Flat panel loudspeaker control and flat panel loudspeaker

Abstract

Flat panel loudspeaker control (100) for controlling a flat panel loudspeaker (101; 300) comprising a plurality of actuators, the flat panel loudspeaker control comprising: a plurality of electrical signal inputs (102), each input being assigned to a corresponding actuator of the planar loudspeaker (101; 300) to be controlled, a plurality of signal processors (104), each signal processor being associated with a respective input and having an output (106) for an electrical signal for controlling an actuator of the planar loudspeaker (101; 300), and each signal processor implementing a transfer function from its input to its output (106) on the basis of each actuator of the planar loudspeaker (101; 300) for a desired sound receiver (sic), and a signal processor controller (108) associated with each of the plurality of signal processors (104), the signal processor controller (108) being preconfigured to improve the phase angle between the signals as an overall output at the outputs (106) of the signal processors (104).

Description

FIELD OF THE INVENTION

[0001] The present invention relates to a flat panel loudspeaker control and a flat panel loudspeaker, such as a resonant plate-shaped loudspeaker. BACKGROUND OF THE INVENTION

[0002] In conventional loudspeakers, a piston motion in the center of a diaphragm is used to vibrate the air, generating sound waves. The outer edge of the diaphragm is supported by a frame, and the driven center of the diaphragm is supported by a damper. The diaphragm is usually conical to provide stiffness in the direction of vibration.

[0003] In contrast, in a flat, plate, or planar loudspeaker, vibrations from actuators are applied to specific points on a flat plate diaphragm to generate bending waves in the diaphragm. Thus, multiple point sound sources are arranged across the entire diaphragm as bending waves, distributed across the diaphragm in random phases over a frequency range. General descriptions of planar or planar loudspeakers can be found in US Patent No. US 6,332,029 B1 and European Patent Application Publication No. EP 0 847 661 B2.

[0004] DM loudspeakers (or DMLs) are flat panel loudspeakers in which sound is generated by inducing evenly distributed vibration modes in the panel. A mode is a predictable standing wave diffraction pattern resulting from stimulating the panel at a single frequency. This mode depends on the physical constraints of the panel and the frequency. DMLs are available in various forms, notably as part of a larger structure with rigid boundaries, as described in US Patent No. US 6 546 106 B2 and European Patent Application Publication No. EP 1 068 770 B1, or as a display element in an electronic device, as described in US Patent No. US 7 174 025 B2 and European Patent Application Publication No. EP 1 084 592 B1.

[0005] While it is common for a DML to be driven by actuators smaller than the plate, this is not necessarily the case. US Patent No. US 6,795,561 B1 and European Patent Application Publication No. EP 1 197 120 B1 describe activation by an electrically active planar actuator similar in size to the plate to be driven.

[0006] There is a demand for thin electronic devices with audio functionality, but many existing DML applications are considered too thick for these use cases. From a technical perspective, large-area, electrically active planar actuators are considered attractive for these use cases. However, these large-area patches are unattractive due to high component costs, low efficiency, and poor acoustic response.

[0007] With the introduction of organic light-emitting diode (OLED) displays, small patches behind the display can be used to provide audio functionality with a display device. These small patches are no longer limited to the localized edge drive of the panels, as was the case with backlit LCDs. Consequently, a method for using a plurality of small patches or patch arrays that are cost-effective and do not excessively stiffen the substrate is needed.

[0008] Each actuator is controlled by an electrical input, and a flat panel loudspeaker controlled by n actuators has n input channels (where n is an integer and n > 1). For example, from Audio Engineering Society Convention Paper 5611, "Multichannel Inverse Filtering of Multiexciter Distributed Mode Loudspeakers for Wave Field Synthesis," by Etienne Corteel, Ulrich Horbach, and Renato S. Pellegrini, presented at the 112th Congress held in Munich from May 10-13, 2013, it is known to calibrate the response of a flat panel loudspeaker with n channels by applying an individual impulse to each input and observing the impulse response of each input individually. This calibration is then used "on the fly" when the flat panel loudspeaker is in use to control the flat panel loudspeaker's actuators. This process is computationally intensive.

[0009] EP 1 959 714 A1 describes a device and a method for regenerating acoustic signals via a loudspeaker equipped with common vibration plates in the left and right channels.

[0010] US 2001 / 0 022 835 A1 describes a loudspeaker device that can ensure a large size and area of a diaphragm so as to improve reproducibility in a low sound range and increase an output sound pressure, and can form a plurality of vibration points (signal control points).

[0011] US 2004 / 0 223 620 A1 describes a sound system that obtains a desired sound field from an array of sound sources arranged on a panel. The desired sound field allows a listener to perceive the sound as if it were coming from a live source and from a specific location. The setup of the sound system involves arranging a microphone array next to the array of sound sources to obtain a generated sound field. Then, for each sound source within the array of sound sources, arbitrary finite impulse response filters are assembled. Through iteration, the filter coefficients are optimized so that the generated sound field resembles the desired sound field, thus performing multi-channel equalization and wave field synthesis. Once the filters are set up, the microphones can be removed. BRIEF SUMMARY OF THE INVENTION

[0012] The inventors of the present patent application have realized that this known arrangement for controlling multiple patches or actuators to drive a planar loudspeaker is not only computationally intensive but also not particularly effective in practice, since different patches or actuators excite modes with opposite phases, thereby canceling out their contributions. In a broader sense, the inventors of the present patent application have realized that intelligently selecting signals to cooperatively drive the multiple patches is advantageous for realizing a practical, efficient planar loudspeaker driven by a plurality of patches or actuators, in other words, so that their contributions do not inadvertently cancel each other out.The inventors of the present patent application have realized that this can be achieved by first observing the frequency response of the planar loudspeaker to inputs applied to a plurality of actuators of the planar loudspeaker simultaneously, and then pre-configuring a controller to control the planar loudspeaker to take this frequency response into account. The pre-configuration can be very simple, such as a filter, e.g., a low-pass filter and / or all-pass. In this way, in use, low computational requirements are placed on a planar loudspeaker controller, and embodiments of aspects of the present invention provide good audio quality over a wide frequency range when a planar loudspeaker is driven by a plurality of patches or actuators.

[0013] The various aspects of the invention are defined in the following independent claims, to which reference is now made. Advantageous features are subject to the dependent claims.

[0014] In general, embodiments of the invention relate to panel-shaped loudspeakers, more particularly to resonant panel-shaped loudspeakers, which are either stand-alone or integrated with another article and typically provide another function, such as a structural function.

[0015] The following describes in more detail arrangements which take the form of a planar loudspeaker control comprising a plurality of actuators for controlling a planar loudspeaker. The planar loudspeaker control comprises a plurality of electrical signal inputs, a plurality of signal processors, and a signal processor control. Each input of the plurality of electrical signal inputs is assigned to a corresponding actuator of the planar loudspeaker to be controlled. Each signal processor of the plurality of signal processors is assigned to a corresponding input and has an output for an electrical signal for controlling an actuator of the planar loudspeaker. Each signal processor implements a transfer function from its input to its output based on each actuator of the planar loudspeaker for a desired sound receiver (sic). The signal processor control is assigned to all of the plurality of signal processors.The signal processor control is preconfigured to improve the phase angle between the signals as the overall output at the outputs of the signal processors.

[0016] A flat panel speaker including the flat panel speaker control may be provided.

[0017] Further arrangements for preconfiguring the signal processor control are described in more detail below. They take the form of an electronic device configured to configure a signal processor control of a flat panel loudspeaker comprising a plurality of actuators. The electronic device has the following configuration: Electrical signals are fed to a plurality of electrical signal inputs of the electrical device. A corresponding actuator of the flat panel loudspeaker to be controlled is assigned to each input. The response of the flat panel loudspeaker to the electrical inputs as a whole is measured. This response is used to configure the signal processor control assigned to all of a plurality of signal processors and, when used, to improve the phase angle between the signals output at the outputs of the plurality of signal processors as a whole.Each signal processor is assigned a corresponding input and has an output for an electrical signal to control an actuator of the flat panel loudspeaker. Each signal processor implements a transfer function from its input to its output based on each actuator of the flat panel loudspeaker for a desired sound receiver, such as a microphone or the user's ear.

[0018] These arrangements provide better or more precise audio control from a flat panel speaker. These arrangements require little computing power.

[0019] According to one aspect of the present invention, a planar loudspeaker controller is provided for controlling a planar loudspeaker comprising a plurality of actuators, the planar loudspeaker controller comprising: a plurality of electrical signal inputs, each input being assigned to a corresponding actuator of the planar loudspeaker to be controlled, a plurality of signal processors, each signal processor being assigned to a corresponding input and having an output for an electrical signal for controlling an actuator of the planar loudspeaker, and each signal processor implementing a transfer function from its input to its output based on each actuator of the planar loudspeaker for a desired sound receiver (sic), and a signal processor controller assigned to each of the plurality of signal processors, the signal processor controller being preconfigured such thatthat it improves the phase difference between the signals as a total output at the outputs of the signal processors.

