EP2257083B1 - Contrôle de champ sonore dans plusieurs régions d'écoute - Google Patents

Contrôle de champ sonore dans plusieurs régions d'écoute Download PDF

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EP2257083B1
EP2257083B1 EP09007142A EP09007142A EP2257083B1 EP 2257083 B1 EP2257083 B1 EP 2257083B1 EP 09007142 A EP09007142 A EP 09007142A EP 09007142 A EP09007142 A EP 09007142A EP 2257083 B1 EP2257083 B1 EP 2257083B1
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audio
measurement positions
filter
precompensation controller
controller
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EP2257083A1 (fr
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Lars-Johan Brännmark
Mikael Sternad
Mathias Johansson
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Dirac Research AB
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Dirac Research AB
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    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04S—STEREOPHONIC SYSTEMS 
    • H04S7/00—Indicating arrangements; Control arrangements, e.g. balance control
    • H04S7/30—Control circuits for electronic adaptation of the sound field
    • H04S7/301—Automatic calibration of stereophonic sound system, e.g. with test microphone
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04R—LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
    • H04R2499/00—Aspects covered by H04R or H04S not otherwise provided for in their subgroups
    • H04R2499/10—General applications
    • H04R2499/13—Acoustic transducers and sound field adaptation in vehicles

Definitions

  • the present invention generally concerns digital audio precompensation and more particularly the design of a digital audio precompensation controller that generates several signals to a sound generating system, with the aim of modifying the dynamic response of the compensated system, as measured in several spatially separated listening regions.
  • An audio reproduction system is affected by imperfect loudspeaker dynamics and room acoustics.
  • the audio system may furthermore have loudspeakers placed in inappropriate positions.
  • sound material intended for a 5.1 surround system is to be reproduced by loudspeakers in standardized positions, but the number and positioning of loudspeakers in a home or in a car may differ from the specified setting. All of these problems are frequently encountered in home-cinema and audio systems and they are particularly hard to solve for car audio systems with their often awkward loudspeaker positions and difficult acoustic environments.
  • crossover filters are set and each loudspeaker is equalized on a per-channel basis; then the delay and level for each channel is set to reach a desired sound stage (spatial sound perception); additional adjustments to filter responses are made with respect to the combined acoustic loudspeaker responses; finally, parameters for up-mixing are adjusted.
  • Up-mixing here refers to the process of distributing stereo or discrete 5.1 material to the N loudspeakers in the car.
  • the end goal of the tuning process for cars or home hifi/cinema systems can be described in terms of a target sound field in the listening environment
  • the target sound field is in general continuous in space.
  • the solution should not require the loudspeakers to be located in particular positions with respect to the listeners and also not require them to consist of arrays with prescribed spatial properties.
  • the Linear Quadratic Control method for audio precompensation controller design presented in [12] provides means for attaining precise control of the time-domain properties as well as the frequency domain properties of the compensated system.
  • the particular solution presented in [12] is based on a filter structure with a nonzero and fixed parallel path between the inputs and the outputs of the precompensator. This would be an inappropriate structural constraint on a solution to the above stated multichannel design problem; there is here no reason for one virtual source to be assigned to one particular subset of loudspeakers via a fixed part of a precompensation controller.
  • US 2004/0223620 A1 relates to a loudspeaker system for virtual sound synthesis.
  • Setup of the system includes arranging a microphone array adjacent the array of sound sources to obtain a generated sound field.
  • Arbitrary finite impulse response filters are then composed for each sound source within the array of sound sources. Iteration is applied to optimize filter coefficients such that the generated sound field resembles the desired sound field so that multi-channel equalization and wave field synthesis occur.
  • the microphones may be removed.
  • WO 2007/016527 A1 relates to an audio tuning system for automated tuning of an audio system to optimize the sound output of the loudspeakers within the listening space. There may be several listening positions, and a number of audio sensors may be used at each listening position.
  • the present invention is based on the recognition that mathematical models of dynamic systems, and model-based optimization of digital precompensation filters, provide powerful tools for designing filters that improve the performance of various types of audio equipment by modifying the input signals to the equipment. It is furthermore based on the recognition that appropriate models can be obtained by measurements at a discrete grid of M listening positions, with a plurality of listening positions located in each of the P listening regions.
  • a basic idea is to determine an audio precompensation controller for an associated sound generating system.
  • the sound generating system comprises a limited number N ⁇ 2 of loudspeaker inputs for emulating a number L ⁇ 1 of virtual sound sources each of which has an available input signal.
  • the audio precompensation controller has the L input signals to the virtual sound sources as inputs and produces N signals as outputs. These precompensation controller output signals are used as input signals to the sound generating system.
  • the novel scheme for designing or determining the audio precompensation controller is based on:
  • the different aspects of the invention include a method, system and computer program for determining an audio precompensation controller, a so determined precompensation controller, an audio system incorporating such an audio precompensation controller as well as a digital audio signal generated by such an audio precompensation controller.
  • the present invention is based on the recognition that mathematical models of dynamic systems, and model-based optimization of digital precompensation filters, provide powerful tools for designing filters that improve the performance of various types of audio equipment by modifying the input signals to the equipment. It is furthermore based on the recognition that appropriate models can be obtained by measurements at a discrete grid of M listening positions, with a plurality of listening positions located in each of the P listening regions.
  • a first key insight is that a solution can be regarded as acceptable for practical applications if we alleviate the requirement on perfect reconstruction of the target sound field and further limit our target to cover only a finite number of measurement positions.
