System and method for compensating for nonlinear behavior of an acoustic transducer based on magnetic flux

Abstract

In at least another embodiment, an audio amplifier system is provided, comprising a memory and an audio amplifier. The audio amplifier comprises the memory and is programmed to receive an audio input signal and generate a target current signal based on the audio input signal and a velocity of a diaphragm of a speaker. The audio amplifier is further programmed to generate a correction current signal based on at least the target current signal and a predicted position of a voice coil of the speaker, and to determine the predicted position of the voice coil of the speaker based on a flux density value. The flux density value corresponds to the product of the magnetic flux of the air gap of the voice coil in the speaker and the length of the voice coil wire in the speaker.

Description

[0001] CROSS-REFERENCE TO RELATED APPLICATIONS

[0002] This application may be related to U.S. Application Serial No. 16 / 402,805, entitled “SYSTEM AND METHOD FOR NON-LINEAR BEHAVIOR FOR ANACOUSTIC TRANSDUCER,” attorney docket number HARM0681PUS and filed on May 3, 2019. Technical Field

[0003] One or more aspects disclosed herein generally relate to a system and method for compensating for nonlinear behavior of an acoustic transducer. Aspects disclosed herein may provide an active sensorless nonlinear correction method and apparatus for a loudspeaker using a moving coil acoustic transducer, housing, and passive radiator. For example, a low machine instruction per second (MIPS) sensorless model and algorithm for correcting distortion caused by a diaphragm suspension and a voice coil motor as a function of voice coil position for enclosed, vented, and passive radiator designs driven by current and voltage sources is provided. These and other aspects will be discussed in greater detail herein. Background Art

[0004] French's PCT / US2018 / 052336 (the "'336 Application") provides an acoustic transducer. The acoustic transducer includes a controller configured to receive an input audio signal and generate a first reference signal indicative of an envelope of the input audio signal. The controller is further configured to provide a fixed coil signal to a fixed coil of the acoustic transducer based on the first reference signal and to measure a current through the fixed coil after providing the fixed coil signal to the fixed coil. The controller is further configured to generate a first output indicative of the current through the fixed coil and to determine a magnetic flux in an air gap of a magnetic material based on the first output. The controller is further configured to generate a voltage output for the moving coil, the voltage output being inversely proportional to the magnetic flux in the air gap. The voltage output provides an undistorted output corresponding to the input audio signal. Summary of the invention

[0005] In at least one embodiment, an audio amplifier system is provided, comprising a memory and an audio amplifier. The audio amplifier comprises the memory and is programmed to receive an audio input signal and generate a target current signal based on the audio input signal. The audio amplifier is configured to generate a first predicted position of a voice coil of a speaker, and to generate a first correction current signal based on the target current signal and the first predicted position of the voice coil. The audio amplifier is further configured to determine a pressure within a speaker housing based on at least the first predicted position of the voice coil, and to determine a position of a passive radiator based on at least the pressure within the speaker housing. The audio amplifier is further configured to determine a flux density value of the speaker, and to generate a second predicted position of the voice coil based on the flux density value and based on at least the pressure within the speaker, the position of the passive radiator, and the first correction current signal.

[0006] In at least another embodiment, an audio amplifier system is provided, comprising a memory and an audio amplifier. The audio amplifier comprises the memory and is programmed to receive an audio input signal and generate a target current signal based on the audio input signal and a velocity of a diaphragm of a speaker. The audio amplifier is further programmed to generate a correction current signal based on at least the target current signal and a predicted position of a voice coil of the speaker, and to determine the predicted position of the voice coil of the speaker based on a flux density value. The flux density value corresponds to the product of the magnetic flux of the air gap of the voice coil in the speaker and the length of the voice coil wire in the speaker.

[0007] In at least another embodiment, a computer program product is provided, embodied in a non-transitory computer readable medium, the computer program product being programmed to amplify an audio input signal. The computer program product includes instructions for receiving an audio input signal and generating a target current signal based on the audio input signal. The computer program product includes instructions for generating a first predicted position of a voice coil of a loudspeaker and generating a first correction current signal based on the target current signal and the first predicted position of the voice coil. The computer program product also includes instructions for determining a pressure within a loudspeaker housing based on at least the first predicted position of the voice coil and determining a position of a passive radiator based on at least the pressure within the loudspeaker housing. The computer program product also includes instructions for determining a flux density value for the loudspeaker and generating a second predicted position of the voice coil based on the flux density value and based on at least the pressure within the loudspeaker, the position of the passive radiator, and the first correction current signal. BRIEF DESCRIPTION OF THE DRAWINGS

[0008] Embodiments of the present disclosure are particularly pointed out in the appended claims. However, other features of the various embodiments will become more apparent and will be best understood by referring to the following detailed description in conjunction with the accompanying drawings, in which:

[0009] Figure 1 Generally depicts an example of a closed speaker system;

[0010] Figure 2 The general description includes various aspects of the transducer;

[0011] Figure 3 A general description includes various aspects of passive radiators;

[0012] Figure 4 A model generally showing the elements of a loudspeaker system associated with a transducer and a passive radiator;

[0013] Figure 5 Schematic diagram of a system for estimating Kms(x) and Rms(x) in a speaker system according to one embodiment;

[0014] Figure 6 An amplifier system for correcting distortion in a speaker system is generally shown according to one embodiment;

[0015] Figure 7 According to one embodiment Figure 6 The amplifier system and further includes a core correction block;

[0016] Figure 8 depicts a correction system used as a voltage source to drive a voice coil according to one embodiment;

[0017] Fig. 9 A system for executing a model for providing Kms and Rms according to one embodiment is shown;

[0018] Fig.10 A system that also implements a model for providing Kms and Rms is shown according to one embodiment;

[0019] Fig.11 Depicting an example of an apparatus for measuring the values ​​of Kms and Rms according to one embodiment;

[0020] Fig.12 depicts corresponding values ​​of Rms and Kms as impedance varies according to one embodiment;

[0021] Fig.13 Generally illustrating a system for providing Kms and Rms according to one embodiment;

[0022] Fig.14Generally illustrating one implementation of an acoustic transducer device according to one embodiment;

[0023] Fig.15 Generally illustrating one implementation for calculating magnetic flux density (BL) according to one embodiment;

[0024] Fig.16 Another implementation for calculating magnetic flux density (BL) according to one embodiment is generally shown;

[0025] Fig.17 Another implementation of an acoustic transducer device according to an embodiment is generally shown;

[0026] Fig.18 Another implementation of an acoustic transducer device according to an embodiment is generally shown;

[0027] Fig.19 Another implementation of an acoustic transducer device according to an embodiment is generally shown;

[0028] Fig. 20 generally depicts another implementation of an acoustic transducer device according to an embodiment;

[0029] Fig.21 generally depicts a method performed by an audio amplifier system according to one embodiment;

[0030] Fig. 22 generally depicts a method performed by an audio amplifier system according to another embodiment;

[0031] Fig.23 generally depicts a method performed by an audio amplifier system according to another embodiment; and

[0032] Fig.24 A method performed by an audio amplifier system according to another embodiment is generally described. DETAILED DESCRIPTION

[0033] As required, detailed embodiments of the present invention are disclosed herein; however, it should be understood that the disclosed embodiments are merely examples of the present invention that may be embodied in different and alternative forms. The drawings are not necessarily drawn to scale; some features may be exaggerated or reduced to show details of particular components. Therefore, the specific structural details and functional details disclosed herein should not be interpreted as limiting, but merely as a representative basis for teaching those skilled in the art to use the present invention in different ways.