[0020] The signal processor controller may include a filter preconfigured to improve the phase difference between the signals as the overall output at the signal processor outputs. The filter may include a low-pass filter and / or an all-pass filter. The low-pass filter may pass signals with a frequency below a cutoff frequency of 500 Hz. Each signal processor may include a digital signal processor. The signal processor controller may include a digital signal processor preconfigured to improve the phase difference between the signals as the overall output at the signal processor outputs. Signal processing may be applied by the signal processor controller to the electrical signal inputs to achieve a maximum or near-maximum overall output at the outputs at all frequencies.Signal processing may be applied by the signal processor controller to the electrical signal outputs to achieve a minimum or near-minimal sound pressure in at least one predetermined spatial location. The predetermined spatial location may be separated from one or more locations with maximum or near-maximum total output. The signal processor controller may include an equalizer to be preconfigured to improve the phase difference between the signals as the overall output at the outputs of the signal processors, the equalizer equalizing the input signals. The equalizer provides a single global equalization of the overall net output. The plurality of actuators may include at least one piezoelectric actuator, such as a piezoelectric patch and / or at least one coil-magnet actuator. The plurality of actuators may include an actuator array.The plurality of actuators may comprise distributed-mode actuators (DMAs). The sound receiver may comprise a user's ear or a microphone.

[0021] A planar loudspeaker may be provided which includes a planar loudspeaker control as described above.

[0022] An electronic device, such as a computer such as a tablet or laptop, or a display device, such as an LCD display, may be provided which includes the planar loudspeaker described above.

[0023] According to a further aspect of the present invention, a method for controlling a planar loudspeaker comprising a plurality of actuators is provided, the method comprising: inputting a plurality of electrical signal inputs, each input being assigned to a corresponding actuator of the planar loudspeaker to be controlled, a plurality of signal processors, each signal processor being assigned to a corresponding input and having an output for an electrical signal for controlling an actuator of the planar loudspeaker, and each signal processor implementing a transfer function from its input to its output based on each actuator of the planar loudspeaker for a desired sound receiver, and a signal processor controller assigned to each of the plurality of signal processors,wherein the signal processor control improves the phase angle between the signals as a total output at the outputs of the signal processors due to a preconfiguration.,

[0024] According to a further aspect of the present invention, an electronic device is provided which is configured to control a signal processor of a planar loudspeaker comprising a plurality of actuators, the electronic device being configured to perform the following functions: inputting electrical signals into a plurality of electrical signal inputs, each input being associated with a corresponding actuator of the planar loudspeaker to be controlled, Measuring a response of the planar loudspeaker to the electrical inputs as a whole and using this response to configure a signal processor controller associated with all of a plurality of signal processors, and when used to improve the phase angle between the signals output at the outputs of the plurality of signal processors as a whole, wherein each signal processor is associated with a respective input and has an output for an electrical signal for controlling an actuator of the planar loudspeaker, and each signal processor implements a transfer function from its input to its output based on each actuator of the planar loudspeaker for a desired sound receiver (sic).

[0025] The input electrical signals, actuators, flat panel loudspeakers, and response can be implemented virtually. The input electrical signals can be in the form of a pulse, and the response can be in the form of an impulse response. The electronic device can be configured to use the response to configure the signal processor control by evaluating differences between the signal processors' transfer functions.

[0026] In a further aspect of the present invention, a method is provided for configuring a signal processor control of a planar loudspeaker comprising a plurality of actuators, the method comprising: inputting electrical signals to a plurality of electrical signal inputs, each input being associated with a corresponding actuator of the planar loudspeaker to be controlled, measuring a response of the planar loudspeaker to the electrical inputs as a whole and using the response to configure a signal processor control associated with all of a plurality of signal processors, and when used to improve the phase angle between the signals output at the outputs of the plurality of signal processors overall,wherein each signal processor is associated with a corresponding input and has an output for an electrical signal for controlling an actuator of the planar loudspeaker, and each signal processor implements a transfer function from its input to its output based on each actuator of the planar loudspeaker for a desired sound receiver.

[0027] The input electrical signals may be in the form of a pulse, and the response may be in the form of an impulse response. Using the response to configure the signal processor controller may involve assessing differences between the signal processors' transfer functions.

[0028] According to a further aspect of the present invention, an electronic device is provided which is configured to configure a signal processor control of a planar loudspeaker comprising a plurality of actuators using a response of the planar loudspeaker to electrical inputs each associated with a corresponding actuator of the planar loudspeaker, wherein the signal processor control is associated with all of a plurality of signal processors and, when used to improve the phase angle between the signals output at the outputs of the plurality of signal processors, is configured as a whole,wherein each signal processor is associated with a corresponding input and has an output for an electrical signal for controlling an actuator of the planar loudspeaker, and each signal processor implements a transfer function from its input to its output based on each actuator of the planar loudspeaker for a desired sound receiver (sic).

[0029] A computer program may be provided to carry out the method described above. A non-volatile computer-readable data carrier comprising instructions may be provided to carry out the method described above. The non-volatile computer-readable data carrier may be a CD-ROM, DVD-ROM, hard disk, or solid-state storage device such as a USB stick. BRIEF DESCRIPTION OF THE DRAWINGS

[0030] The invention will now be described in more detail by way of example and with reference to the accompanying drawings. Fig. a schematic representation of a flat panel loudspeaker control according to one aspect of the present invention, Fig. a schematic representation of a planar loudspeaker according to one aspect of the present invention, Fig. a graphical representation of a simulated sound pressure level response versus frequency of the two sources of the flat panel loudspeaker of the Fig. , Fig. a schematic representation of the flat panel loudspeaker of the Fig. , Fig. a graphical representation of a simulated sound pressure level response of the two sources of the flat panel loudspeaker of the Fig. which were combined using a naive summation and a summation using a planar loudspeaker control according to one aspect of the present invention, Fig. a graphical representation of the surface deformation and pressure distribution of the flat panel loudspeaker of the Fig. Hz, Fig. a graphical representation of the surface deformation and pressure distribution of the flat panel loudspeaker of the Fig. , Fig. a block diagram of a parallel solver of an example of the flat panel loudspeaker control of the Fig. , Fig. a block diagram of a recursive solver of an example of the flat panel loudspeaker control of the Fig. , Fig. a schematic representation of part of another planar loudspeaker according to one aspect of the present invention, Fig. a schematic representation of a back of a device that has a Fig. partially shown flat panel loudspeakers, Fig. a schematic representation of the back of another device that has a Fig. partially shown flat panel loudspeakers, Fig. a schematic representation of the back of the Fig. and a pair of Fig. partially shown flat panel loudspeakers, Fig. a graphical representation of a simulated sound pressure level response versus frequency of the combined and single source of a flat panel loudspeaker, including the Fig. shown part, Fig. a graphical representation of a simulated sound pressure level response versus a frequency of the device of the Fig. at different air distances from the device, Fig. a graphical representation of a simulated sound pressure level response versus a frequency of the device of the Fig. , Fig. a schematic representation of another planar loudspeaker according to one aspect of the present invention, Fig. a graphical representation of a simulated sound pressure level response versus a frequency of the device of the Fig. for two different patch sizes, Fig. a graphical representation of a simulated sound pressure level response versus the frequency of the device of the Fig. which were combined using a naive summation and a summation using a planar loudspeaker control according to one aspect of the present invention, Fig. a graphical representation of amplitude transfer functions versus the frequency of the device of the Fig. for two different patch sizes ( Fig. refers to a relatively small patch, Fig. on a relatively large patch). DETAILED DESCRIPTION OF THE INVENTION

[0031] Now, with reference to the Fig. an exemplary flat panel loudspeaker control 100 for controlling a flat panel loudspeaker 101 is described. The flat panel loudspeaker control of the Fig. is to control n actuators (where n>1) to excite a surface of a planar loudspeaker.

[0032] The flat panel loudspeaker control 100 of the Fig. has a plurality of electrical signal inputs 102. It is a single or unified device with n input channels. Each input is assigned to one of the n actuators of the planar loudspeaker to be controlled. The controller has n signal processors 104. Each signal processor is assigned to a corresponding input. Each signal processor has an output 106 for an electrical signal for controlling an actuator of the planar loudspeaker. Each signal processor performs a transfer function from its input to its output, based on each actuator of the planar loudspeaker, to a desired sound receiver, such as an ear or ears of a person expected to hear audio from the planar loudspeaker, or a microphone spaced apart from the planar loudspeaker. Also provided is a signal processor controller 108 assigned to all of the plurality of signal processors.The signal processor control is preconfigured to improve the phase angle between the signals overall or as a total output at the signal processor outputs. The preconfiguration is described in more detail below.

[0033] Fig. shows an example of a flat panel loudspeaker 101 controlled by the flat panel loudspeaker controller 100 of the Fig. The flat panel loudspeaker has a flat radiating panel 110, whose dimensions in this example are 150 mm x 100 mm. The panel comprises a plurality of different material layers, the details of which are not directly relevant to the operating principle. Fig. is a conceptual or schematic representation of the planar loudspeaker. The other half is the exact mirror image in the YZ plane 111 and is not shown for clarity.