  • By sampling the sound field at a limited number M of positions in the listening area positions that with adequate resolution cover all relevant listener positions, we discretize the problem and can work directly with N x M transfer functions.
  • a second key observation is that such a set of measurement positions needs to cover several disjoint volumes in space, centered on head positions at several intended listening positions. Concentrating the design accuracy on these spatial volumes, instead of targeting the whole room volume, improves the possibility of obtaining a good result with a limited number of loudspeakers.
  • FIG. 1 is a schematic flow diagram illustrating a method for determining an audio controller according to an exemplary embodiment.
  • Step S1 involves estimating, for each of said N loudspeaker input signals, an impulse response at each of a plurality M of measurement positions in a listening environment based on sound measurements at said M measurement positions.
  • the M measurement positions are distributed in at least two spatially disjoint listening regions, where each listening region has at least four measurement positions.
  • the listening regions correspond to different human listening positions and the distance between regions is larger than the largest distance between adjacent measurement positions within any region.
  • Step S2 involves specifying a target impulse response for each of the L virtual sound sources at each of the M measurement positions in the spatially disjoint regions.
  • Step S3 involves determining adjustable filter parameters of the audio precompensation controller so that a criterion function is optimized under the constraint of stability of the dynamics of the audio precompensation controller.
  • the criterion function preferably includes a weighted summation of powers of differences between the compensated estimated impulse responses and the target impulse responses over a discrete grid of the M measurement positions.
  • a basic idea is to base the design on linear dynamic system models that describe the acoustic responses from each of the N loudspeakers to each of the M listening positions that are distributed among the P listening regions.
  • a target impulse response is also specified for each of the L virtual sound sources as perceived in each of the M listening positions.
  • the audio system controller is based on a linear dynamic precompensation filter that has the L virtual sound source input signals as inputs and produces input signals to the N audio channels of the sound reproduction system.
  • the precompensation controller is adjusted with the aim of letting the series connection of compensator and system models approximate the target impulse responses. This is accomplished by adjusting the free parameters in the precompensation filter so that a criterion is optimized. This criterion is typically defined by a sum over all M listening positions of possibly frequency weighted powers of approximation errors.
  • the optimal precompensation controller can be calculated by performing a Linear Quadratic Gaussian (LQG) optimization of the parameters of a stable, linear and causal multivariable feedforward servo filter, provided that a multivariable stochastic dynamic model is available that describes the assumed second order properties of the virtual sound sources.
  • LQG Linear Quadratic Gaussian
  • the magnitude response of the resulting compensated system is equalized. This to some extent compensates for approximation errors in the previous step and also compensates undesired spectral coloring that may have been introduced in the target stage design.
  • the result is one scalar equalizer filter for each of the virtual sound sources. These filters are placed in the signal chain before the precompensator.
  • the design of the precompensation controller - a set of virtual loudspeakers in a virtual room is created which aims at replacing the physical loudspeakers and room acoustics. Each virtual loudspeaker may then be tuned inside the virtual room to a desired tonal characteristic.
  • the target stages may include parameters that are adjustable within prescribed limits. If so, the design can be iterated between adjustment of target stage parameters and adjustment of precompensator parameters, with the aim to attain an improved approximation between target stage and precompensated audio system, and thus an improved criterion value. Focusing the approximation accuracy on disjoint listening regions and allowing some variability in the target stages are both means for relaxing unnecessary constraints on the problem and thus attaining better approximation solutions.
  • the resulting precompensation filter may have elements with long impulse responses. If the computational complexity needs to be reduced, then it is proposed that scalar elements of the precompensation filter matrix are approximated by implementing these filters as a parallel connection of a finite impulse response (FIR) filter that corresponds to the initial part of the filter impulse response and an infinite impulse response (IIR) recursive filter that approximates the tail of the filter impulse response.
  • FIR finite impulse response
  • IIR infinite impulse response
  • Section 1 provides a brief overview of the structure of an exemplary digital sound pre-compensation system.
  • Section 2 then describes an example of the modelling and the target stage definition, while Section 3 defines an example of a particular optimization problem to be solved.
  • Section 4 presents an exemplary design of a precompensation controller based on Linear Quadratic Gaussian (LQG) optimal feedforward control.
  • Section 5 provides an exemplary technique to reduce the complexity of the resulting set of filters used by the precompensator and Section 6 discusses further implementation aspects of the design and the resulting audio precompensation system.
  • LQG Linear Quadratic Gaussian
  • Transfer function matrices Linear filters, dynamic systems or models that may have multiple inputs and/or multiple outputs are represented by transfer function matrices in the following and are denoted by boldface letters. Transfer function matrices that include only FIR filters as elements will be denoted polynomial matrices and are denoted by italic capitals.
  • the operator H represents a model of the acoustic impulse response, represented by a transfer function matrix.
  • the transfer function matrix H represents the effect of the whole or a part of the sound generating or sound reproducing system, including any pre-existing digital compensators, digital-to-analog converters, analog amplifiers, loudspeakers, cables and the room acoustic response. In other words, the transfer function matrix H represents the dynamic response of relevant parts of a sound generating system.
  • the input signal u(t) to this system which is a N -dimensional column vector, may represent input signals to N individual amplifier-loudspeaker chains of the sound generating system.
  • the signal y m (t) (with subscript m denoting measurement) is a M -dimensional column vector representing the true (measured) sound time-series at the M measurement locations and e(t) represents noise, unmodelled room reflexes, effects of an incorrect model structure, nonlinear distortion and other unmodelled contributions.