[0034] It should be recognized that the controllers disclosed herein may include various microprocessors, integrated circuits, memory devices (e.g., flash memory, random access memory (RAM), read-only memory (ROM), electrically programmable read-only memory (EPROM), electrically erasable programmable read-only memory (EEPROM), or other suitable variants of the above memories) and software, each of which cooperates with each other to perform one or more operations disclosed herein. In addition, such controllers as disclosed utilize one or more microprocessors to execute a computer program embodied in a non-transitory computer-readable medium, the computer program being programmed to perform any number of functions as disclosed. In addition, one or more controllers as provided herein include a housing and various numbers of microprocessors, integrated circuits, and memory devices (e.g., flash memory, random access memory (RAM), read-only memory (ROM), electrically programmable read-only memory (EPROM), electrically erasable programmable read-only memory (EEPROM)) positioned within the housing. One or more controllers as disclosed also include hardware-based inputs and outputs, which are used to receive data from and transmit data to other hardware-based devices as discussed herein, respectively.

[0035] When dynamic transducers (or dynamic speakers) increase their acoustic output, such transducers increase their distortion. This fundamental relationship determines the size, weight, cost, and inefficiency of the transducer, all of which are undesirable. This is especially true for transducers used in automotive applications where all of these performance issues are significant. At the same time, there is a growing need for higher output, lower distortion systems that can achieve or provide the desired active noise cancellation (ANC), engine order noise cancellation (EOC), independent sound zones (ISZ), and echo cancellation for voice recognition.

[0036] Therefore, there are sensorless methods, such as those described by Klippel, which attempt to minimize the distortion of the transducer through signal processing, which (if used properly) in turn can enable transducer designers to achieve smaller, lighter, lower cost or more efficient solutions based on the desired trade-offs. However, these methods can be computationally expensive (e.g., 100 million instructions per second (MIPS) or more), especially in multi-channel applications (such as those found in automobiles). In addition, these methods typically require embedded microcontrollers and digital signal processors (DSPs). Therefore, there is a need for low MIPs algorithms (e.g., which provide relatively low processing requirements) and low hardware cost methods for nonlinear distortion correction as provided herein. In addition, the solution should be compatible with automotive hardware that requires relatively low processing requirements.

[0037] In general, at a basic level, once the control or correction of nonlinearities in the transducer is actively controlled and or corrected, transducer and system designers have flexibility in the tradeoffs that may be necessary in the loudspeaker. This can improve size, weight, cost, and efficiency depending on the design goals. For example, the embodiments disclosed herein can provide better control of the displacement or offset of the transducer and the voice coil current, which can allow the transducer to be driven closer to its limits and therefore provide greater output. In addition, the embodiments disclosed herein provide enhanced control of the nonlinear performance of the transducer and can enhance the performance of acoustic algorithms that depend on the linearity or response of the transducer, such as ANC, RNC, EOC, ISZ, echo cancellation, etc.

[0038] The embodiments disclosed herein can be: (i) robust and inherently predictable in terms of stability, repeatability, and auditability (i.e., not a black box), (ii) computationally simple, with low to very low MIPs, sensorless, (iii) adaptive in the case of simple current sensing, and (iv) algorithmically simplified and operable in a DSP environment that may not require an accompanying embedded controller in order to be adaptive.

[0039] Figure 1 An example of a closed speaker system 100 is generally depicted according to one embodiment. The system 100 includes a housing 101 that generally includes a speaker 102 (or transducer) (e.g., an active speaker or main driver) and a passive radiator 104 (or a passive radiating cone that does not receive electrical energy in the form of an audio input signal). The housing 101 generally represents a common speaker housing for transmitting audio signals, and aspects related to the transducer 102 and the passive radiator 104 will be discussed in more detail below.

[0040] Figure 2 The general depiction includes various aspects of the transducer 102. For example, the transducer 102 often includes a cone (or diaphragm) 110 and a voice coil 112. A surround (or suspension) 114 is attached at the end of the diaphragm 110. A bobbin 116 surrounds the voice coil 112 and is positioned within an air gap 118. An external magnet (or magnets) 120 surrounds the air gap 118 and at least a portion of the voice coil 112 and bobbin 116. A centering blade 122 surrounds a portion of the bobbin 116.

[0041] Typically, an audio input signal corresponding to audio data is provided to the voice coil 112. The voice coil 112 and the magnet 120 are magnetically coupled to each other, and the audio input signal causes the diaphragm 110 to move linearly on a vertical axis based on the polarity of the audio input signal. The diaphragm 110 is typically flexible and deflects in two directions on the vertical axis in response to the magnetic field transmitted between the voice coil 112 and the magnet 120. The skeleton 116 is attached to the diaphragm 110 and undergoes similar displacement (or movement along the vertical axis) as the diaphragm 110. Due to the linear displacement of the diaphragm 110, the transducer (or speaker) 100 transmits the audio input signal to a room or other environment for consumption by a user. The centering blade 122 is typically configured to prevent the diaphragm 110 from moving horizontally during the linear movement of the diaphragm 110 in the vertical direction or axis.

[0042] Figure 3 The general depiction includes various aspects of the passive radiator 104. In general, the passive radiator 104 may include all of the mentioned components that make up the transducer 102, except for the voice coil 112 and the magnet 120. The passive radiator 104 may use the sound trapped within the housing 101 to generate resonance to provide low frequencies (i.e., bass). The passive radiator 104 may generate frequencies based on the mass and elasticity (or compliance) of the air within the housing 101. The passive radiator 104 may be tuned to the housing 101 by changing its total diaphragm mass (including the weight of the diaphragm 110 or the cone). When the transducer 102 generates air pressure due to the linear displacement of the diaphragm 110, this air pressure causes the passive radiator 104 to move.

[0043] Figure 4 Models of elements associated with transducer 102 and passive radiator 104 in speaker system 100 are shown. In general, by mathematically modeling the behavior of voice coil 112 (or moving coil of transducer 102) and other mechanical elements in speaker system 100, nonlinear behavior can be calculated and corrected for using amplifiers and real-time signal processing. These aspects will be discussed in more detail herein.

[0044] There are many ways to model a loudspeaker system. However, if the physical elements of the system are well understood up front, as is the case here, a model that fits those elements can be computationally simplest and most easily tuned. Aspects disclosed herein attempt to model the physical elements (e.g., transducer 102 and passive radiator 104) and their interactions in a loudspeaker system 100 in a way that is directly computable, adaptively tuneable, and corrects when the elements behave in a nonlinear manner.

[0045] There are generally four subsystems in the speaker system 100: (1) a transducer 102 (which transduces an electrical signal from an amplifier (not shown) into a mechanical output (not shown) (e.g., the mechanical output can be considered to be motion, which in turn transduces the mechanical output into an acoustic signal); (2) a passive radiator 104 (which resonates with the housing 101 and transducer 102 to produce an acoustic output at a lower frequency); (3) the housing 101, which couples the passive radiator 104 to the transducer 102 (via pressure) and isolates the back pressure of both the passive radiator 104 and the transducer 102 from the front pressure; and (4) an amplifier and signal processing (not shown). Two simplified subsets of the speaker system 100 may also be used, such as a vented system that replaces the passive radiator 104 with an acoustic mass created using a port in the housing 101, and a closed box system with a simple sealed housing without vents or passive radiators 104. Figure 4 Three mechanical subsystems are shown and are similar to a two-body resonant system.