[0034] The plate 110 is connected to the rest of a device, e.g., an LCD television cabinet (not shown), via a mixture of continuous 112 and localized 114 boundary terminations. The former seal the edges of the plate. The latter provide an anchor point in the center.

[0035] In this example, two identical coil-magnet actuators 116, 117 are used on each half of the plate 110 (for clarity, Fig. Only the coil coupling rings are shown. The placement of the actuators is largely dictated by industrial design constraints, such as the position of other components of the LCD television, and particularly its backlight. The placement of the actuators can be chosen by referring, for example, to US Patent No. US 6,332,029 B1 or US Patent No. US 6,546,106 B2.

[0036] Fig. shows simulated sound pressure levels (SPLs) (in dB) versus the input frequency from the flat panel loudspeaker 101 of the Fig. (The frequency response of actuator 1 or source 1, 116, is shown by a solid line 119, and the frequency response of actuator 2 or source 2, 117, is shown by a dashed line 121.) Of particular note in these responses are the peaks 118 and 120 at approximately 150 and 350 Hz, respectively; the exact frequencies depend on the components used. The former peak is due to the resonance of the actuators, the latter is due to the main plate mode.

[0037] As with reference to Fig. As shown, Source 1 (Actuator 116) generally produces a higher pressure response in this example. For stereo separation reasons, using Source 2 (Actuator 117) would be preferable at higher frequencies, but at lower frequencies, both must be used to improve the frequency response. COMBINATION STRATEGIES

[0038] Fig. is a schematic representation of the two-actuator system of the Fig. P1 is a transfer function of actuator 1 and P2 is a transfer function of actuator 2.

[0039] In this example, a common input signal is supplied to the two actuators, actuator 1 and actuator 2.

[0040] There is a transfer function from the input of each actuator to a target T where we want to control the signal level. These (frequency-dependent) transfer functions are the transfer functions P1 and P2.

[0041] We want to apply (frequency-dependent) gains to both channels, namely gain “a” to channel 1 and gain “-b” to channel 2. The total signal arriving at T is therefore: T=a⋅P1−b⋅P2

[0042] All variables can be complex, i.e. have an amplitude and a phase or real and imaginary parts, which is equivalent.

[0043] The total energy input to the actuators is: A=|a|2+|b|2=a⋅a*+b⋅b* where a* is the complex conjugate of a and b* is the complex conjugate of b (generally, an * next to a variable means a complex conjugate of the variable).

[0044] The total energy arriving at T is given by: |T|2=|a⋅P1−b⋅P2|2=(a⋅P1−b⋅P2)⋅(a*P1*−b*⋅P2*)

[0045] We are interested in the stationary points of |T| 2, which can be determined using simple integral calculus.

[0046] d|T| 2 / da* = (a·P1 - b·P2) -P1* and d|T| 2 / db* = (a·P1 - b·P2)·(- P2*) simultaneously.

[0047] There are two main sets of solutions for this pair of equations, namely: (a·P1 - b·P2) = 0, or a=P2, b=P1, which gives the minimum local output energy. a = P1*, b = -P2*, which gives the maximum local output energy.

[0048] The values of a and b can be normalized by restricting the input energy.

[0049] If we express the simultaneous equations in matrix form, we get (the overline indicates a complex conjugation): M⋅v=(P1¯⋅P1−P1¯⋅P2P2¯⋅P1−P2¯⋅P2)(ab)=(00)

[0050] The two eigenvectors of M correspond to the two solutions, with their corresponding eigenvalues giving the total energy.

[0051] The same principles can be extended to any number of actuator channels as well as to multiple targets.

[0052] The maximum response that can result from the combined individual input powers of two actuators is the square root of the sum of squares. In other words, |a-P1-b·P2| 2 in accordance with |a 2 |+|b 2 |=1 to maximize.

[0053] One solution is that: a=P1¯|P12|+|P22|,b=−P2¯|P12|+|P22| (where the overline indicates a complex conjugation)

[0054] One solution would be to add the response pressures, but this is divided by the square root of 2 to obtain the power limitation.

[0055] This gives the solution (a naive solution): a=b=12

[0056] Fig. shows a comparison between a naive solution and a solution showing an example of the present invention. Fig. shows sound pressure levels (SPLs) versus frequency for a naive summation (shown by solid line 140), a naive subtraction (shown by dashed line 143), and an optimal summation (shown by solid line 142) provided by an example of a flat panel loudspeaker control system according to the invention. In Fig. It can be seen that this naive summation solution works quite well at frequencies up to about 600 Hz, but not so well between 600 Hz and 4 kHz. This is explained with reference to Fig. to explain.

[0057] Fig. shows surface deformation and pressure distribution of the flat panel loudspeaker 101 of the Fig. Hz. In Fig. The following applies: The lighter the shading, the greater the displacement of the flat panel loudspeaker. Fig. It can be seen that the entire surface moves with similar polarity at low frequency (500 Hz), so the in-phase inputs add up constructively.

[0058] Fig. shows surface deformation and pressure distribution of the flat panel loudspeaker 101 of the Fig. kHz. In Fig. The following applies: The lighter the shading, the greater the displacement of the flat panel loudspeaker. Fig. It can be seen that at higher frequencies the surface moves with opposite polarity at the two source points, ie the in-phase inputs add up destructively.

[0059] The inventors of the present application have realized that these features, if effectively considered at the design stage of the planar panel loudspeaker 101 rather than only after it is already in use, can be solved without undue computational effort or cost-effectively when the planar panel loudspeaker is in use. These features can be taken into account by an electronic device, e.g., a general-purpose computer such as a desktop or laptop computer on which appropriate software or a computer program is installed. The input of the electrical signals is input, simulated, or virtually provided in the form of a pulse from the computer to a plurality of electrical signal inputs, each input being associated with each actuator of the planar panel loudspeaker to be controlled.The computer then measures a response of the planar loudspeaker, in the form of an impulse response, to the electrical inputs as a whole (real, simulated, or virtual). The computer then uses the response to configure a signal processor controller associated with all of a plurality of signal processors to, in use, improve the overall phase difference between the signals output at the outputs of the plurality of signal processors. The response is used by the computer to configure the signal processor controller by assessing differences between transfer functions of the signal processors. The preconfigured signal processor controller 108 of the planar loudspeaker controller 100, in use, ensures an improvement in the phase difference between the signals output by the loudspeaker controller. A frequency response of such an arrangement is shown by the solid line 142 in FIG. Fig. .

[0060] In the example of Fig. Various arrangements can be provided to preconfigure the flat panel loudspeaker controller 100 or to provide it with predetermined features. These ensure phase reversal at different operating frequencies of the actuators 116, 117 of the flat panel loudspeaker 101. The signal processor controller 100 can, for example, be preconfigured to have at least one of the following features.

[0061] The signal processor control 108 of the Fig. can be preconfigured to include a filter to filter out one of the inputs 102 to one of the actuators 116, 117 of the flat panel loudspeaker starting at approximately 500 Hz. The signal processor control can be preconfigured to include all-pass filters to switch the polarity of an actuator or source 116, 117 from approximately 600 Hz and optionally switch back again at 4 kHz. The signal processor control can be preconfigured to apply digital signal processing to the input signals 102 to the actuators 116, 117 to achieve near maximum overall power at all frequencies. The signal processor control can be preconfigured to equalize the input signals 102 to the actuators 116, 117 to ensure a flatter frequency response.

[0062] If different motor systems are provided for the two sources or actuators 116, 117, e.g. if a larger, more powerful motor with more inductance is provided for the low-frequency source and a smaller motor with less inductance is provided for the high-frequency source, the frequency response and thus also the preconfiguration of the signal processor control 108 will also differ.

[0063] For a larger system with more input channels and thus more actuators, the inventors of the present application have found that the frequencies at which phase becomes more important are lower, so the filter selection for pre-configuring the flat panel loudspeaker controller 100 is correspondingly more complicated.

[0064] The preconfiguration of the signal processor controller 108 of the flat panel loudspeaker controller 100 can be as follows. These methods calculate the optimal filtering to be applied to the various input signals 102. They can be implemented by a computer on which appropriate software is installed. A simple maximization problem and its solution using the “tan-theta approach”

[0065] Now we will look at the example of Fig. and the schematic representation of a system comprising two actuators in Fig. , ie a system with two inputs and one output. The transfer function of input 1 (ie the first input 102 of the Fig. ) to the output is represented by P1 and the transfer function of input 2 (ie the second exciter 102 of the Fig. ) to the output 106 through P2. For the input signals a and -b, the output signal spectrum T is thus given by: T=a⋅P1−b⋅P2 where a, b, P1, P2 and T are all complex frequency functions.