  • the objective is to modify the dynamics of the sound generating system represented by (1.1) in relation to a reference dynamics.
  • the elements of the vector w(t) may, for example, represent channels of digitally recorded sound, or analog sources that have been sampled and digifized.
  • D is a stable transfer function matrix of dimension MxL that is assumed to be known. This linear discrete-time dynamic system is to be specified by the designer.
  • Its desired effect at the M measurement positions is represented by column i of the transfer function matrix D in (1.2).
  • the desired responses for different listening regions are represented by filters in disjoint sets of rows of D .
  • the system D may include a set of adjustable parameters. Alternatively, it may indirectly be affected by such a set via its specification.
  • This audio precompensation controller includes a set of adjustable parameters. These parameters should allow sufficient flexibility to modify its input-output dynamic properties, for example allowing some elements of R or the whole of R to be zero for appropriate parameter settings. The optimization of R should however be constrained to parameter settings that make R an input-output stable dynamic system.
  • the attainable approximation quality depends on the nature of the problem set-up. For a fixed given acoustic environment, the quality of the approximation can in general be improved if the number of loudspeaker channels N is increased. It can likewise be improved by increasing the number M of measurement points within fixed listening regions, since this gives a denser sampling of the sound field. Enlargement of the listening regions or addition of regions for a fixed N would, in general, result in larger approximation errors. Adding more sound stages (increasing L ) would result in the need for proportionally more compensation filters, but it would not decrease the attainable approximation accuracy for previously designed sound stages if other basic parameters were kept constants. If the elements of w(t) are assumed uncorrelated then by linearity, the optimal precompensation filters presented in Section 4 below can be computed separately for different sound stages and their individual contributions to the total approximation error will be additive.
  • Linear discrete-time dynamic systems are in the following represented using the discrete-time backward shift operator here denoted by q -1 .
  • a signal vector s(t) is shifted backward by one sample by this operator.
  • q -1 s(t) s(t-1) .
  • the backward shift operator corresponds to the complex variable z -1 or e -j ⁇ in the discrete-time frequency domain.
  • a causal matrix of FIR filters (polynomial matrix) A ( q -1 ) operates only on input signals that are current or past with respect to the present time index t . It will thus have matrix elements that are polynomials in the backward shift operator q -1 only.
  • the room-acoustic impulse responses of each loudspeaker at each listener position are estimated from measurements at M positions, which are partitioned into several spatially separated listening areas. It is recommended that at least four measurement positions are used within each listening area, to obtain adequate fidelity within extended spatial volumes, since listeners are expected to move their heads within prescribed areas.
  • the measurement positions within a listening area can, for example, be located in a plane or be distributed within a 3D volume.
  • the dynamic acoustic responses can then be estimated by sending out test signals from the loudspeakers, one loudspeaker at a time, and recording the resulting acoustic signals at all M listening positions. White or colored noise may be used as test signals for this purpose.
  • Models of the linear dynamic responses from one loudspeaker to M outputs can then be estimated in the form of FIR or IIR filters with one input and M outputs.
  • Various system identification techniques such as the least squares method or spectral analysis-based techniques can be used for this purpose.
  • the measurement procedure is repeated for all loudspeakers, finally resulting in a model H that is represented by a M x N matrix of dynamic models.
  • the multi input - multi output model may alternatively be represented by a state space description.
  • M 64 measurement positions are used.
  • P 4 separate listener regions at head heights each centered at a car seat (front left, front right, rear left, rear right).
  • a quadratic horizontal grid of 4 x 4 measurement positions is employed, resulting in two sets of measurements for each input channel (loudspeaker).
  • the resulting set of M x N measurements can be used to estimate the set of M x N impulse responses that define the model in (1.1) and (2.1).
  • a target stage is composed of M desired impulse responses (or equivalently, transfer functions) that are preferably nonzero, one for each measurement position.
  • One target stage is defined for each of L virtual sources that are to be created and it is represented by a column of the matrix D in (1.2). For example, in the case of reproducing stereo material via two virtual loudspeakers, the vector w(t) would have two elements and two target stages would be defined.
  • the target stages can be measured inside a reference listening room using the same technique as when modeling the acoustic impulse response, or the target stages can be simulated.
  • the target stages may be defined so that all the P listening areas are located in a "sweet-spot" of the virtual listening environment.
  • the target stages are obtained by computing acoustic impulse responses from a simulated acoustic environment, then some controlled variability can be introduces into the target stages.
  • the angles and distances of the virtual loudspeakers, the size of the room and properties such as strength and diffuseness of first reflexes can be left adjustable within prescribed limits.
  • Such flexibility of the target can help attain better approximation to the selected targets, better criterion values and better perceived audio quality. This type of flexibility can be utilized by adjusting the parameters of the stage D and the parameters of the precompensation filter R iteratively:
  • a precompensator is first optimized for an initial set of target stage parameters.
  • the target stage parameters are then adjusted within prescribed admissible limits, a new stage D is defined and the precompensator is optimized again for the new target stage parameters.
  • the resulting criterion value is then evaluated. This procedure is repeated until no improvement of the criterion value can be found.
  • the search of the target stage parameter space can be performed by a search routine such as a gradient-based or a conjugated gradient optimization method, by the Simplex method or by genetic algorithms. If the number of adjustable stage parameters is not too large, an exhaustive search of grid points for a discrete grid of target stage parameter values is feasible.
  • the vector ⁇ (t) of errors at the measurement positions is related to the vector w ( t ) via (1.5).