[0046] In general, the mechanical elements of the transducer 102 can be modeled as a spring with a stiffness (e.g., Kms_TD), damping (e.g., Rms_TD), and a moving mass (e.g., M_TD). M_TD corresponds to the mass of all moving parts (including the air coupled to the diaphragm 110). Rms_TD corresponds to the friction losses of the combined surround 114 and centering support 122. Kms_TD corresponds to the spring stiffness of the combined surround 114 and centering support 122. In a similar manner, the passive radiator 104 can be modeled as a stiffness (e.g., Kms_PR), damping (e.g., Rms_PR), and a moving mass (e.g., M_PR). The transducer 102 and the passive radiator 104 can be considered as two bodies of the system 100. The force of the coupled body can be modeled by the pressure in the housing 101 (e.g., relative to the ambient pressure outside the housing 101) multiplied by the surface area of ​​the diaphragm 110 of the transducer 102 (e.g., Sd_TD) and the surface area of ​​the diaphragm 110 of the passive radiator 104. The compressibility of the air in the housing 101 can be modeled as a spring with a stiffness of kappa "κ" (i.e., the adiabatic index of air, approximately 1.4) multiplied by the box pressure.

[0047] In the case of the voice coil 112 (or moving coil of the transducer 102 ), the driving force F_1 may be modeled by the magnetic field strength (eg, “B”) in the air gap 118 multiplied by the length “L” of the conductor in the field multiplied by the current in the conductor (eg, the voice coil 112 ).

[0048] F 1 (t) = B·L·i vc (t) = BL·i Equation (1)

[0049] Reference frame x 1(t) is defined with respect to the position of the diaphragm 110 of the transducer 102. Similarly, the reference frame x 2 (t) is defined for the position of the diaphragm 110 of the passive radiator 104. 1 The positive direction of (t) is defined as moving into the housing 101, and x 2 The positive direction of (t) is defined as moving out of the housing 101.

[0050] The force of a moving mass is the mass times the acceleration, the force of a spring is equal to the distance from rest times the spring stiffness, and the force of friction (or damping) is the velocity times the friction.

[0051] The force on the moving mass of transducer 102 (eg, MmsTD) may be represented by the following equation:

[0052]

[0053] where x 1 (t) is shown as x 1 .

[0054] In a similar manner, the force on the moving mass of the passive radiator 104 may be represented by the following equation:

[0055]

[0056] where x 2 (t) is shown as x 2 .

[0057] Next, it may be necessary to calculate the pressure "p" based on the position of the diaphragm 110 of the transducer 102 and the diaphragm 110 of the passive radiator 104. This may be accomplished by first calculating the volume change of the housing 101 (e.g., Vol_1), which in turn may be the volume of the housing 101 (e.g., Vol_0) minus the volume occupied by the displacement of the diaphragm 110 of the transducer 102 and the passive radiator 104 from the rest position. It is known that the volume of air is proportional to the pressure, and therefore:

[0058] Vol(x 1 , x 2 )=Vol 0 +(s D_TD ·x 1 -s D_PR ·x 2 ) Equation (4)

[0059] Next, by relating the relative pressure "p" in the housing to the relative volume and the pressure p_amb outside the housing (of the surroundings), the new pressure caused by the volume change can be calculated by the following equation:

[0060]

[0061] Note that in the free body diagram (i.e. Figure 4 The “p” in equation (5) is p(x1,x2).

[0062] If Vol_0 is allowed to be the volume of housing 101 when diaphragms 110 (of both transducer 102 and passive radiator 104) are at rest, the change in pressure relative to the ambient pressure can be shown by Equation 6 as shown below.

[0063] By combining equations (4) and (5) to calculate the pressure in housing 101 relative to the surrounding environment based on X1 and X2, the following equation is obtained:

[0064]

[0065] This system of ordinary differential equations may then describe the motion of the diaphragm 110 (ie, the diaphragm 110 of the transducer 102 and the passive radiator 104) given a driving force from the voice coil 112. However, this does not yet take into account non-linear behavior.

[0066] Due to the shape of the magnetic field near the voice coil 112, BL is a nonlinear function of the position X1 of the diaphragm 110 of the loudspeaker 102. There are several ways to model this aspect, but a simple method can use an nth-order polynomial. For example, the following equation can express BL as a function of the position normalized to the rest position multiplied by the nominal value at the rest position:

[0067] BL=(cBL 4 ·x 4 +cBL 3 ·x 3 +cBL 2 ·x 2 +cBL 1 ·x+1)·BL (0) Equation (7)

[0068] Although equation (7) shows a 4th order polynomial, it is recognized that an nth order polynomial can be implemented for equation (7). Due to the physical properties of the suspension of the diaphragm 110, Kms and Rms are nonlinear functions of the position X1. As with BL, Rms and Kms can be expressed as polynomials. The polynomial has been decomposed into two parts, such as a normalized part and a scalar part corresponding to the rest position when X1=0. The benefits of doing so will become clear in the following refinement:

[0069] Kms=(cK 4 ·x 4 +cK 3 ·x3 +cK 2 ·x 2 +cK 1 ·x+1)·Kms(0) Equation (8)

[0070] Rms=(cR 4 ·x 4 +cR 3 ·x 3 +cR 2 ·x 2 +cR 1 ·x+1)·Rms(0) Equation (9)

[0071] From Figure 5 Equations (8) and (9) are shown in terms of the signal flow through the first normalization circuit 130, the second normalization circuit 132, the first multiplier circuit 134, and the second multiplier circuit 136. Recognizing that cR as depicted in the brackets of equations (8) and (9) 4 .x 4 Etc. correspond to the first normalization circuit 130 and the second normalization circuit 132, respectively. Each of the first normalization circuit 130 and the second normalization circuit 132 generally includes hardware and software to perform the calculations required by equations (8) and (9).

[0072] In the case of Rms, it can also be a function of the velocity of the diaphragm 110, which can also be modeled as a multi-top formula, for example:

[0073] Rms=((cV 2 ·speed 2 +cv 1 ·Speed+1)·Rms(x) Equation (10)

[0074] In equation (10), Rms(x) represents Rms of equation (9)

[0075] These equations can then be solved using a numerical method, such as Euler's method, where the equations are iterated at small time steps (small relative to the rate of change of position of any variable in the system 100). In particular, solving the set of equations 1-10 will provide the velocity of the diaphragm 110. This will be described in more detail below.

[0076] Correction by current source

[0077] Now that models have been developed for estimating the position and velocity of the transducer 102 and the diaphragm 110 of the passive radiator 104, these aspects can be inserted into the system (or audio amplifier system) 150 to correct for distortion (see Figure 6). The system 150 may be implemented as a current source amplifier (or audio amplifier) ​​and typically includes an equalization block 152, a core correction block 154, and a transducer prediction model block 156. The computationally simplest approach is to use a current source 158 to drive the voice coil 112. Because of the nature of the current source 158, the system 150 cancels out the effects of resistance and inductance in the voice coil 112 on the current and so these may be cancelled. By definition, the current source 158 feeds the desired current regardless of the load. In this approach, it may only be necessary to determine the correction current for the voice coil 112.

[0078] The equalization block 152 generates a current target (or I_target) corresponding to the desired current based on the audio input signal. The transducer model block 160 is typically fed an input current I_vc (or I_correction), which represents the current of the voice coil 112 and is generated by the amplifier 150 in response to at least the target current (i.e., I_target). The transducer prediction model block 156 includes a combination of hardware and software and calculates the position X1 of the diaphragm 110 of the speaker 102 (or the predicted position of the voice coil 112) according to equations 2, 3, 6, 7, 8, 9, and 10. The system 150 provides I_correction to the voice coil 112 to move the voice coil 112 to the predicted position of X1 as determined by the transducer prediction model block 156. The transducer prediction model block 156 includes a transducer model block 160, a pressure model block 162, and a passive radiator model block 164). The transducer model block 160 implements equations 2, 7, 8, 9, and 10. The stress model block 162 generally implements Equation 6, and the passive radiator model block 164 generally implements Equation 3. Given Kms_TD(X1), BL(x), and a target current (I_target from the equalization block 152) from their respective polynomials, a correction current (e.g., I_current) that compensates for nonlinearities in Kms_TD(x) and BL(x) may be calculated as follows:

[0079]

[0080] In general, if BL(x) is less than BL(0), the target current may be increased proportionally and by an amount that offsets the force error due to the change in spring stiffness. However, in such a system, the frequency response may be incorrect because the electrical damping provided by the resistance of the voice coil 112 may be cancelled by the amplifier 150 (or current source). This aspect may be compensated by using a fixed equalization filter in the equalization block 152. Figure 7 express Figure 6 The amplifier 150 also includes a core correction block 155, which can be improved in a later implementation.