[0066] The problem to be solved is to find the stationary points (points on a curve where the gradient is zero) T for all frequencies. There is no unique solution to this problem, but it is clear from observations that a and b should be related, namely: b=a⋅P1 / P2 or a=b⋅P2 / P1

[0067] Using these ratios is generally not a good idea, since either P1 or P2 may contain zeros. A simple solution, as described above, is to set a = P2 and b = P1. The solution can be normalized to energy units, so |a| 2 + |b| 2 = 1. Since P1 and P2 are generally complex quantities, the absolute values are important. Therefore, a stationary value of T is given by: a=P1¯|P1|2+|P2|2,b=P1|P1|2+|P2|2

[0068] By the way, T is maximized to unity by: a=P1¯|P1|2+|P2|2,b=−P2¯|P1|2+|P2|2

[0069] If P1 or P2 are measured far from the input, which is generally the case in the field of acoustics, the transfer function includes excess phase in the form of a delay. Consequently, these values of a and b may not be the best choice. If we set a = cos(θ) and b = sin(θ) (i.e., convert from Cartesian to polar coordinates), the problem changes from an underdetermined simultaneous equation with two variables to a simple equation in the new variable θ (the other implicit variable is the radius, which is given by r 2 = a 2 + b 2, but we want to keep this constant and therefore set this value to unity). If a = cos(θ) and b = sin(θ), then: tan(θ) = P1 / P2. This solution is called the "tan-theta solution" and results in a and b with a much smaller phase excess. It is clear from the trigonometric identity that a 2 + b 2 = 1, but since θ is generally complex, |a| 2 + |b| 2 ≠ 1, so normalization is required.

[0070] In this simple example, the problem is solved by investigation. Since this is not necessarily possible in general, it is advantageous to have a systematic procedure for finding the solution; this is explained below. VARIATION PROCESS

[0071] The goal here is to determine the values of parameters that yield stationary values as a function (ie, to find nodes, lines, or pressures). The first step of the procedure is to form the energy function. For our example, the squared remainder of T can be used, ie, E = |T| 2 = |a·P1 - b·P2| 2 . The stationary values occur at the maximum and minimum of E. E=(a⋅P1−b⋅P2)⋅(a⋅P1−b⋅P2¯)

[0072] The values of a and b are subject to a constraint, meaning they cannot both be zero. This constraint can be expressed using a so-called "Lagrange multiplier," λ, to modify the energy equation. λ is a new variable introduced to enforce the constraint equation: |a| 2 + |b| 2 = 1. So (if E is the energy): E=(a⋅P1−b⋅P2)⋅(a⋅P1−b⋅P2¯)+λ⋅(a¯⋅a+b¯⋅b−1)

[0073] The complex conjugate of each variable can be considered as an independent variable. E is differentiated with respect to each conjugate variable in turn, so: ∂E∂a¯=(a⋅P1−b⋅P2)⋅P1¯+λ⋅a

[0074] At the stationary points, both must be equal to zero. It is immediately apparent that the solutions found in the previous section also hold here. However, if one further solves the system of equations formally, the equations are first combined to eliminate λ by finding the values: (a⋅P1−b⋅P2)⋅P1¯⋅b+(a⋅P1−b⋅P2)⋅P2¯⋅a=0

[0075] The resulting equation is quadratic at a and b, the two solutions corresponding to the maximum and minimum values of E. By introducing a = cos(θ) and b = sin(θ), a quadratic equation in tan(θ) is obtained, although strictly speaking, the Lagrangian constraint is not satisfied. P1⋅P2¯+(|P1|2−|P2|2)⋅tan(θ)−P2⋅P1¯⋅tan(θ)2=0

[0076] It should be noted that in many cases, (P1| 2 -|P2| 2 ) 2 + 4 · P1 · P2 · P2 · P1 = (P1| 2 + |P2| 2 ) 2 , find the same answers as before, namely: θ=arctan(P1P2) for the minimum value and θ=arctan(−P2P1)¯ for the maximum value.

[0077] For the sake of completeness, it should be noted that this identity does not necessarily hold in the typical case where P1 and P2 are sums or integrals of responses. Nevertheless, it is possible to systematically find both stationary values using this modification of the "tan-theta approach." A use case is discussed in more detail below to demonstrate how these solutions can be used in the examples described above. USE CASE 1: MAXIMUM ACOUSTIC RESPONSE

[0078] If everything is perfectly symmetrical, the stationary points are trivial—a and b are equated. If there is asymmetry in the system, this assumption no longer holds. The task to be solved is to find two sets of input values, a and b, that yield maximum audio performance when desired and minimum audio performance when undesired. This is precisely the task solved in the section on "variational methods."

[0079] P1 and P2, which in Fig. shown as sound pressure levels (SPL) in dB are the acoustic responses at 10 cm resulting from a finite element simulation of the plate-shaped loudspeaker configuration of the Fig. they could just as easily have been determined by measurement.

[0080] With reference to Fig. the result of using an optimal filter pair (line 142) (max and min according to the two solutions for θ) with the pair of simple sum (line 140) and difference (line 143) in Fig. The summed response is higher than the subtracted response over much of the band, but this is not always the case. Although the axial response (the response at an air gap from the flat panel speaker) is not very informative, the averaged results across the front hemisphere show similar characteristics.

[0081] The solution described above can be applied to extended areas by measuring the target at several discrete sampling points. In this case, it is desirable to find the stationary points of the outputs simultaneously by manipulating the inputs. There are now more output signals than inputs, so the result is not exact. This is one of the strengths of the variational method—it allows the best approximation to be determined. ∑iTi=∑i|a⋅P1i−b⋅P2i|2=∑i(a⋅P1i−b⋅P2i)⋅(a⋅P1i−b⋅P2i)¯ E=∑i(a⋅P1i−b⋅P2i)⋅(a⋅P1i−b⋅P2i)¯+λ⋅(a¯⋅a+b¯⋅b−1) ∂E∂a¯=∑i(a⋅P1i−b⋅P2i)⋅P1i¯+λ⋅a ∂E∂b¯=∑i(a⋅P1i−b⋅P2i)⋅P2i¯+λ⋅b

[0082] If these are solved as before, the result is: S12+(S11−S22)⋅tan(θ)−S21⋅tan(θ)2=0 where Snm=∑iPni⋅Pmi¯(for the Pn. actuator at the i. measuring point) θm=arctan(S11−S22+(S11−S22)2+4⋅S12⋅S212⋅S21) gives the minimum θp=arctan(S11−S22+(S11−S22)2+4⋅S12⋅S212⋅S21) gives the maximum

[0083] The method also extends to integrals and to more than two inputs.

[0084] The error function and the sums can be replaced by integrals, for example: E=∯|a⋅P1(r_)−b⋅P2(r_)|2dA+λ⋅(a¯⋅a+b¯⋅b−1) Snm=∯Pn(r_)⋅Pm(r_)¯dA USE CASE 2: DUAL-REGION ACOUSTICS

[0085] It is possible to specify a minimum response at a selected location or spatial position and a non-zero response at another selected location or spatial position. In other words, the signal processor control of the flat panel loudspeaker control can apply signal processing to the electrical signal inputs to achieve a minimum or near-minimal sound pressure at at least one specified location. This is very useful in dual-region systems. STRONG SOLUTION

[0086] We have (for example) two inputs to generate a node and an acoustic response at another point. Define transfer functions Pi_j from input i to output j.

[0087] Solve a·P1_1 + b·P2_1 = 0 and a·P2_1 + b·P2_2 = g simultaneously. (P1_1P2_1P1_2P2_2)(ab)=(0g),(ab)=(P1_1P2_1P1_2P2_2)−1(0g), a=−P2_1P1_1⋅P2_2−P1_2⋅P2_1⋅g, b=P1_1P1_1⋅P2_2−P1_2⋅P2_1⋅g,

[0088] As long as the denominator is never equal to zero, this pair of transfer functions results in a nodal response at point 1 and a complex transfer function exactly equal to g at point 2. WEAK SOLUTION

[0089] |a·P1_1 + b·P2_1| 2 = 0 and |a·P2_1 + b·P2_2| 2 = |g| 2 solve simultaneously.

[0090] Using the variational methods explained below, the first minimization for a and b is solved and the result is then normalized to satisfy the second equation. a=r⋅cos(θ), b=−r⋅sin(θ), tan(θ)=−P1_1P1_2, r2⋅|(cos(θ)⋅P2_1−sin(θ)⋅P2_2)|2=|g|2, i.e. r.

[0091] As long as the denominator is never equal to zero, this pair of transfer functions results in a nodal response at point 1 and a power transfer function equal to |g| 2is, at point 2. This resulting output at point 2 does not necessarily have the same phase response as g, so the constraint is not as strong.

[0092] There are other extensions to the methods described above that are particularly relevant when considering more than two input channels. These are general extensions that would be equally applicable to the two-channel case. Furthermore, using eigenvalue analysis as a tool yields the best solution, which is not the exact solution, if no exact solution is available. ON THE RELATIONSHIP BETWEEN THE VARIATIONAL METHOD AND THE PROBLEM OF EIGENVALUES

[0093] The following minimization of an energy function of the form E results in a set of simultaneous equations: E=|∑nan⋅Pn|2,∂E∂an¯=Pn¯⋅∑nan⋅Pn=0 for all n, where P iare the inputs to the system and ai are the constants applied to these inputs, ie a and b in the previous two-channel system.