  • the expectation E ( ) in (3.1) is to be taken with respect to the statistical properties of the signal w(t) , and any other parts of the model structure that are described statistically.
  • the expression in (3.1c) represents the squared 2-norm of a random process.
  • the first right-hand sum of this criterion represents a weighted summation over the M measurement positions of powers of differences between the compensated estimated impulse responses represented by elements of HR and the target impulse responses represented by elements of D , where the weighting is performed by the polynomial matrix V ( q - 1 ) and by the spectral properties of the signal w(t) .
  • the square diagonal polynomial matrix W ( q -1 ) diag[ W ⁇ ( q -1 )] can, for example, to be used to focus the control energy into frequency ranges that are appropriate for particular loudspeaker inputs. Each penalty FIR filter is then given low gain within the operating range of the loudspeaker and high gain outside of that range.
  • the criterion (3.1) or other forms of quadratic criteria could be optimized by various means.
  • the arbitrary introduction of structural constraints would always limit the performance.
  • the optimization should preferably be performed without structural constraints on the precompensation matrix, except for the necessary constraints of causality and stability of its dynamics.
  • the precompensation controller design problem then becomes a Linear-Quadratic Gaussian design problem for a multivariable feedforward control element R .
  • Linear quadratic theory provides optimal linear controllers for linear systems and quadratic criteria [13],[14]. If the problem formulation is such that signals are assumed to have Gaussian statistics, then this solution can be shown to be optimal also within the class of all (linear as well as nonlinear) controllers. The optimization is performed under the constraint of causality of the controller and stability of the controlled system. In the feedforward control setting discussed here, with the systems H and D assumed stable, stability of the controlled system D - HR is equivalent to stability of the controller R .
  • the polynomial matrix H ( q -1 ) has dimension L x L .
  • the optimization of the criterion (3.1) is thus performed by first solving the quadratic polynomial matrix right spectral factorization equation (4.4) to obtain the polynomial matrix ⁇ ( q -1 ) and then solving the Diophantine equation (4.6) to obtain the polynomial matrix Q ( q -1 ).
  • the regulator (4.5) is then represented by a structure which could be realized as a series connection of three multivariable filters as follows.
  • This last step inverts the autoregressive dynamics of the model (4.1) that is represented by the factor A -1 (q -1 ) on the input side of equation (4.1).
  • the controller (4.5) is in the form of a recursive infinite impulse response filter with multiple inputs and multiple outputs. An approximation of the matrix elements of this controller, that uses a set of scalar filters of lower orders, is discussed in Section 5 below.
  • An optimization of the precompensation controller as exemplified here is designed to jointly perform equalization of the original room acoustics and loudspeaker dynamics, crossover filter design and delay and level calibration, sum response optimization and up-mixing of L sources to N loudspeaker inputs to approximate the prescribed sound field response (4.2) according to the criterion (3.1).
  • the prescribed sound field may itself have introduced some undesired spectral features.
  • An optional post-processing step can be used to handle such remaining issues.
  • the target stage is specified by using a simulator that creates plane wave impulse responses. If the target stage consists of only direct sound, then the resulting target frequency response is flat. If in addition to the direct wave the target stage also includes reflections, spectral coloration will arise. Moreover, the designed controller matrix R inevitably will have remaining approximation errors since the number of measurement positions is typically much larger than the number of loudspeakers. These approximation errors may have different magnitude at different frequencies. Magnitude response imperfections are generally undesirable and the controller matrix should preferably be adjusted so that an overall target magnitude response is reached on average in all the listening regions.
  • a final design step is therefore preferably added after the criterion minimization with the aim of adjusting the controller response so that, on average, a target average magnitude response for each virtual source is well approximated in all the listening regions.
  • the magnitude responses of the overall system are evaluated in the various listening positions, based on the design models or based on new measurements.
  • a minimum phase filter is then designed so that on average (in the RMS sense) the target magnitude response is reached in all listening regions.
  • variable fractional octave smoothing based on the spatial response variations may be employed in order not to overcompensate in any particular frequency region.
  • the result is one scalar equalizer filter for each of the virtual sound sources. These filters are placed in the signal chain between the elements of w(t) and the inputs to the precompensator that was designed in the previous step.
  • the performance of the proposed sound control technique is illustrated by measured results obtained in a car equipped with one tweeter in the center of the dashboard, four mid-range + tweeter pairs in the front and rear doors, four low-range woofers (working range roughly 15-5000 Hz) in the front and rear doors, and a pair of subwoofer speakers (working range roughly 15-300 Hz) in the rear shelf.
  • the subwoofers are driven by the same signal source and are thus treated as one single subwoofer.
  • N 10 loudspeaker input channels are used in this setting. This is a rather representative premium car sound system.
  • a model was estimated for 16 measurement positions at each seat, for horizontal square listening regions of dimension 30 x 30 cm with 10 cm distance between measurement points, as illustrated by the front seat part of Fig. 3 .
  • a model with M x N 32 x 10 individual impulse responses was then estimated.
  • precompensation controllers were calculated and their performance was evaluated on the estimated model. All results were obtained using 10 FIR filters per single target stage, each of length 10 000 coefficients using 44.1 kHz sampling.
  • Fig. 4 shows the measured delays at each measurement position for a virtual source placed at 90 degrees to the right relative to the front direction. None of the physical loudspeakers is positioned in this direction. Only minor errors occur, the delay surfaces show a clear tilt towards the intended direction. Possible sources of error include delay estimation errors and inadequate sound field reconstruction. On the whole, the algorithm reproduces sound waves from the desired directions in both front seat listening regions.