[0081] Correction by voltage source

[0082] Figure 8 An audio amplifier system 180 is depicted that is used as a voltage source to drive the voice coil 112. The system 180 includes a current transformation block 182, an adaptation block 184, and a voltage transformation block 186. The system 180 provides a correction voltage to the voice coil 112 of the transducer in response to an audio input signal. The adaptation block 184 includes a core correction block 190 and a transducer prediction model block 156. Generally, the system 180 converts a target voltage (from an equalization block (the target voltage is generated based on the audio input signal)) to a target current (i.e., I_target)) through the current transformation block 182. The core correction block 190 corrects the target current to generate a correction current (i.e., I_corrected). The voltage transformation block 186 converts I_corrected to a correction voltage (i.e., V_corrected) for driving the voice coil 112. A voltage source amplifier (not shown) applies V_corrected to the voice coil 112. The system 180 ignores the effects of the inductance of the voice coil 112, which generally works if the correction is for the lower frequencies of the system 180. This may be valid because most of the movement and nonlinearity occurs at low frequencies.

[0083] In addition to the position X1 of the diaphragm 110 (see the output from the transducer prediction model block 156), the system 180 also uses the predicted velocity of the diaphragm. The current transformation block 182 uses the velocity of the diaphragm 110 to convert the audio signal (which is proportional to the voltage) into a target current I_target, and transmits the target current I_target to the core correction block 190. The voltage transformation block 186 also converts I_corrected into a signal proportional to the voltage to be applied to the voice coil 112. The transducer prediction model block 156 also provides the predicted BL (or the predicted magnetic flux X and the length of the air gap 118). In order to convert I_corrected to V_corrected according to equation 13 set forth below, the voltage transformation block 186 also needs the predicted BL.

[0084] Typically, the target voltage (i.e., the input into the current transformation block 182) must be converted to I_target for use in the transducer prediction model block 156. For example, the movement of the voice coil 112 introduces a current that produces a voltage proportional to the velocity times "B" times "L", where "L" corresponds to the length of the air gap; this can be referred to as the back EMF of the voice coil 112. This provides a voltage that is subtracted from the voltage applied to the voice coil 112 (i.e., V_corrected) to balance the resistance across the voice coil resistance (e.g., Rvc). The linear target current (i.e., I_corrected) that would match the voice coil current if BL(x) were linear can then be calculated by the following equation:

[0085]

[0086] Once the target current is corrected as similarly pointed out above, it is necessary to convert the target current back to the corrected voltage (i.e., Vcorrected). Based on the same relationship, this can be achieved by the following equation:

[0087] V corrected =I corrected ·Rvc Avg +BL·Speed ​​equation (13)

[0088] Changes in voice coil DC resistance (Rvc)

[0089] In a simple approach, it may be assumed that the resistance of the voice coil 112 is constant. Assuming the resistance of the voice coil 112 is constant, Rvc in equation (13) is Avg Will be set to Rvc nominal . Typically, voice coils are formed from copper or aluminum. These materials may experience changes in resistance as their corresponding temperature changes. Therefore, in order to improve the voltage source implementation of system 180, a thermal model may be used to estimate the temperature rise of voice coil 112 and thereby calculate the temperature corrected resistance of voice coil 112. Because the current is predicted as I_corrected, the power in voice coil 112 may be obtained. There are several thermal models that may be used based on accuracy. The simplest model may be an RC model, where R represents the thermal resistance of voice coil 112 to the surrounding environment, and C represents the specific heat capacity of voice coil 112. The RC model may also be solved iteratively using Euler's method.

[0090] An example of the Euler method for iteratively solving a system of equations is directly described below. By looping through the code of the algorithm shown below over and over again, the algorithm solves various systems of equations in a small time step so that the equation can be moved in a small time step so as to be considered and viewed as linear. For example, a time step of 200uS (for a sampling rate of 5kHz) can fully model a typical loudspeaker. This model may require downsampling or sampling at the input (e.g., an audio input that can be, for example, 48KHz) and a Vcorrected and Icorrected output that can be 48KHz), and upsampling using an interpolation filter at the output (e.g., and a Vcorrected and Icorrected output that can be 48KHz). Using this method, a fixed-point complete implementation requires about 5-6MIPS / channel for a complete passive radiator system, and requires at least 1-2MIPS for a closed box system.

[0091] * /

[0092] / / Solve for transducer motion:

[0093] / / dt is defined as the small time step for sampling the system

[0094] X1=X1+Velocity_TD*dt;

[0095] Force_damping_TD=-Velocity_TD*Rms(X1)_TD;

[0096] Force_spring_TD=-X1*Kms(X1)_TD

[0097] Force_pressure_TD=-k*pressure*Sd_TD;

[0098] Force_motor=BL(X1)*Ivc_corrected;

[0099] Force_net_TD=Force_damping_TD+Force_spring_TD+Force_pressure_TD+Force_motor;

[0100] Velocity_TD=Velocity_TD+Force_net_TD / M_TD*dt;

[0101] / / Solve the motion of passive radiation 104:

[0102] Force_damping_PR=-Velocity_PR*Rms(X2,Velocity_PR)PR;

[0103] Force_spring_PR=-X2*Kms(X2)_PR;

[0104] Force_pressure_PR=k*pressure*Sd_PR;

[0105] Force_net_PR=Force_damping_PR+Force_spring_PR+Force_pressure_PR;

[0106] Velocity_PR=Velocity_PR+Force_net_PR / M_PR*dt;

[0107] X2 = X2 + Velocity_PR*dt;

[0108] / / Solve for the pressure change of shell 101:

[0109] pressure=p_0*(Sd_TD*X1-Sd_PR*X2) / (Vb+Sd*X1+Sd_PR*X2);

[0110] / / Solve for the correction current of voice coil 112:

[0111] Ivc_corrected=Ivc_target*BL(0) / BL(X1)+(Kms(X1)-Kms(0)*X1 / BL(X1);

[0112] / / For the voltage source algorithm, add the following C code:

[0113] / / Solve Ivc_target

[0114] Ivc_target=(EQ_out–Velocity_TD*BL(X1)) / Rvoice_coil;

[0115] / / Solve for the correction voltage of voice coil 112:

[0116] V_voicecoil=Ivc_corrected*Rvoice_coil+BL(X1)*Velocity_TD.

[0117] Changes in Kms and Rms as a result of exercise history

[0118] The model also assumes that Kms and Rms during motion are defined by a polynomial. In reality, these parameters may vary with the "history" of the movement. For example, when the diaphragm 110 moves with significant velocity and displacement, the suspension 114 of the diaphragm 110 may soften. This may change both Rms and Kms.

[0119] As an improvement, estimates of the time-varying values ​​of Rms(0) and Kms(0) can be used to scale the values ​​of Kms and Rms. Since the polynomials of Kms(x) and Rms(x) are normalized to a stationary position, the time-varying parameters can be directly multiplied by the normalized position-varying parameters to determine more accurate Kms and Rms.

[0120] The softening and stiffening of the suspension 114 of the diaphragm 110 as a function of position can be predicted as a time-varying average value that can be modeled as an exponentially decaying sum, where the input used for the average corresponds to the steady-state values ​​of Kms and Rms that would result if the amplitude of motion was applied infinitely. This steady-state value of Kms can be expressed as a polynomial equation (14) for the envelope of the varying position.

