoustics. The method provides a unified joint solution to the problems of equalizer design, crossover design, delay and level calibration, sum-response optimization and up-mixing. A multi-input multi-output audio precompensation controller is designed for an associated sound generating system including a limited number of loudspeaker inputs for emulating a number of virtual sound sources.

Abstract: A discrete-time audio precompensation filter is designed based on a linear model that describes the dynamic response of a sound generating system at p>1 listening positions. The filter construction is based on providing information representative of n non-minimum phase zeros {Zi} that are outside of the stability region |z|=1 in the complex frequency domain. A causal Finite Impulse Response (FIR) filter, of user-specified degree d, having coefficients corresponding to a causal part of a delayed non-causal impulse response is determined based on the information representative of n non minimum phase zeros. The resulting precompensation filter is determined as the product of at least two scalar dynamic systems, represented by an inverse of a characteristic scalar magnitude response in the frequency domain representing the power gains at the listening positions, and the causal FIR filter designed to approximately invert only non-minimum phase zeros that are safely inverted.