  • Fig. 5 shows the resulting power responses from 16 different positions in the left front seat and 16 positions at the right front seat respectively, for a virtual left front speaker, located at -35 degrees in the plane of the microphones and with a flat target magnitude response.
  • the red solid curves show magnitudes of the averages over 16 measurement positions of the complex gains of the compensated models at those positions, while the dotted blue curves show individual magnitude responses.
  • the algorithm has evidently evened out the average spectral responses over space by a proper combined use of the 10 loudspeakers. For individual loudspeakers, there are significant differences in the uncompensated measured responses at the different listening positions. In the precompensated model shown in Fig 5 , the magnitudes in the individual listening positions are very close at frequencies up to 300 Hz.
  • the distance between the microphones (10 cm) becomes on the order of or larger than the wavelength.
  • An exact control of the received phase at each position in the higher frequency regions is therefore not possible, and it is notably not necessary from a psychoacoustic perspective.
  • the average response is of most importance for human perception at higher frequencies, and the averages over 16 positions follow the flat target response within 5 dB over the whole audible frequency range.
  • the resulting matrix filter R by (4.x) can be realized in any number of ways, in state space form or in transfer function form.
  • the required filters are in general of very high order, in particular if a full audio range sampling rate is used and if also room acoustic dynamics needs to be taken into account.
  • methods for limiting the computational complexity of the precompensator are of interest.
  • the relevant scalar impulse response elements R ij ( q - 1 ) of the pre-compensator R are first represented as very long FIR filters. For each precompensator impulse response R i j ( q -1 ),
  • the aim of this procedure is to obtain realizations in which the sum of the number of parameters in the FIR filter M ( q - 1 ) and the IIR filter N( q - 1 ) is much lower than the original number of impulse response coefficients.
  • Various different methods for approximating the tail of the impulse response can be used, for example adjustment of autoregressive models to a covariance sequence based on the Yule-Walker equations.
  • first order filters or second order IIR filter elements may be used.
  • the design equations are solved on a separate computer system to produce the filter parameters of the precompensation filter.
  • the calculated filter parameters are then normally downloaded to a digital filter, for example realized by a digital signal processing system or similar computer system, which executes the actual filtering.
  • the filter design scheme proposed by the invention is preferably implemented as software in the form of program modules, functions or equivalent.
  • the software may be written in any type of computer language, such as C, C++ or even specialized languages for digital signal processors (DSPs).
  • DSPs digital signal processors
  • the computer program used for the design of the audio precompensation filter is normally encoded on a computer-readable medium such as a DVD, CD or similar structure for distribution to the user/filter designer, who then may load the program into his/her computer system for subsequent execution.
  • the software may even be downloaded from a remote server via the Internet.
  • Fig. 6 is a schematic block diagram illustrating an example of a computer system suitable for implementation of a filter design algorithm according to the invention.
  • the system 100 may be realized in the form of any conventional computer system, including personal computers (PCs), mainframe computers, multiprocessor systems, network PCs, digital signal processors (DSPs), and the like.
  • the system 100 basically comprises a central processing unit (CPU) or digital signal processor (DSP) core 10, a system memory 20 and a system bus 30 that interconnects the various system components.
  • the system memory 20 typically includes a read only memory (ROM) 22 and a random access memory (RAM) 24.
  • ROM read only memory
  • RAM random access memory
  • the system 100 normally comprises one or more driver-controlled peripheral memory devices 40, such as hard disks, magnetic disks, optical disks, floppy disks, digital video disks or memory cards, providing non-volatile storage of data and program information.
  • Each peripheral memory device 40 is normally associated with a memory drive for controlling the memory device as well as a drive interface (not illustrated) for connecting the memory device 40 to the system bus 30.
  • a filter design program implementing a design algorithm according to the invention may be stored in the peripheral memory 40 and loaded into the RAM 22 of the system memory 20 for execution by the CPU 10. Given the relevant input data, such as a model representation and other optional configurations, the filter design program calculates the filter parameters of the precompensation filter.
  • the determined filter parameters are then normally transferred from the RAM 24 in the system memory 20 via an I/O interface 70 of the system 100 to a precompensation filter system 200.
  • the precompensation filter system 200 is based on a digital signal processor (DSP) or similar central processing unit (CPU) 202, and one or more memory modules 204 for holding the filter parameters and the required delayed signal samples.
  • DSP digital signal processor
  • CPU central processing unit
  • the memory 204 normally also includes a filtering program, which when executed by the processor 202, performs the actual filtering based on the filter parameters.
  • the filter parameters may be stored on a peripheral memory card or memory disk 40 for later distribution to a precompensation filter system, which may or may not be remotely located from the filter design system 100.
  • the calculated filter parameters may also be downloaded from a remote location, e.g. via the Internet, and then preferably in encrypted form.
  • any conventional microphone unit(s) or similar recording equipment 80 may be connected to the computer system 100, typically via an analog-to-digital (A/D) converter 80.
  • A/D analog-to-digital
  • the system 100 can develop a model of the audio system, using an application program loaded into the system memory 20. The measurements may also be used to evaluate the performance of the combined system of precompensation filter and audio equipment If the designer is not satisfied with the resulting design, he may initiate a new optimization of the precompensation filter based on a modified set of design parameters.
  • system 100 typically has a user interface 50 for allowing user interaction with the filter designer. Several different user-interaction scenarios are possible.