[0094] This system of equations can be written in matrix form, i.e.: M__⋅v_=0, where M__i,j=Pi¯⋅Pj and where v_i=ai

[0095] It should be noted that M is conjugate symmetric, ie M__j,i=M__i,j¯.

[0096] We want to find a non-trivial solution, that is, a solution other than the trivial v = 0, which is mathematically valid but not particularly useful.

[0097] Since any linear scaling of v also represents a solution to the equation, the ai are not uniquely defined. An additional equation is required to constrain the scaling. Another way of looking at this is that for an exact solution, the number of input variables must be greater than the number of measurement points. In any case, there is one more equation than free variables, so the determinant of M will be zero.

[0098] Consider the matrix eigenvalue problem, where we have to find a non-trivial solution to the following equation: M__⋅v_−λ⋅v_=0, where λ is an eigenvalue and the associated v_ is the eigenvector.

[0099] Since M is conjugate symmetric, all eigenvalues will be real and non-negative. If λ = 0 is a solution to the eigenvalue problem, we are back to our original equation. Therefore, v is the eigenvector for λ = 0.

[0100] The special significance of this procedure is that even if there is no solution for (1), the solution of (2) with the smallest λ value represents the closest approximation to the answer.

[0101] Based on the task described above, for example: (P1¯⋅P1−P1¯⋅P2−P2¯⋅P1P2¯⋅P2)⋅(ab)−λ⋅(ab)=0 has a solution 1 = 0, b / a = P1 / P2.

[0102] The other eigenvalue corresponds to the maximum, i.e. λ = |P1| 2 + |P2| 2 , b / a=-P2 / P1.

[0103] When an eigenvalue solver is used to determine the values of ai, the scaling applied is essentially arbitrary. The usual practice is to normalize the eigenvector, which fixes the amplitudes: ∑i|ai|2=1 For example: a=P2|P1|2+|P2|2, b=P1|P1|2+|P2|2

[0104] However, the reference phase is still arbitrary - if v represents a normalized solution of the eigenvalue problem, then so is ve jθ . What constitutes the best value for θ and how to determine it is the subject of a later section.

[0105] The magnitude of the eigenvalue λ is simply the energy associated with this choice of eigenvector. The proof follows: E=|∑nan⋅Pn|2=∑nan⋅Pn⋅∑mam¯⋅Pm¯=∑mam¯⋅(∑nPm¯⋅Pn⋅an)=∑mam¯⋅(∑nMmn⋅an)

[0106] Based on our eigenvalue equation and the normalization of the eigenvector, we can further say: E=∑mam¯⋅(∑nMmn⋅an)=∑mam¯⋅(λ⋅am)=λ⋅∑mam¯⋅am=λ SOLUTION TO THE EIGENVALUE PROBLEM

[0107] In principle, a system of order n has n eigenvalues, which are determined by solving an nth-order polynomial equation. However, not all eigenvalues are needed. The best solution to the minimization problem is the smallest eigenvalue. If the eigenvalue happens to be zero, it is an exact solution. The best solution to the maximization problem is the largest eigenvalue. M__⋅v_−λ⋅v_=0 yields |M−λ⋅I|=0, which yields ∏i=1n(λ−λi)=0.

[0108] If e gives its exact solution to the problem, then the determinant will have λ as a factor. For example: |(abb¯c)−λ⋅(1001)|=|(a−λbb¯c−λ)|=(a−λ)(c−λ)−|b|2=0 a⋅c−|b|2−(a+c)⋅λ+λ2=0

[0109] If a · c - |b| 2 = 0, there is an exact solution.

[0110] Since the number of equations is greater than the number of unknowns, there is more than one possible set of solutions for v, but they are all equivalent: (a−λ)⋅v0+b⋅v1=0, v1v0=λ−ab b¯⋅v0+(c−λ)⋅v1=0,v1v0=bλ−c

[0111] For example: a=2, b=1+1j, c=3; 6−2−5⋅λ+λ2=0; λ=1.4 (λ−2) / (1+1j)=(−1+1j) / 2 or 1−1j (1−1j) / (λ−3)=(−1+1j) / 2 or 1−1j

[0112] The best solution of the pair of equations is therefore v1 / v0 = (-1 + 1j) / 2. CHOOSING THE BEST SCALING SOLUTION

[0113] Mathematically speaking, any solution to the problem of preconfiguring a signal processor controller to improve the phase angle between the signals output by the signal processor controller as a total output at the signal processor outputs is as good as any other. However, here we have to solve a technical problem. Both the matrix M and its eigenvectors v are functions of frequency. We want to use the components of v as transfer functions, so it is preferable that no sudden sign or phase changes occur. M(ω)__⋅v(ω)_=0

[0114] In the problem with two variables, the substitution a = cos(θ) and b = sin(θ) was used to then calculate tan(θ). This procedure yields values for a and b with a small phase excess. However, this approach quickly becomes unwieldy, as it is always complicated to form these equations, let alone solve them. For 3 variables, for example, we have 2 angles and can use the spherical polar mapping to give a = cos(θ) · cos(φ), b = cos(θ) · sin(φ), c = sin(θ).

[0115] Instead, we want to use the variational method to determine the best value for θ. The "best" value is defined as the value with the smallest overall imaginary component.

[0116] Now: v' = v e jθ , v = vr + j·vi, and our error energy is defined as follows: SSE=∑iIm(v'i)2=∑iIm((vri+j⋅vii)⋅(cos(θ)+j⋅sin(θ)))2=∑i(vii⋅cos(θ)+vri⋅sin(θ))2

[0117] The following applies: rr=Re(v)⋅Re(v)=∑vr12,ii=Im(v)⋅Im(v)=∑vii2,ri=Re(v)⋅Im(v)=∑vri⋅vii

[0118] Then: SSE=cos(θ)2⋅ii+2.cos(θ)⋅sin(θ)⋅ri+sin(θ)2⋅rr (where θ = 0, SSE = ii, which represents our initial costs. We want to reduce these if possible).

[0119] Now differentiate with respect to θ to give the following equation: 2⋅(cos(θ)2−sin(θ)2)⋅ri+2⋅cos(θ)⋅sin(θ)⋅(rr−ii)=0

[0120] If by 2 cos(θ) 2 divided, the following quadratic equation results in tan(θ); ri+tan(θ)⋅(rr−ii)−tan(θ)2⋅ri=0

[0121] Of the two solutions, the one that gives the minimum value of SSE is: tan(θ)=rr−ii−(rr−ii)2+4⋅ri22⋅ri

[0122] If ri = 0, two special cases arise:

[0123] If ri = 0 and rr >= ii, then θ = 0.

[0124] If ri = 0 and rr < ii, then θ = π / 2.

[0125] The last step in choosing the best value for v is to ensure that the real part of the first component is positive (any component could be used for this), i.e. v'=v⋅ejθ If v'0<0,v'=−v' Example v=(0.908−0.419j0.770−0.638j0.9999−0.01j0.343−0939j) rr = 2.534. ii = 1.466· ri = -1.204; the solution gives θ = 0.577 v'=(0.990+0.143j0.993−0.115j0.844+0.537j0.800−0.600j) rr' = 3.318, ii' =0.682, ri = 0

[0126] It should be noted that minimizing ii simultaneously maximizes rr and sets ri equal to zero. COMPARISON OF METHODS - AN EXAMPLE

[0127] Consider a device with two inputs and two outputs (i.e., the device described above). There will be exact solutions for minimizing each output individually, but only an approximate solution for minimizing them simultaneously. P1_1=0.472+0.00344j, P2_1=0.479−0.129j P1_2=−0.206−0.195j, P2_2=0.262+0.000274j Two error contribution matrices form: M1=(0.2230.226−0.063j0.226+0.063j0.246);|M1|=0, i.e. exact solution possible; M2=(0.080−0.054+0.050j0.054−0.050j0.069);|M2|=0, i.e. exact solution possible; M1+M2=(0.3030.171−0.012j0.171+0.012j0.315);|M1+M2|=0.066

[0128] The “Tan-Theta method” is now used to solve the three cases. (ba)1=(−0.682−0.098j0.718−0.093j),(ba)2=(0.692+0.244j0.623−0.270j),(ba)1+2=(−0.694−0.025j0.719−0.024j)

[0129] There are two eigenvector solvers for the eigenvector method: one calculates all vectors simultaneously, and the other calculates a specific eigenvalue. They yield numerically different answers when the vectors are complex (both answers are correct). After applying the "best" scaling algorithm, both solvers yield the same answers, which are listed above.

