Dirac Research Ab

Anders Ahlen

Adrian Bahne

Lars-Johan Brännmark

Mikael Sternad

Lars Johan Brannmark

Disclosed is a method for determining filter coefficients of an audio precompensation controller for the compensation of an associated sound system, including N&#x2265;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&#x2265;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.

Personal multichannel audio controller design

Disclosed is a method and a system to determine an audio precompensation controller for an associated sound generating system including a total of N&#x2267;2 loudspeakers, each having a loudspeaker input. The audio precompensation controller has a number L&#x2267;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).

Audio precompensation controller design using a variable set of support loudspeakers

A method for determining an audio precompensation controller for an associated sound generating system comprising a total of N&#x2267;2 loudspeakers, each having a loudspeaker input. The audio precompensation controller has a number L&#x2267;2 inputs for L input signal(s) and N outputs for N controller output signals, one to each loudspeaker. It is relevant to: estimate (S), for each one of at least a subset of the N loudspeaker inputs, an impulse response at each measurement position; specify (S), 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 (S) at least one loudspeaker pair, that is required to be symmetrical with respect to the listing position; and specify (S), for each primary loudspeaker, a target impulse response at each measurement position.

Audio precompensation controller design with pairwise loudspeaker channel similarity

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&#x3e;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.

Spatially robust audio precompensation

The invention concerns digital audio precompensation, and particularly the design of digital precompensation filters. The invention proposes an audio precompensation filter design scheme that uses a novel class of design criteria. Briefly, filter parameters are determined based on a weighting between, on one hand, approximating the precompensation filter to a fixed, non-zero filter component and, on the other hand, approximating the precompensated model response to a reference system response. For design purposes, the precompensation filter is preferably regarded as being additively decomposed into a fixed, non-zero component and an adjustable compensator component. The fixed component is normally configured by the filter designer, whereas the adjustable compensator component is determined by optimizing a criterion function involving the above weighting. The weighting can be made frequency- and/or channel-dependent to provide a very powerful tool for effectively controlling the extent and amount of compensation to be performed in different frequency regions and/or in different channels.

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Knivsta, Sweden

Anders Ahlen