[0121] Kms steadystate =a1 ·|x|+a 2 Equation (14)

[0122] Exponential decay can take the form of the following equation.

[0123]

[0124] Then, the average Kms (or Kms) can be calculated by multiplying equation (15) with equation (14). Avg ). This average Kms will then replace Kms(0) in equation (8) to provide:

[0125] Kms=(cK 4 ·x 4 +cK 3 ·x 3 +cK 2 ·x 2 +cK 1 ·x+1)·Kms Avg Equation (16)

[0126] The same form of equation can be used for the Rms steady state

[0127] Rm steadystate =b 1 |x|+b 2 Equation (17) Steady-state Rms

[0128] As with Kms, steady-state Rms can be related to the amplitude of motion using equations (15) and (17). The average Rms can then be calculated by multiplying equation (15) by equation (17). This average Rms will then replace Rms(0) in equation (9) to provide:

[0129] Rms=(cR 4 ·x 4 +cR 3 ·x 3 +cR 2 ·x 2 +cR 1 ·x+1)·Rms Avg Equation (18)

[0130] Kms as stated in Equations 15 and 16 Avg and Rms Avg The history of the predicted position of the voice coil 112 is taken into account by averaging X1 over the recent history.

[0131] Fig. 9A system 200 is shown that executes a model for providing Kms and Rms. The system 200 may be part of the transducer prediction model block 156. The system 200 generally includes a controller 202 to execute a model for providing Kms and Rms. The system 200 also includes a conversion block 204, a Kms normalization block 206 and an RMS normalization block 208, a first multiplier circuit 210 and a second multiplier circuit 212. The conversion block 204 is generally configured to convert the history of X1 into an average value of Kms (or KMS avg ) (eg, an average spring model signal). The conversion block 204 is generally configured to convert the history of X1 into an average value of Rms (or Rms avg ) (e.g., an average damping model signal). The system 200 provides Kms as represented by Equation 16 above. The system 200 also provides Rms as represented by Equation 18 above. The Kms normalization block 206 (or spring normalization block) is programmed to normalize the spring stiffness of the speaker 102 to a rest position corresponding to setting the predicted position of the voice coil 112 to zero. The Rms normalization block (or damping normalization block) 208 is programmed to normalize the damping of the speaker 102 to a rest position (e.g., setting the predicted position of the voice coil 112 to zero).

[0132] Fig.10A system 250 is generally shown that also implements a model for providing Kms and Rms. The system 250 includes a conversion block 204, a Kms normalization block 206, an RMS normalization block 208, a first multiplier circuit 210, and a second multiplier circuit 212. The conversion block 204 generally includes an envelope block 252, an average block 254, a Kms steady-state block 256, an Rms steady-state block 258, a Kms exponential decay block 260, an Rms exponential decay block 262, a Kms selection circuit 264, and an RMS selection circuit 266. The system 250 provides, but is not limited to, improved approximations of Kms_avg and Rms_avg by using different time constants as set forth in equation (15). The improved Kms_avg and Rms_avg can be based on whether the suspension 114 of the diaphragm 110 is softening or hardening. In other words, the system 250 takes into account whether the amount of movement of the diaphragm 110 is increasing or decreasing compared to the most recent movement. This aspect may be performed using a selection operation that compares Kms_avg or Rms_avg to a predicted steady-state Kms or Rms. If the average value is less than the steady-state value, the suspension 114 of the diaphragm 110 may be softened, and a softening time constant may be selected. Otherwise, if the average value is greater than the steady-state value, the suspension 114 of the diaphragm 110 may be stiffened, and then a stiffening time constant may be selected. This will be described in more detail below. Typically, the conversion block 204 is configured to generate and store any number of predicted positions (e.g., X1) of the voice coil 112 to provide a history of the generated predicted positions of the voice coil 112. For example, the conversion block 204 is configured to perform integration, peak detection, and / or averaging over a period of time to determine a history of the generated predicted positions of the voice coil 112. The system 250 may be such as Figure 6 A portion of the transducer model block 160 is illustrated, which is a portion of the transducer prediction model block 156. It is appreciated that the system 150 includes a memory (not shown) to store a history of the generated predicted positions of the voice coil 112.

[0133] At the input, the envelope block 252 receives a position signal (e.g., X1) and provides an envelope of the position signal. The average block 254 applies peak hold to the envelope generated by the envelope block 252. The output of the average block 254 represents the short-term average motion level, and therefore represents the softening possibility of the motion of the suspension 114. The Kms steady-state block 256 provides a steady-state Kms value (e.g., predicted steady-state), and the Rms steady-state block 258 provides a steady-state Rms value (e.g., predicted steady-state). Each of the Kms selection circuit 264 and the Rms selection circuit 266 includes a first-order or high-order polynomial to convert the average motion into a steady-state amount of suspension softening in response to the steady-state Kms value and the steady-state Rms value, respectively. The Kms exponential decay block 260 and the Rms exponential decay block 262 receive the steady-state Kms value and the steady-state Rms value, respectively, to model the time-dependent aspects of the softening behavior of the suspension 114 of the diaphragm 110.

[0134] As an improvement, different exponential decay time constants may be better fitted compared to the actual behavior of the suspension 114. In particular, the suspension 114 may soften faster than the stiffening of the suspension 114 of the diaphragm 110 (when the motion is reduced or stopped). To model this aspect, the Kms selection circuit 264 compares the steady-state predicted softness (or steady-state Kms value) with the modeled Kms average value, and if the steady-state Kms value (or predicted softness) is softer than the current value predicted by the exponential decay (or the modeled Kms average value), the softening τ is used in the exponential decay (or it is applied to the Kms exponential decay block 260). On the other hand, if the steady-state Kms value (or predicted softness) is stiffer than the current value predicted by the exponential decay (or the modeled Kms average value), the stiffening τ is used in the exponential decay (or it is applied to the Kms exponential decay block 26). Similar aspects may apply to the steady-state Rms value (or predicted softness), the modeled Kms average, the comparison of the steady-state Rms value and the modeled Kms average, and the softening τ and hardening τ applied by the Rms selection circuit 266 to the Rms exponential decay block 262 .

[0135] Provides adaptive Kms and Rms

[0136] In some cases, the measurement may be performed in real time while music or other audio is playing, such as in conjunction with system 180 (e.g., in conjunction with Figure 8 The current of the voice coil 112 as used in conjunction with the system 150 (e.g., the system 150 used as a current source to drive the voice coil 112) can be measured in real time while music or other audio is playing. If this is performed, there are other implementations to measure the values ​​of Kms and Rms or the average values ​​of Kms and Rms over time. Fig.11An example of such an implementation (or device) 300 for measuring the values ​​of Kms and Rms according to one embodiment is depicted. The device 300 includes a first bandpass filter 302, a second bandpass filter 304, a first low pass filter 306, a second bandpass filter 308, a divider circuit 310, a first polynomial block 312, and a second polynomial block 314. The first bandpass filter 302 may receive a signal corresponding to a measured voltage of the voice coil 112. The second bandpass filter 304 may receive a signal corresponding to a measured current across the voice coil 112. The first bandpass filter 302 and the second bandpass filter 304 bandpass filter the voltage and current at a predetermined frequency. The first low pass filter 306 and the second low pass filter 308 convert the instantaneous values ​​of the measured Vvc (e.g., the measured voltage of the voice coil 112) and the measured Ivc (e.g., the measured current of the voice coil 112), respectively, and convert them into corresponding rms values ​​of the average voltage (e.g., Vvc_avg) of the voice coil 112 and the average current (e.g., Ivc_avg) of the voice coil 112. The divider circuit 310 divides Vvc_avg by Ivc_avg to provide an impedance (e.g., Zavg).