[0130] M1: Eigenvalues, 0 and 0.469: Eigenvector before scaling: (-0.698 + 0.195j, 0.689 - 0.0013j) or (0.724, -0.664-0.184j) Eigenvector after scaling: (0.718 - 0.093j, -0.682 - 0.098j)

[0131] M2: Eigenvalues, 0 and 0.149: Eigenvector before scaling: (-0.5 + 0.46j, 0.734 - 0.0030j) or (0.498 - 0.462j, 0.724) Eigenvector after scaling: (0.623 - 0.270j, 0.692 + 0.244j)

[0132] M1 + M2: Eigenvalues, 0.137 and 0.480: Eigenvector before scaling: (-0.717 + 0.051j, 0.695 - 0.0007j) or (0.719, -0.693-0.049j) Eigenvector after scaling: (0.719 - 0.024j, -0.694 - 0.025j) ADDING A 3RD ENTRANCE

[0133] Now consider the contributions of a third input channel: P3_1=−0.067−0.180j P3_2=0.264+0.0014j Adding these contributions to the error matrices: M1=(0.2230.226−0.063j−0.032−0.085j0.226+0.063j0.246−0.009−0.095j−0.032+0.085j−0.009+0.095j0.037); |M1|=0 M2=(0.080−0.054+0.050j−0.055+0.051j−0.054−0.050j0.0690.069−0.0004j−0.055−0.051j0.069+0.0004j0.070); |M2|=0 M1+M2=(0.3030.171−0.012j−0.087−0.034j0.171+0.012j0.3150.061−0.095j−0.087+0.034j0.061+0.095j0.107); |M1+M2|=0

[0134] Now there is an exact solution to the joint problem, and the eigenvalue of M1 + M2 is zero.

[0135] (Note that M1 and M2 each have two eigenvalues equal to zero; in other words, they have a degenerate eigenvalue. There are now two completely orthogonal solutions to the problem, and any linear sum of these two solutions is also a solution.)

[0136] M1 + M2: Eigenvalues are 0, 0.218 and 0.506: Eigenvector after scaling: (0.434 - 0.011j, -0.418 + 0.199j, 0.764 + 0.115j)

[0137] As shown above, the tan-theta method is simpler and faster to implement for two inputs, but the scaled eigenvector method is easier for three or four inputs. The results are the same for both methods. For an exact solution, the number of input variables must be greater than the number of measurement points. Using eigenvalue analysis as a tool for the general problem yields the best solution when an exact solution is unavailable.

[0138] For the general task of minimizing input "m" and output "n", there are two principal variations of an algorithm to determine the best inputs m. These are called the parallel method ("all at once") and the serial method ("one at a time"). In general, they can be combined. If m > n, all paths lead to the same exact answer (taking rounding errors into account). If m <= n, only approximate answers are available, and the path taken will influence the final result. The serial method is useful when m <= n and some of the n outputs are more important than others. The important outputs are then solved exactly, and the best-fitting solution is found for the others. THE PARALLEL ALGORITHM (“ALL AT ONCE”)

[0139] Fig. is a block diagram of a parallel solver 150 for n x m data sets 152. An error matrix or data set 154 is formed. The eigenvector corresponding to the smallest eigenvalue is selected. If m > n, the eigenvalue is zero and the result is exact. RECURSIVE OR SEQUENTIAL ALGORITHM (“ONE AT A TIME”)

[0140] Fig. is a block diagram of a recursive solver 160. An error matrix of the most important output is formed, and the eigenvectors corresponding to the smallest eigenvalues (m - 1) are calculated. These are used as new input vectors, and the procedure repeats. The process ends with a 2 x 2 eigenvalue solution. By resetting, the solution to the original problem is compiled.

[0141] As with all recursive algorithms, this process can be converted into an iterative (or sequential) process. For the first m - 2 cycles, all outputs have exact solutions. For the remaining cycle, the best linear combination of these solutions is determined to minimize the remaining errors. Example 1: m = 3, n = 2 P1_1=0.472+0.00344j P1_2=−0.206−0.195j P2_1=0.479−0.129j P2_2=0.262+0.000274j P3_1=−0.067−0.180j P3_2=0.264+0.0014j ALL AT ONCE M1+M2=(0.3030.171−0.012j−0.087−0.034j0.171+0.012j0.3150.061−0.095j−0.087+0.034j0.061+0.095j0.107); |M1+M2|=0

[0142] M1 + M2: Eigenvalues are 0, 0.218 and 0.506: Eigenvector after scaling: (0.434 - 0.011j, -0.418 + 0.199j, 0.764 + 0.115j) ONE BY ONE

[0143] Exit 1 is solved, followed by exit 2. Since 3 > 2, the answer should be the same. M1=(0.2230.226−0.063j−0.032−0.085j0.226+0.063j0.246−0.009−0.095j−0.032+0.085j−0.009+0.095j0.037); |M1|=0

[0144] M1 + M2: Eigenvalues are 0, 0 and 0.506: Eigenvector V1: (0.748, -0.596 - 0.165j, 0.085 - 0.224j) Eigenvector V2: (-0.062 + 0.026j, 0.096 + 0.350j, 0.929)

[0145] New task: a and b are to be chosen such that a·V1 + b·V2 minimizes output 2.

[0146] The new transfer impedances are: pv1=(P1_2 P2_2 P3_2),V1=−0.287−0.250j pv1=(P1_2 P2_2 P3_2),V1=0.287−0.100j

[0147] Now the process is repeated with these two transfer impedances as outputs.

[0148] The new error matrix is: M1'=(0.145−0.107+0.043j−0.107−0.043j0.093); |M1'|=0,, ie exact solution possible;

[0149] M1' eigenvalues, 0 and 0.237

[0150] Eigenvector after scaling: (0.608 - 0.145j, 0.772 + 0.114j)

[0151] Now V1 and V2 are combined to determine the inputs: (0.608−0.145j)V1+(0.772+0.114) V2=(0.404−0.095j,−0.352+0.268j,0.737−0.042j)

[0152] Normalize and scale the result: (0.434 - 0.011j, -0.418 + 0.199j, 0.764 + 0.115j)

[0153] It should be noted that the result is the same as before, exactly as it should be.

[0154] Example 2: m = 3, n >= 3

[0155] There is 1 sound pressure output and various speed outputs.

[0156] The scaled acoustic error matrix is M1, the scaled error matrix for the summed velocity is M2: M1=(3.928−2.667+2.473j−2.674+2.506j−2.667−2.473j3.3673.393−0.018j−2.674−2.506j3.393+0.018j3.418); |M1|=0 M2=(1.0230.602−0.112j−0.528+0.409j0.602+0.112j0.977−1.144+0.205j−0.528−0.409j−1.144+0.205j5.473); |M2|=2.510 ALL AT ONCE

[0157] All n output error matrices are summed and the eigenvector corresponding to the smallest eigenvalue is determined.

[0158] Eigenvalues: (M1 + M2) = 1.146, 3.869, 13.173

[0159] Solution = (0.739 - 0.235j, 0.483 + 0.306j, 0.246 + 0.104j) ONE BY ONE

[0160] Only the acoustic task is solved, then everything else is done at once. This results in an exact solution to the acoustic task.

[0161] Eigenvalues: (M1) = 0, 0, 10.714 V1=(0.770−0.199j, 0.376+0.202j, 0.377+0.206j) V2=(0.097−0.071j, 0.765+0.010j, −0.632+0.0016j)

[0162] Since V1 and V2 both correspond to an eigenvalue equal to zero, a·V1 + b·V2 is also an eigenvector corresponding to an eigenvalue equal to zero, ie it is an exact solution to the acoustic problem.

[0163] The all-at-once minimization of the structural task is formed using a and b: M2'=(1.314−0.381+0.341j−0.381−0.341j0.093);|M2'|=5.098

[0164] M1' eigenvalues, 1.222 and 4.172

[0165] Eigenvector after scaling: (0.984 - 0.016j, 0.113 + 0.115j)

[0166] Now V1 and V2 are combined to determine the inputs: (0.984−0.016j)V1+(0.113+0.115j)V2=(0.776−0.207j,0.473+0.283j,0.290−0.124j)

[0167] Normalize and scale the result: (0.755 - 0.211j, -0.466 + 0.270j, 0.246 + 0.104j)

[0168] It should be noted that this is similar to the all-at-once approach, but not identical. Extending this algorithm to a frequency range yields a precise result for the acoustic task, although numerical rounding in the sequential algorithm accounts for the very low non-zero pressure.

[0169] As stated above, the two methods are not mutually exclusive, and the parallel method can be adopted at any point in the sequential method, especially when the goal is to complete the process. The sequential method is useful when the number of inputs does not exceed the number of outputs, especially when some outputs are more important than others. The important outputs are then solved exactly, and the best-fit solution is determined for the remaining ones.

[0170] In a setup where only the maximization of the outputs as a whole is of interest, there is no point in using the sequential algorithm.

[0171] In this way, the signal processor controller 108 of the flat panel loudspeaker controller 100 can be preconfigured by an electronic device such as a computer. In other words, this configuration is performed at the design stage before it is used to improve the phase angle between the signals as the overall output at the signal processor outputs.