  • the filter designer may decide that he/she wants to use a specific, customized set of design parameters in the calculation of the filter parameters of the filter system 200.
  • the filter designer then defines the relevant design parameters via the user interface 50.
  • the filter designer can select between a set of different pre-configured parameters, which may have been designed for different audio systems, listening environments and/or for the purpose of introducing special characteristics into the resulting sound.
  • the preconfigured options are normally stored in the peripheral memory 40 and loaded into the system memory during execution of the filter design program.
  • the filter designer may also define the reference system by using the user interface 50.
  • the bulk delay d of the reference system may be selected by the user, or provided as a default delay.
  • the filter designer may select a model of the audio system from a set of different preconfigured system models. Preferably, such a selection is based on the particular audio equipment with which the resulting precompensation filter is to be used.
  • the audio filter is embodied together with the sound generating system so as to enable generation of sound influenced by the filter.
  • the filter design is performed more or less autonomously with no or only marginal user participation.
  • the exemplary system comprises a supervisory program, system identification software and filter design software.
  • the supervisory program first generates test signals and measures the resulting acoustic response of the audio system. Based on the test signals and the obtained measurements, the system identification software determines a model of the audio system. The supervisory program then gathers and/or generates the required design parameters and forwards these design parameters to the filter design program, which calculates the precompensation filter parameters.
  • the supervisory program may then, as an option, evaluate the performance of the resulting design on the measured signal and, if necessary, order the filter design program to determine a new set of filter parameters based on a modified set of design parameters. This procedure may be repeated until a satisfactory result is obtained. Then, the final set of filter parameters are downloaded/implemented into the precompensation filter system.
  • the filter parameters of the precompensation filter may change.
  • the position of the loudspeakers and/or objects such as furniture in the listening environment may change, which in turn may affect the room acoustics, and/or some equipment in the audio system may be exchanged by some other equipment leading to different characteristics of the overall audio system.
  • continuous or intermittent measurements of the sound from the audio system in one or several positions in the listening environment may be performed by one or more microphone units or similar sound recording equipment
  • the recorded sound data may then be fed into a filter design system, such as system 100 of Fig. 6 , which calculates a new audio system model and adjusts the filter parameters so that they are better adapted for the new audio conditions.
  • the invention is not limited to the arrangement of Fig. 6 .
  • the design of the precompensation filter and the actual implementation of the filter may both be performed in one and the same computer system 100 or 200. This generally means that the filter design program and the filtering program are implemented and executed on the same DSP or microprocessor system.
  • a sound generating or reproducing system 300 incorporating a precompensation filter system 200 according to the present invention is schematically illustrated in Fig. 7 .
  • a vector w(t) of audio signals from a sound source is forwarded to a precompensation filter system 200, possibly via a conventional I/O interface 210.
  • the audio signals w(t) are analog, such as for LPs, analog audio cassette tapes and other analog sound sources, the signal is first digitized in an A/D converter 210 before entering the filter 200.
  • Digital audio signals from e.g. CDs, DAT tapes, DVDs, mini discs, and so forth may be forwarded directly to the filter 200 without any conversion.
  • the digital or digitized input signal w(t) is then precompensated by the precompensation filter 200, basically to take the effects of the subsequent audio system equipment into account.
  • the resulting compensated signal u(t) is then forwarded, possibly through a further I/O unit 230, for example via a wireless link, to a D/A-converter 240, in which the digital compensated signal u(t) is converted to a corresponding analog signal.
  • This analog signal then enters an amplifier 250 and a loudspeaker 260.
  • the sound signal y m (t) emanating from the set of N loudspeaker 260 then has the desired audio characteristics, giving a close to ideal sound experience. This means that any unwanted effects of the audio system equipment have been eliminated through the inverting action of the precompensation filter.
  • the precompensation filter system may be realized as a standalone equipment in a digital signal processor or computer that has an analog or digital interface to the subsequent amplifiers, as mentioned above. Alternatively, it may be integrated into the construction of a digital preamplifier, a computer sound card, a compact stereo system, a home cinema system, a computer game console, a TV, an MP3 player docking station or any other device or system aimed at producing sound. It is also possible to realize the precompensation filter in a more hardware-oriented manner, with customized computational hardware structures, such as FPGAs or ASICs.
  • the precompensation may be performed separate from the distribution of the sound signal to the actual place of reproduction.
  • the precompensation signal generated by the precompensation filter does not necessarily have to be distributed immediately to and in direct connection with the sound generating system, but may be recorded on a separate medium for later distribution to the sound generating system.
  • the compensation signal u(t) in Fig, 1 could then represent for example recorded music on a CD or DVD disk that has been adjusted to a particular audio equipment and listening environment It can also be a precompensated audio file stored on an Internet server for allowing subsequent downloading of the file to a remote location over the Internet.