[0137] In one approach, since the impedance presented by the voice coil 112 at any given frequency is the rms voltage at that frequency divided by the rms current at that frequency (by bandpass filtering the current and voltage at a particular frequency), converting the filtered results to rms values ​​and dividing them can provide the average impedance presented by the voice coil 112 at the given frequency. In general, the first polynomial block 312 and the second polynomial block 314 are configured to fit curves of Kms(z) and Rms(z), such as Fig.12 As shown. It is recognized that, with the help of equations 2 and 3, Kms and Rms are as follows Figure 6 As part of the illustrated transducer model block 160, the equations 2 and 3 include KmsTD and RmsTD, and then Kms_PR and Rms_PR. The transducer model block 160 provides X1 (e.g., the predicted position of the voice coil 112, and the correction block 154 generates I_corrected corresponding to the current to be provided to the voice coil 112), so that the voice coil 112 moves to the predicted position within the speaker 102.

[0138] Fig.12 Depicts the corresponding values ​​of Rms and Kms as impedance varies according to one embodiment. In general, many combinations of Rms and Kms can produce the same impedance at a particular frequency. With respect to the speaker 102, since the Rms and Kms of the speaker 102 tend to vary in a predictable manner, in many cases there may be one or a unique set of Rms and Kms values ​​for a particular frequency and a particular average impedance. This is generally Fig.12 Shown in.

[0139] Therefore, if the impedance at a specific frequency is known, the values ​​of Kms and Rms can be predicted. One way to calculate Kms and Rms from the impedance at a specific frequency may involve using a polynomial to match Fig.12 Curves for each of Kms and Rms are shown.

[0140] In some cases, combined with the use of Fig.11 The suspension 114 of the diaphragm 110 may soften faster than the indicated measurement impedance implementation (e.g., system 300) can detect. This may be the case because the system 300 relies on obtaining an average value, which takes time. If this is a problem, a hybrid approach may be used in which a hardening τ and measured Kms and Rms are selected when the suspension 114 of the diaphragm 110 is hardened, and a softening τ and estimated steady-state Kms and Rms are selected when the suspension 114 of the diaphragm 110 is softened, as described above in conjunction with Fig.10 As indicated in system 250 .

[0141] Fig.13 A system 320 for providing Kms and Rms according to one embodiment is generally shown. Figure 5 In addition to the illustrated first normalization circuit 130, the second normalization circuit 132, the first multiplier circuit 134, and the second multiplier circuit 136, the system 320 also includes the device 300. As discussed above, the device 300 provides a measured average value of Kms and a measured average value of Rms. In order to model the nonlinear Kms and Rms including the effects of position and motion history, the average static value (e.g., the measured average Kms and the measured average Rms) can be multiplied by the first multiplier circuit 134 and the second multiplier circuit 136 with the normalized polynomial models of Kms(x1) and Rms(x1) of the first normalization circuit 130 and the second normalization circuit 132 to provide Kms and Rms, respectively.

[0142] It is recognized that the systems 200, 250, 300 and 320 may be used in Figure 6 , Figure 7 and Figure 8 The transducer prediction model described is implemented in block 156. For example, although Fig. 9 , Fig.10 , Fig.11 and Fig.13 The systems 200 , 250 , 300 , and 320 depict an input designated as X1 , but it is recognized that once generated by the transducer prediction model block 156 , the systems 200 , 250 , 300 , and 320 utilize X1 as feedback.

[0143] In some cases, it is possible that Rms and Kms may have similar behavior with respect to both location and / or history. If this is the case, then as a computational simplification, only Kms may be calculated, and Rms provided as a scalar multiple of the Kms value. In the same manner, the Kms and Rms of the passive radiator 104 may be appropriately scaled to the Kms and Rms of the transducer 102.

[0144] Provides an adaptive model for the Rdc of the voice coil

[0145] Furthermore, by picking a frequency that is above the resonance of the loudspeaker 102 (i.e., where the velocity and back EMF may be low) and below the frequency where the inductance of the voice coil 112 becomes significant and adds to the impedance, a close approximation of the resistance of the voice coil 112 may be measured by measuring the impedance at mid-band frequencies in a manner similar to that performed to determine Kms and Rms. The measured impedance may approximate the DC resistance of the voice coil 112. Since the DC resistance may be a function of temperature, the DC resistance may be used to determine the temperature of the voice coil 112 to determine whether the voice coil 112 requires thermal protection, etc. Therefore, the measured DC resistance may also be used in place of the voice coil resistance (e.g., Rvc_avg) calculated based on the thermal model in equation (13) to make the model further adaptive.

[0146] Ventilation system examples

[0147] For various reasons, some speakers are designed with tuning ports or vents instead of passive radiators. Such vents have an acoustic mass, like the mass of the passive radiator 104, which can resonate with the system to produce low frequency output. However, the vent may not have a stiffness equivalent to the stiffness Kms_PR of the suspension of the passive radiator 104. Therefore, in order to account for the movement of the acoustic mass in the vent, the value of Kms_PR can be set to zero. The characteristics of the movement of air in the vent are that it can also have friction losses, especially at higher speeds. Therefore, it may be more accurate to include Rms_vent, where Rms_vent is also a function of speed, as described for Rms of the transducer 102.

[0148] Sealed box case

[0149] Some loudspeakers may be designed without passive radiators or tuned vents. Such loudspeakers are often referred to as closed box systems. In this case, the position of the passive radiator (x2) or the position of the vent may be set equal to zero. For computational simplicity, it may not be necessary to solve for the motion of the passive radiator diaphragm since there is no passive radiator in the system.

[0150] Infinite baffle case

[0151] In the case where the transducer 102 is mounted in a very large housing 101 (e.g., such as a rear rack in a vehicle, which may include the entire trunk as housing 101; or in a door that is acoustically open to the outside), the system may be considered an infinite baffle system. In this case, p(x1, x2) in equation (2) is set to zero, and equation (3) may be ignored.

[0152] Modeling Passive Radiators

[0153] The same method used to predict the Kms and Rms of the transducer (i.e., Kms_TD, Rms_TD, see equation (1)) can be applied to predict the Kms and Rms of the passive radiator 104, because the suspension of the passive radiator 104 behaves similarly to the suspension of the transducer 102. However, as discussed above, it may be sufficient to simply vary the Kms and Rms of the passive radiator 104 in proportion to Kms_avg (measured or predicted). In general, it may be sufficient to model Kms_PR and Rms_PR as fixed values ​​to provide improvements. This is applicable to Kms_PR and Rms_PR from equation (2).

[0154] Adaptation to non-permanent magnet based moving coil transducers

[0155] In a moving coil transducer (or SAM driver) based on a non-permanent magnet, "B" (the magnetic field) is constantly changing as the magnetizing current (i.e., the current in the fixed coil) is constantly adjusted. The magnetic field effects can be corrected by using fixed equalization (e.g., equalization similar to that in the current source correction method described above) in the case of a current source amplifier for the voice coil 112, or by dynamic equalization that adjusts its frequency response based on B as B changes over time in the case of a voltage source amplifier. Further correction is then performed by multiplying the equalized signal by B_nominal / B.

[0156] If the nonlinear correction method is to be adapted to a SAM drive, a similar method for determining the target (or magnetizing) current for the current of the fixed coil can be applied. However, the equalization functions of the algorithm in the SAM drive can be replaced by those described herein. For example, as described above in conjunction with Figure 6 The current source implementations described (eg, system 150) may be used in conjunction with current source implementations of moving coil transducers based on non-permanent magnets. Figure 8The voltage source implementations described (e.g., system 180) may be used in conjunction with voltage source implementations of moving coil transducers based on non-permanent magnets. This may indicate that the dynamic equalization function for the voltage source implementation of the SAM driver may be replaced by a conversion from voltage to target current. Thus, dynamic equalization may no longer be required and may in fact be improved upon by the aspects disclosed herein.