[0172] Fig. shows an integrated module 200 with piezoelectric elements 204, or in other words, an array of addressable piezoelectric elements forming an actuator array component that can be part of a planar loudspeaker, in this example for use in a portable computer such as a tablet or laptop computer (not shown). For manufacturing thin portable computers, a direct drive using electrically active materials is a very attractive solution.

[0173] The module 200 of piezoelectric elements comprises an array of relatively small piezoelectric patches 204 (in this example 20 mm 2 ) with corresponding electrode connection to provide a small number of input channels. The example arrangement of patches in Fig. In this example, the patches are arranged in 3 rows of 5 columns. The inventors of the present patent application have realized that the degree of activation is directly proportional to the patch area and, especially at low frequencies, is almost independent of the aspect ratio or shape. The degree of activation is the amount of output or activity generated by the patch area—in this example, sound pressure. The proportionality of the area and the invariance of the shape can be determined through simulations.

[0174] Module 200 is an audio-only application of the direct drive to the rear of a portable computer. In this case, the module is intended to provide direct drive to a display with a diagonal length of approximately 300-350 mm. Fig. shows a simple example version of the back 206 of the portable device on which the module 200 is applied. It is made of 1 mm thick glass or aluminum. The back has a flat surface 208 with rectangular dimensions of 280 x 170 mm with 18 mm wide beveled edges 210. The overall external dimensions are 316 x 206 x 5 mm. A variant of the plate of the Fig. shows Fig. . The appearance of plate 220 of the Fig. is in most respects similar to the record of the Fig. , and like features are identified by like reference numerals. The plate 220 of the Fig. also includes ribs 222 to reinforce a 1 mm thick glass-filled polymer (PBT-GF30%) from which the plate is made (approximately equivalent to the thickness of 1.5 mm thick ABS (acrylonitrile butadiene styrene) plastic).

[0175] Fig. shows plate 206 of the Fig. with a pair of actuator assembly components or assemblies 200 of the Fig. (Like features are designated by like reference numerals throughout the figures). The piezoelectric elements 204 of each array are wired to provide three channels of five elements each. One array is located on one side of the plate, and the other module is located on the other side of the plate. Each array provides a single channel of a stereo speaker system. In this example, the two arrays are arranged as mirror images of each other, with the mirror line, which in this example is a single central rib 223, dividing the plate lengthwise.

[0176] A parameterized finite element model of the arrangement of the Fig. , which includes the plate 206, two arrays 200 of patches 204 described above, and outside air up to a radius of 250 mm, was constructed on a computer. The variables considered were the positioning of the patch arrays and the electrodes to be switched on. From this model, the axial pressure (response in the air at the selected distance of 250 mm from the arrays) on the driven side and the other (display) side was recorded. The difference between the two pressures is almost independent of any of the variables considered or of which version of the two plates described above was simulated.

[0177] Electrodes were switched on simultaneously and symmetrically (both arrangements simultaneously) in each row (of 5 patches) in each arrangement 200 of 204 patches (ie 5 x 2 patches = 10 patches at once) (row 1, row 2 and row 3 from the inside out, as in Fig. shown), which produces two pairs of frequency or impulse responses for each Fig. The best responses were obtained from a procedure described above which averaged the root mean square (rms) of each of the three responses for a normalized input energy (SMR max line of Fig. ). In the most sensitive arrangement, both arrangements are close to the center (row 1); this prevents any stereo separation. As in Fig. As shown, some arrangements result in a row of patches intersecting a node line, essentially making that row redundant. An air cavity with a 1 mm gap and a total volume of 117.5 cm 3 was added to the model, and the result is the Fig. In this configuration, the frequency of the lowest (middle ear) mode is shifted upwards, which affects the bass response of the system. The sound pressure level (SPL) on the driven side is measured at different air distances from the glass-filled polymer plate 220 in Fig. The distances are 23 mm (dashed line), 48 mm (dotted line), and 73 mm (solid line).

[0178] Fig. represents for three lines of patches 204 (line 1, line 2 and line 3 from the inside out on the plate 206 (as in Fig. shown)) the sound pressure levels versus frequency, with the values presented individually and (in an example method according to one aspect of the present invention) combined. The frequency or impulse response of the individual rows of patches is shown in Fig. represented by lines 252 (line 1), 254 (line 2) and 256 (line 3). The frequency response of the patches combined according to an example of the present invention is shown in Fig. the line SMR max 258 at 250 mm on the axis (spaced from the plate) and in Fig. at different distances from the plate, the dashed line 260 (23 mm from the plate), the dotted line 262 (48 mm from the plate) and the solid line 264 (73 mm from the plate). In all cases, it can be seen that from about 700 Hz (especially on the driven side) onwards, with some output to plate f0, the sensitivity increases considerably. Plate f0 is the lowest acoustically active mode of the plate. This marks the point in the frequency response where there is a significant increase in sensitivity. There may also be other lower frequency modes that lead to peaks in the sound power; if these are too isolated from plate f0 (e.g. because they originate from the actuator rather than the plate), there will be a gap in the response.

[0179] In the example of Fig. There are indications of plate modes at about 400 and 800 Hz. The isolated mode is in Fig. at about 160 Hz, but at about 280 Hz in Fig. . Fig. shows a gap with relatively low sound power, while the gap in Fig. filled, since the isolated resonance frequency of the plate is closer to f0. The range between f0 and 700 Hz is not so good, and is particularly weak if f0 is too low.

[0180] As in the electromagnetic example of the arrangement of the Fig. The optimal drive potentials do not all need to have the same polarity. Therefore, driving them all at the same voltage will always result in a lower SPL (assuming the same net input - ie, all at 1 / √3 V). In fact, the patches balance each other out at some frequencies, as in Fig. by the line labeled “same drive”. However, as can be seen from the line “SMR max” of the Fig. As shown, using the method according to an above example of the present invention, it is demonstrated that the back of a portable computer, e.g. a tablet or laptop computer, of this size can provide audio of sufficient level and sufficient bandwidth. In the method described above, a signal processor controller is associated with all of a plurality of signal processors, each signal processor associated with each input, each input associated with each actuator of the planar loudspeaker to be controlled, and each signal processor having an output to allow an electrical signal to control an actuator of the planar loudspeaker. The signal processor controller is preconfigured to improve the phase angle between the signals as an overall output at the outputs of the signal processors.

[0181] The degree of activation of this device is directly proportional to the total area of the patches. The positioning of the patches depends on the number and shape of the modes to be activated, the aspect ratio of the panel, and the number of sources.

[0182] Since drive potentials don't all need to have the same polarity, intelligent electrode utilization is required to ensure optimal performance. Furthermore, since performance is significantly more efficient at frequencies above 1 kHz, the number of driven patches can be reduced at these frequencies, thus saving energy. In fact, with other configurations, a significantly smaller number of actuators can be used in terms of plate design and assembly while still providing sufficient power.

[0183] Fig. shows the use of an example of the use of a back cover 300 of a portable computer or device, e.g., a tablet computer or an electronic book. The example device is approximately A5 in size and includes a polymer-based optoelectronic display, e.g., an OLED or electrophoretic display (not shown). The device includes a front lens (not shown) made of a cured polymer, a display stack (not shown), and a stiffening plate 302. For clarity, the internal air cavities and chassis are also omitted. The display is secured around the entire perimeter of the polymer lens, as well as at discrete bolt points on the stiffening plate, which are indicated by the small tabs 304 in the illustration of the Fig. shown, connected to the rest of the device.

[0184] In Fig. Also shown are two piezoelectric elements or patches 306, 308 of unequal size, directly connected to the back of the stiffening plate 302. The patch 306 near the center has planar dimensions that are 50% larger, and thus has 2.25 times the area of the offset patch 308. Thus, it also has 2.25 times the capacitance and activity.

[0185] The placement and size of this larger patch 306 makes it a more powerful source, especially at low frequencies, but this means that it also draws 2.25 times more current from the power supply than the smaller patch 308. From a power consumption perspective, it would therefore be better to use the smaller patch whenever possible, especially at higher frequencies.

[0186] Example frequency responses are the Fig. The frequency or impulse response of the small patch shows a dashed line 310, and the frequency response of the large patch shows a solid line 312. The frequency responses show that from about 600 Hz, the power is completely sufficient to begin reducing the electrical input power. The clumped response of the smaller patch indicates that its positioning is not optimal. COMBINATION STRATEGIES

[0187] Summated frequency or impulse responses of the two patches 306, 308 of the Fig. are the Fig. can be found.

[0188] The naive sum represented by a dashed line 350 works quite well above 600 Hz, but not below 600 Hz. The reason for this is shown Fig. . Fig. For the smaller patch 308, compares the amplitude (solid line 360) with the phase (dashed line 362). Fig. compares the amplitude (solid line 364) with the phase (dashed line 366) for the larger patch 306. The main reason why the navi sum does not work well below 600 Hz is that the patches must have the opposite polarity at low frequencies, as shown in Fig. can be seen from the 180° phase difference between the smaller and larger patches at 250 Hz.