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Claims (18)

  1. Procédé de détermination d'un contrôleur de précompensation audio pour un système de génération de son associé, ledit contrôleur de précompensation audio ayant un nombre L ≥ 1 de signaux d'entrée à L sources sonores virtuelles en tant qu'entrées et produisant un nombre limité N ≥ 2 de signaux en tant que sorties à N entrées de haut-parleur du système de génération de son pour émuler lesdites L sources sonores virtuelles, ledit contrôleur de précompensation audio ayant un nombre de paramètres de filtre ajustables et la propriété de produire une sortie nulle pour un certain ajustement de ses paramètres de filtre ajustables, ledit procédé comprenant les étapes consistant à :
    - estimer, pour chacun desdits N signaux d'entrée de haut-parleur, une réponse d'impulsion à chacune d'une pluralité M de positions de mesure agencées dans une grille discrète dans un environnement d'écoute en se basant sur des mesures de son au niveau desdites M positions de mesure, où lesdites M positions de mesure sont distribuées en au moins deux régions d'écoute spatialement disjointes, chaque région d'écoute ayant au moins quatre positions de mesure, où lesdites régions d'écoute correspondent à des positions d'écoute humaine différentes et la distance entre des régions est plus grande que la plus grande distance entre des positions de mesure adjacentes au sein d'une région quelconque ;
    - spécifier une réponse d'impulsion cible pour chacune desdites L sources sonores virtuelles au niveau de chacune desdites M positions de mesure dans lesdites régions spatialement disjointes ;
    - déterminer des paramètres de filtre ajustables dudit contrôleur de précompensation audio de sorte qu'une fonction de critère est optimisée sous la contrainte de stabilité de la dynamique du contrôleur de précompensation audio, ladite fonction de critère incluant une sommation pondérée de puissances de différence entre les réponses d'impulsion estimées compensées et les réponses d'impulsion cibles sur ladite grille discrète desdites M positions de mesure, ou ladite étape de détermination des paramètres de filtre dudit contrôleur de précompensation audio est basée sur une optimisation gaussienne quadratique linéaire (GQL) des paramètres d'un servofiltre à action directe multivariable, stable, linéaire et causal basé sur un système dynamique cible donné, un modèle dynamique du système de génération de son et sur des modèles dynamiques stochastiques multivariables qui décrivent des statistiques de second ordre des sources sonores virtuelles.
  2. Procédé selon la revendication 1, dans lequel un jeu de N filtres audio est déterminé pour chacun d'un jeu de L signaux de source sonore, et ledit contrôleur audio comprend N x L filtres de précompensation scalaires linéaires dynamiques discrets dans le temps avec des paramètres ajustables qui ont chacun des L signaux d'entrée aux sources sonores virtuelles comme entrées, et l'une des N entrées sonores aux hauts-parleurs comme sorties.
  3. Procédé selon la revendication 1, dans lequel la distance entre les régions d'écoute est au moins deux fois plus grande que la plus grande distance entre des positions de mesure adjacente dans une région quelconque.
  4. Procédé selon la revendication 1, dans lequel ladite étape de détermination de paramètres de filtre du contrôleur de précompensation audio est également basée sur l'ajustement de paramètres de filtre dudit contrôleur de précompensation audio pour atteindre une réponse de grandeur cible du système de génération de son incluant le contrôleur audio dans au moins un sous-jeu desdites M positions de mesure.
  5. Procédé selon la revendication 4, dans lequel ladite étape d'ajustement de paramètres de filtre dudit contrôleur de précompensation audio est basée sur l'évaluation de réponses de grandeur, puis une détermination d'un modèle de filtre de phase minimal du système de génération de son incluant le contrôleur audio dans au moins un sous-jeu desdites M positions de mesure.
  6. Procédé selon la revendication 1, dans lequel les réponses d'impulsion cibles sont non nulles et incluent des paramètres ajustables qui peuvent être modifiés dans des limites prescrites.
  7. Procédé selon la revendication 6, dans lequel les paramètres ajustables des réponses d'impulsion cibles ainsi, que les paramètres ajustables du contrôleur de précompensation audio sont ajustés conjointement, dans le but d'optimiser la fonction de critère.
  8. Procédé selon la revendication 1, dans lequel ladite étape d'estimation, pour chacun desdits N haut-parleurs, une réponse d'impulsion au niveau de chacune d'une pluralité M de positions de mesures est basée sur un modèle décrivant la réponse dynamique du système de génération de son associé au niveau desdites M positions de mesure, pour lesquelles ladite réponse dynamique diffère d'au moins deux de ces positions de mesure.
  9. Procédé selon la revendication 8, dans lequel ledit modèle est déterminé en se basant sur des mesures de son au niveau de M positions de mesure, ledit son étant produit par ledit système de génération de son, et ladite étape de détermination dudit jeu de N filtres audio comprend l'étape de détermination de paramètres de filtre correspondants, et ledit contrôleur de précompensation audio est créé en implémentant les paramètres de filtre déterminés dans une structure de filtre audio.
  10. Procédé selon la revendication 9, dans lequel ladite structure de filtre audio est intégrée conjointement avec ledit système de génération de son associé de façon à permettre la génération d'un champ sonore cible souhaité au niveau desdites M positions de mesure dans ledit environnement d'écoute.
  11. Procédé selon la revendication 1, dans lequel ledit système de génération de son est un système audio pour voiture, et ledit environnement d'écoute est une partie d'une voiture.