[0157] The function of multiplying the equalized signal by B_nominal / B can be replaced by replacing BL(0) with BL(I_stationary) in equation (7). This replaces equation (7) with the following equation:

[0158] BL=(cBL 4 ·x 4 +cBL 3 ·x 3 +cBL 2 ·x 2 +cBL 1 ·x+1)·BL(I stationary_coil ) Equation (19)

[0159] In the case of a SAM driver, a function is provided that converts the magnetising current to a flux density 'B' in the voice coil gap. Simply multiplying this B by the fixed voice coil wire length L provides BL(I_stationary) (ie the magnetic flux density and the length of the fixed voice coil wire).

[0160] Fig.14 One implementation of an acoustic transducer arrangement (or audio amplifier system) 500 according to one embodiment is depicted. The acoustic transducer arrangement 500 includes an input terminal, a control block (or controller) 503, and a transducer (or speaker) 506. An input audio signal (e.g., Vi) 502 is provided to an input terminal of the control block 503. The control block 503 generates a moving coil control signal (e.g., I m ) and the stationary coil control signal (e.g., I S ). The transducer 506 includes a magnetic material 512, a diaphragm 514, a bobbin 516, a fixed coil 518, and a moving coil (or voice coil) 520. The moving coil 520 is attached to the bobbin 516.

[0161] The magnetic material 512 is generally toroidal and has a toroidal cavity. The stationary coil 518 is positioned within the cavity. In various embodiments, the magnetic material 512 may be formed from one or more portions, which may allow the stationary coil 518 to be more easily inserted into the cavity or formed therein. The magnetic material 512 is magnetized in response to the stationary coil signal, thereby generating magnetic flux in the magnetic material 512. The magnetic material 512 includes a toroidal air gap 536 in the magnetic circuit 538 and the magnetic flux flows through and near the air gap 536.

[0162] The magnetic material 512 may be formed of any material that can be magnetized in the presence of a magnetic field. In various embodiments, the magnetic material 512 may be formed of two or more such materials. In some embodiments, the magnetic material 512 may be formed of laminates. In some embodiments, the laminates may be radially assembled and may be wedge-shaped so as to form a composite magnetic material with no gaps between the laminates.

[0163] The moving coil 520 is mounted on the skeleton 516 and receives the moving coil signal from the control block 503. The diaphragm 514 is mounted to the skeleton 516 so that the diaphragm 514 moves with the skeleton 516 and the moving coil skeleton 520. The skeleton 516 and the moving coil 520 move within the air gap 536 in response to the moving coil signal and the flux in the air gap 536. In general, various components of the acoustic transducer 506 that move with the skeleton 516 may be referred to as moving components. Components that are fixed when the skeleton 516 moves may be referred to as fixed components. The fixed components of the acoustic transducer 606 generally include the magnetic material 512 and the fixed coil 518.

[0164] In various embodiments, the acoustic transducer 506 can be adapted to vent the air space between the dust cover 532 and the magnetic material 512. For example, pores can be formed in the magnetic material 512 or pores can be formed in the skeleton 516 to allow the air space to vent, thereby reducing or preventing the effect of air pressure on the movement of the diaphragm 514.

[0165] The control block 103 generally includes the core correction block 154, a filter 552 (e.g., a second order filter), a conversion circuit 554, a correction block 556, a first current source 558, a second current source 560, a square root circuit 562, and a peak detector circuit 564. Generally, the transducer arrangement 500 utilizes the first current source 558 and the second current source 560 to replace the voltage source, which enables the transducer arrangement 500 to be suitable for automotive applications.

[0166] There is generally a need for a control method for providing current to the stationary coil 518 and the moving coil 520 that incorporates the benefits of both voltage and current sources and improves transient response and provides improved latency, accuracy, and appropriate protection and diagnostics. The control block 503 provides a moving coil signal (e.g., I m ) (or voice coil signal) to provide simplified frequency compensation. The first current source 558 eliminates the damping effect of the resistance of the moving coil 520 because the current (i.e., the moving coil signal) does not depend on the impedance of the moving coil 520. Therefore, the frequency response of the transducer 506 no longer depends on the moving coil signal (i.e., the current) and is instead fixed. This, on the one hand, enables a single fixed, time-invariant second-order filter 552 to be used to compensate the frequency response.

[0167] The optimum efficiency of the transducer 506 (including power for both the fixed coil 518 and the moving coil 520) is achieved when the power for the fixed coil 518 and the moving coil 520 is balanced (i.e., until the motor assembly steel of the transducer 506 begins to saturate, at which time there is no benefit in further increasing the current of the fixed coil 506). In the case of the transducer arrangement 500, since the power in a resistor is proportional to the square of the current, this is approximated by a square root circuit 562 of the peak of the audio signal level detected by the peak detector circuit 564 to set the current for the fixed coil 518. In order to use the output 504 from the square root circuit 562 directly, a proportional current source (i.e., the second current source 560) is used to drive the fixed coil 518.

[0168] To compensate for the sensitivity of the output 504 to changes that are proportional to the fixed coil signal (or the current of the fixed coil 518), a conversion circuit 554 may be used to calculate the magnetic flux in the air gap 536 of the moving coil 520 having a function B(i). The conversion circuit 554 provides the flux density (B(i)) to the core correction block 154. The flux density generally corresponds to the flux density in the air gap 536 of the moving coil 520 (or voice coil) that varies with the magnetization or fixed coil current.

[0169] The transducer prediction model block 156 calculates BL as described above by replacing BL(0) described in equation (7) with BL(I_stationary). Fig.15One implementation (or circuit) 580 for calculating the product of magnetic flux density B and the length (L) of the voice coil wire (or moving coil wire) according to one embodiment is generally shown. Circuit 580 generally includes a multiplier 582 and a normalization circuit 584. As shown, BL (0) (i.e., the product of magnetic flux density B and the length of the moving coil wire (when the moving coil 520 is stationary)) is provided to multiplier 582, and the position (x1) of the moving coil 520 is provided to the normalization circuit 584. Multiplier 582 takes the product of BL (0) and the output from the normalization circuit 584 to provide the product of magnetic flux density B and the length of the moving coil wire (or BL (x1)). As noted above, BL can be obtained by Equation 7.

[0170] Fig.16 A circuit 580 is generally shown for calculating the product of magnetic flux density B and the length (L) of the voice coil wire (or moving coil wire) according to one embodiment. In this case, BL (I stationary_coil )(or BL(I S )) replaces BL(0) to provide BL(x1) or BL, which is a function of the current in the fixed coil 518 and the position of the voice coil (or moving coil) 512. For example, "B" from the core correction block 154 can be replaced by the length of the voice coil "L" (which is a constant) to provide a new scalar portion when x=0, which is a function of the current in the fixed coil 518 and the position of the voice coil 520. This can be described in the following equation.

[0171] BL=(cBL 4 ·x 4 +cBL 3 ·x 3 +cBL 2 ·x 2 +cBL 1 ·x+1)·BL(I stationary_coil ) Equation (19)

[0172] Fig.17 Another implementation of an acoustic transducer arrangement (or audio amplifier system) 600 is generally shown according to one embodiment. The transducer arrangement 600 generally includes a transducer 506 and a control block 602. The control block 602 generally includes a current conversion block 182 (from Figure 8 ), adaptive block 184 (from Figure 8 ), voltage conversion block 186 (also from Figure 8), a first voltage source 607, a mobile coil power estimation block 610, a fixed coil power estimation block 612, a subtractor circuit 614, a second voltage source 616, a fixed coil building block 618, and a conversion circuit 620. Typically, the first voltage source 608 and the second voltage source 616 replace the first current source 558 and the second current source 560, respectively, such as in combination Fig.14 In this case, Fig.14 The filter 552 of the control block 503 in may no longer be fixed, since the impedance of the moving coil 520 may no longer be cancelled by the first current source 558. In the case of the arrangement 600, the conversion circuit 620 provides an output representing the magnetic flux in the air gap 536 of the moving coil 520 having the function B(i). Figure 8 , the adaptation block 184 uses the output of the conversion circuit 620 to determine the correction BL of the model based on the changing "B" (or magnetic flux density) * the length "L" of the voice coil 520 through Equation 19. The output of the conversion circuit 620 represents BL, which is BL(I_stationary) and this represents BL(Is) (or BL(I_stationary)) of Equation 19. Generally, the output of the conversion circuit 220 is scaled by the length of the winding (or wire) of the voice coil (or moving coil 520), which is the L in the output variable BL. The voltage conversion block 186 generates V_corrected based on BL, which is provided as an input to the moving coil 520.