[0189] As seen from the through line 370 of the Fig. shown and expressed as the optimal sum (voltage), the arrangement in which an electronic device preconfigures the flat panel speaker controller and then provides a preconfigured flat panel speaker controller according to embodiments of the present invention clearly provides a significantly better frequency response at frequencies below 600 Hz.

[0190] In practice, it is not necessary to implement the full-bandwidth transfer functions presented here. A reasonable approximation is to use simple filtering techniques to achieve summation that is better than naive summation. For example, a combination of all-pass and high-pass filters for the smaller patch can provide the low-frequency response. In other words, the signal processor controller can include or consist of a filter that is preconfigured to improve the overall phase shift between output signals. NORMALIZATION STRATEGIES

[0191] In the flat panel loudspeaker arrangement 300 of the Fig. Normalization strategies can be used to reduce or minimize energy requirements, as explained below.

[0192] The type of actuators or patches 306, 308 of the Fig. acts as a capacitive load. The energy stored on these capacitive loads C at a DC voltage V is CV22. The losses in the circuit are more likely due to the incoming and outgoing currents that I=CV2πf where f is the frequency. The losses are related to I 2 The so-called reactive power flow is given by IV.

[0193] We can normalize out input sensitivities to minimize any of these energy measurements. ΣV 2 = 1, as explained above, the optimization is based on equivalent input voltages, ΣVI = 1, the optimization is based on equivalent input energies. ΣI 2 = 1, the optimization is based on equivalent input currents.

[0194] In order to ensure low energy consumption, a flat panel loudspeaker control of the flat panel loudspeaker 300 of the Fig. be preconfigured to shift the balance of the contribution to the signal amplitude from the larger patch 306 to the smaller patch 308, since the smaller patch draws less current.

[0195] Embodiments of the present invention have been described. It should be understood that variations and modifications of the described embodiments are possible within the scope of the present invention.

Claims

[1] Flat panel loudspeaker control (100) for controlling a flat panel loudspeaker (101; 300) comprising a plurality of actuators, the flat panel loudspeaker control comprising: a plurality of electrical signal inputs (102), each input being assigned to a corresponding actuator of the planar loudspeaker (101; 300) to be controlled, a plurality of signal processors (104), each signal processor being associated with a respective input and having an output (106) for an electrical signal for controlling an actuator of the planar loudspeaker (101; 300), and each signal processor implementing a transfer function from its input to its output (106) on the basis of each actuator of the planar loudspeaker (101; 300) for a desired sound receiver (sic), and a signal processor controller (108) associated with each of the plurality of signal processors (104), the signal processor controller (108) being preconfigured to improve the phase angle between the signals as an overall output at the outputs (106) of the signal processors (104). [2] A flat panel loudspeaker controller (100) according to claim 1, wherein the signal processor controller (108) comprises a filter to be preconfigured to improve the phase angle between the signals as a total output at the outputs (106) of the signal processors (104). [3] Flat panel loudspeaker controller (100) according to claim 2, wherein the filter comprises a low-pass filter and / or an all-pass filter. [4] A flat panel loudspeaker controller (100) according to claim 3, wherein the low-pass filter passes signals having a frequency below a cutoff frequency of 500 Hz. [5] A flat panel loudspeaker controller (100) according to any preceding claim, wherein each signal processor comprises a digital signal processor. [6] A flat panel loudspeaker controller (100) according to any preceding claim, wherein the signal processor controller (108) comprises a digital signal processor to be preconfigured to improve the phase shift between the signals as an overall output at the outputs (106) of the signal processors (104). [7] A flat panel loudspeaker controller (100) according to any preceding claim, wherein signal processing is applied by the signal processor controller (108) to the electrical signal inputs (102) to achieve a maximum or near maximum total output at the outputs (106) at all frequencies. [8] A flat panel loudspeaker control (100) according to any one of the preceding claims, wherein the signal processing is applied by the signal processor control (108) to the electrical signal inputs (102) to achieve a minimum or near-minimal sound pressure in at least one predetermined spatial position. [9] A flat panel loudspeaker controller (100) according to any preceding claim, wherein the signal processor controller (108) comprises an equalizer to be preconfigured to improve the phase difference between the signals as a total output at the outputs (106) of the signal processors (104), the equalizer equalizing the input signals. [10] Flat panel loudspeaker control (100) according to one of the preceding claims, wherein the plurality of actuators comprises at least one piezoelectric actuator, such as a piezoelectric patch (204; 306, 308) and / or at least one coil-magnet actuator (116, 117) and / or a DM actuator. [11] Flat panel loudspeaker control (100) according to one of the preceding claims, wherein the plurality of actuators comprises an actuator arrangement (200). [12] A planar loudspeaker controller (100) according to any one of the preceding claims, wherein the sound receiver comprises a user's ear or a microphone. [13] Flat panel loudspeaker (101; 300) comprising a flat panel loudspeaker controller (100) according to one of the preceding claims. [14] An electronic device, e.g. a computer such as a tablet or a laptop, or a display device, e.g. an LCD display, comprising the planar loudspeaker (101; 300) according to claim 13. [15] Method for controlling a planar loudspeaker (101; 300) comprising a plurality of actuators, the method comprising: Inputting a plurality of electrical signal inputs (102), each input being assigned to a corresponding actuator of the planar loudspeaker (101; 300) to be controlled, a plurality of signal processors (104), each signal processor being associated with a respective input and having an output (106) for an electrical signal for controlling an actuator of the planar loudspeaker (101; 300), and each signal processor implementing a transfer function from its input to its output (106) on the basis of each actuator of the planar loudspeaker (101; 300) for a desired sound receiver (sic), and a signal processor controller (108) associated with each of the plurality of signal processors (104), wherein the signal processor controller (108) improves the phase angle between the signals as a total output at the outputs (106) of the signal processors (104) based on a preconfiguration. [16] An electronic device configured to control a signal processor (108) of a planar loudspeaker (101; 300) comprising a plurality of actuators, the electronic device being configured to perform the following functions: Inputting electrical signals into a plurality of electrical signal inputs (102), each input being assigned to a corresponding actuator of the planar loudspeaker (101; 300) to be controlled, Measuring a response of the planar loudspeaker (101; 300) to the electrical inputs as a whole and Using this response to configure a signal processor controller (108) associated with all of a plurality of signal processors (104), and when used to improve the overall phase shift between the signals output at the outputs (106) of the plurality of signal processors (104), each signal processor being associated with a respective input and having an output (106) for an electrical signal for controlling an actuator of the planar loudspeaker (101; 300), and each signal processor implementing a transfer function from its input to its output (106) on the basis of each actuator of the planar loudspeaker for a desired sound receiver (sic). [17] Electronic device according to claim 16, wherein the electrical input signals, actuators, flat panel loudspeakers (101; 300) and response are implemented virtually. [18] An electronic device according to claim 16 or 17, wherein the electrical input signals are in the form of a pulse and the response is in the form of an impulse response. [19] The electronic device of any of claims 16-18, wherein the electronic device is configured to use the response to configure the signal processor controller (108) by assessing differences between transfer functions of the signal processors (104). [20] A method for configuring a signal processor control (108) of a planar loudspeaker comprising a plurality of actuators, (101; 300), the method comprising: Inputting electrical signals into a plurality of electrical signal inputs (102), each input being associated with a corresponding actuator of the planar loudspeaker to be controlled, Measuring a response of the planar loudspeaker (101; 300) to the electrical inputs as a whole and Using the response to configure a signal processor controller (108) associated with all of a plurality of signal processors (104), and when used to improve the overall phase shift between the signals output at the outputs (106) of the plurality of signal processors (104), wherein each signal processor is associated with a corresponding input and has an output (106) for an electrical signal for controlling an actuator of the planar loudspeaker (101; 300), and each signal processor implements a transfer function from its input to its output (106) on the basis of each actuator of the planar loudspeaker (101; 300) for a desired sound receiver (sic). [21] A method according to claim 20, wherein the electrical input signals are in the form of a pulse and the response is in the form of an impulse response. [22] The method of claim 20 or 21, wherein using the response to configure the signal processor controller (108) comprises assessing differences between transfer functions of the signal processors (104). [23] An electronic device configured to collectively configure a signal processor controller (108) of a planar loudspeaker (101; 300) comprising a plurality of actuators, using a response of the planar loudspeaker to electrical inputs each associated with a corresponding actuator of the planar loudspeaker (101; 300), the signal processor controller (108) being associated with all of a plurality of signal processors (104) and, when used, being configured to improve the phase shift between the signals output at the outputs (106) of the plurality of signal processors (104), each signal processor being associated with a corresponding input and having an output (106) for an electrical signal for controlling an actuator of the planar loudspeaker, and each signal processor having a transfer function from its input to its output (106) based on each actuator of the planar loudspeaker (101;300) for a desired sound receiver (sic). [24] Computer program for carrying out the method according to one of claims 20-22. [25] A non-transitory computer-readable data carrier comprising instructions for carrying out the method according to any one of claims 20-22.

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