  12. Système de détermination d'un contrôleur de précompensation audio pour un système de génération de son associé, ledit contrôleur de précompensation audio ayant des entrées permettant de recevoir un nombre L ≥ 1 de signaux d'entrée à L sources sonores virtuelles et produisant un nombre limité N ≥ 2 de signaux en tant que sorties à N entrées de haut-parleur du système de génération de son pour émuler lesdites L sources sonores virtuelles, ledit contrôleur de précompensation audio ayant un nombre de paramètres de filtre ajustables et la propriété de produire une sortie nulle pour un certain ajustement de ses paramètres de filtre ajustables, ledit système comprenant :
    - un moyen d'estimation, pour chacun desdits N signaux d'entrée de haut-parleur, une réponse d'impulsion à chacune d'une pluralité M de positions de mesure agencées dans une grille discrète dans un environnement d'écoute en se basant sur des mesures de son au niveau desdites M positions de mesure, où lesdites M positions de mesure sont distribuées en au moins deux régions spatialement disjointes, chaque région ayant au moins quatre positions de mesure, où lesdites régions d'écoute correspondent à des positions d'écoute humaine différentes et la distance entre des régions est plus grande que la plus grande distance entre des positions de mesure adjacentes au sein d'une région quelconque
    - un moyen permettant de spécifier une réponse d'impulsion cible pour chacune desdites L sources sonores virtuelles au niveau de chacune desdites M positions de mesure dans lesdites régions spatialement disjointes ;
    - un moyen permettant de déterminer des paramètres de filtre ajustables dudit contrôleur de précompensation audio de sorte qu'une fonction de critère est optimisée sous la contrainte de stabilité de la dynamique du contrôleur de précompensation audio, ladite fonction de critère incluant une sommation pondérée de puissances de différence entre les réponses d'impulsion estimées compensées et les réponses d'impulsion cibles sur ladite grille discrète desdites M positions de mesure, où ledit moyen permettant de déterminer les paramètres de filtre dudit contrôleur de précompensation audio est configuré pour fonctionner sur la base d'une optimisation gaussienne quadratique linéaire (GQL) des paramètres d'un servofiltre à action directe multivariable, stable, linéaire et causal basé sur un système dynamique cible donné, un modèle dynamique du système de génération de son et sur des modèles dynamiques stochastiques multivariables qui décrivent des statistiques de second ordre des sources sonores virtuelles.
  13. Produit de programme informatique permettant de déterminer, une fois exécuté sur un système d'ordinateur, un contrôleur de précompensation audio pour un système de génération de son associé, ledit contrôleur de précompensation audio ayant un nombre L ≥ 1 de signaux d'entrée à L sources sonores virtuelles en tant qu'entrées et produisant un nombre limité N ≥ 2 de signaux en tant que sorties à N entrées de haut-parleur du système de génération de son pour émuler lesdites L sources sonores virtuelles, ledit contrôleur de précompensation audio ayant un nombre de paramètres de filtre ajustables et la propriété de produire une sortie nulle pour un certain ajustement de ses paramètres de filtre ajustables, ledit produit de programme informatique comprenant :
    - un moyen de programme permettant d'estimer, pour chacun desdits N signaux d'entrée de haut-parleur, une réponse d'impulsion à chacune d'une pluralité M de positions de mesure agencées dans une grille discrète dans un environnement d'écoute en se basant sur des mesures de son au niveau desdites M positions de mesure, où lesdites M positions de mesure sont distribuées en au moins deux régions spatialement disjointes, chaque région ayant au moins quatre positions de mesure, où lesdites régions d'écoute correspondent à des positions d'écoute humaine différentes et la distance entre des régions est plus grande que la plus grande distance entre des positions de mesure adjacentes au sein d'une région quelconque ;
    - un moyen de programme permettant de spécifier une réponse d'impulsion cible pour chacune desdites L sources sonores virtuelles au niveau de chacune desdites M positions de mesure dans lesdites régions spatialement disjointes ;
    - un moyen de programme permettant de déterminer des paramètres de filtre ajustables dudit contrôleur de précompensation audio de sorte qu'une fonction de critère est optimisée sous la contrainte de stabilité de la dynamique du contrôleur de précompensation audio, ladite fonction de critère incluant une sommation pondérée de puissances de différence entre les réponses d'impulsion estimées compensées et les réponses d'impulsion cibles sur ladite grille discrète desdites M positions de mesure, où ledit moyen de programme permettant de déterminer des paramètres de filtre dudit contrôleur de précompensation audio est configurée pour fonctionner sur la base d'une optimisation gaussienne quadratique linéaire (GQL) des paramètres d'un servofiltre à action directe multivariable, stable, linéaire et causal basé sur un système dynamique cible donné, un modèle dynamique du système de génération de son et sur des modèles dynamiques stochastiques multivariables qui décrivent des statistiques de second ordre des sources sonores virtuelles.
  14. Contrôleur de précompensation audio déterminé en utilisant le procédé selon la revendication 1.
  15. Contrôleur de précompensation audio déterminé en utilisant le procédé selon la revendication 2, pour lequel certains des filtres scalaires qui sont des éléments matriciels du contrôleur de précompensation audio sont réalisés sous forme de connexions parallèles entre un filtre à ligne à retard à prises non récursif (FIR) non nul et un filtre récursif (IIR) et où la composante du filtre IIR est ajustée pour être une approximation de la réponse d'impulsion du filtre de précompensation scalaire au sein d'un jeu [t1, t2] de retards de temps, où t1 > 1 et t2 > t1.
  16. Contrôleur de précompensation audio selon la revendication 15, dans lequel le filtre IIR est réalisé sous forme de connexions parallèles de filtres IIR de composante ou sous forme de connexions série de filtres IIR de composante, ou de l'une de leurs combinaisons.
  17. Système audio comprenant un système de génération de son et un contrôleur de précompensation audio dans le chemin d'entrée vers ledit système de génération de son, dans lequel ledit contrôleur de précompensation audio est déterminé en utilisant le procédé selon la revendication 1.
  18. Signal audio numérique généré par un contrôleur de précompensation audio déterminé en utilisant le procédé selon la revendication 1.
EP09007142A 2009-05-28 2009-05-28 Contrôle de champ sonore dans plusieurs régions d'écoute Active EP2257083B1 (fr)

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