[0173] However, now that the second voltage source 616 replaces the second current source 560, the target output current 605 is no longer proportional to the actual current of the stationary coil 518 due to the impedance of the stationary coil 518. To compensate for this aspect, the stationary coil modeling block 618 (e.g., inductance model) assumes that the resistance of the stationary coil 518 is known. In practice, this is not the case due to temperature effects that can change the resistance by 50% or more. These errors result in errors in both frequency compensation and sensitivity compensation under both steady-state and transient conditions when the current of the stationary coil 518 changes.

[0174] It is recognized that the best efficiency of the transducer 506 can be achieved when the power of the fixed coil 518 and the mobile coil 520 is balanced. Therefore, the arrangement 600 utilizes a square root approximation method. For example, the mobile coil power estimation block 610 determines the average power of the mobile coil 520, while the fixed coil power estimation block 612 determines the average power of the fixed coil 518. The subtractor circuit 614 compares the power of the mobile coil 520 to the power of the fixed coil 518. If the average power of the mobile coil 520 is greater than the average power of the fixed coil 518, the subtractor circuit (or difference block) 614 increases the output 605 used as the target current (or fixed coil signal) for the fixed coil 518 (or increases the current of the fixed coil 518 to avoid unnecessary audible sound. This condition causes the power to the mobile coil 520 to be reduced and the power between the fixed coil 518 and the mobile coil 520 to be balanced. When this condition occurs, it essentially necessitates a slight reduction in efficiency, however, because it is generally necessary to force the current to decrease quickly in the fixed coil 518, this condition may not be avoided. This can generate audible sounds on the one hand, so as a trade-off, it is more advantageous to proceed with a slight reduction in efficiency. The fixed coil modeling block 618 is generally configured to model the behavior of the fixed coil inductance. For example, in Fig.17 In the case of an implementation of , no measurement of the current is performed. Instead, the stationary coil modeling block 618 is configured to predict the inductive behavior of the stationary coil 518 using a model (e.g., which is an L / R inductance / resistance model). In addition, the stationary coil modeling block 618 is further configured to determine the current in the inductor of the stationary coil 518 when a voltage is applied to the stationary coil modeling block 618.

[0175] However, the average power should be estimated over a significantly longer period of time to avoid distortion and may be between 0.1 and 1 second. This causes the fixed coil signal (or the current of the fixed coil 518) to not follow quickly during transient conditions when the level of the audio signal increases rapidly. The result is that the sensitivity of the transducer 506 remains low for a longer period of time during the transient and thus requires significantly higher transient moving coil 520 amplifier peak power, or the output SPL of the transducer 506 may be limited during the transient. In addition, the slow following of the fixed coil signal relative to the level of the input audio signal 502 may impair efficiency because the balance of power may not be maintained when the level of the input audio signal 502 has high dynamic content.

[0176] Fig.18Another implementation of an acoustic transducer arrangement (or audio amplifier system) 700 according to one embodiment is generally shown. The acoustic transducer arrangement 700 includes a transducer 506 and an acoustic transducer controller (or controller) 702. The acoustic transducer controller 702 typically includes at least one digital processor 701 and a memory 703. The digital processor 701 typically performs the functions performed by the controller 702. In response to receiving an input audio signal 502, the controller 702 generates a moving coil signal and a fixed coil signal and transmits them to the moving coil 520 and the fixed coil 518, respectively.

[0177] The arrangement 700 is generally configured to: balance power between the stationary coil 518 and the moving coil 520 while achieving fast transient response; improve frequency accuracy and sensitivity compensation in the presence of a changing current of the stationary coil 518 (i.e., a changing stationary coil signal); and improve the efficiency of the transducer 506 without relying on a current source to generate the stationary coil signal and the moving coil signal without introducing distortion. In addition, as will be described in more detail, the arrangement 700 is generally configured to provide protection and diagnostics for the electronics used in conjunction with the stationary coil 518.

[0178] The controller 702 includes a current transformation block 182, an adaptation block 184, a voltage transformation block 186, a voltage source 708, a complex source block 710, a conversion circuit 712, and a peak setting block 714. The complex source block 710 is provided as an alternative to combining Fig.14 The indicated second current source 560 and the alternative combination Fig.17 Indicated second voltage source 616. In general, complex source block 710 is configured to control or regulate its impedance to generate a stationary coil signal for transmission to stationary coil 518.

[0179] The average and peak setting block 714 takes the peak values ​​of the input audio signal 502 and uses a low pass filter to remove the peak values ​​that are associated with a simple peak detector (similar to Fig.14 The ripple associated with the peak detector circuit 564 of the input audio signal can be detected. During the transient, by setting the value of the low pass filter directly to the instantaneous maximum absolute value of the input audio signal, the slowly varying low pass filter can be forced to respond immediately to the transient. In this way, a clean envelope of the input audio signal with minimal ripple of the input audio signal 502 can be generated, which can respond to the transient increase in the level of the audio signal and be provided to the complex source block 710 as the reference signal 705.

[0180] With this arrangement 700, the level of the input audio signal 502 can be used without ( Fig.14The square root circuit 562 (or square root function) of FIG. 5 is used because the output voltage of the source for the fixed coil 518 is a known voltage, where power is proportional to the square of the voltage. This means that the power in the mobile coil 520 is proportional to the power in the fixed coil 518 because the fixed coil signal (or the current provided to the fixed coil 518) is proportional to the input audio signal 502 through the reference signal 505 and the power in the mobile coil 520 is proportional to the square of the current in the fixed coil 518. This approach may not be as accurate as the power balancing performed by the mobile coil power estimation block 610 and the fixed coil power estimation block 612 because the approach indicated ignores the effect of the frequency-dependent impedance of the mobile coil 520. However, for music and noise signals, the average scaling value can be chosen to roughly estimate the effect of the impedance of the mobile coil 520 on music and noise well enough.

[0181] The complex source block 710 measures the current (or fixed coil signal) provided to the fixed coil 518 using a current measurement circuit 707. The current measurement circuit 707 can be a resistor, a current transformer, a Hall effect sensor, etc. The measured current (i.e., the measured fixed coil signal) is provided as feedback to the compensation block 720 to provide an error signal to the adder circuit 722. The adder circuit 722 compares the reference signal 705 with the error signal (or subtracts the error signal from the reference signal 705) and adjusts the voltage source 724. It is recognized that the voltage source 724 can be implemented as a pulse width modulated ("PWM") (or other modulation scheme) buck (or other topology) regulator with a filter 725. The filter 725 typically includes an inductor 726 and a capacitor 728 to filter the voltage output from the voltage source 724. The compensation block 720 and the filter 725 typically have an output impedance such that the complex source block 710 looks like the desired case of a current source, a voltage source, or a mixed frequency-dependent source. In particular, it may be desirable to have complex source block 710 behave as a current source at low frequencies, and behave as a voltage source at frequencies above the mechanical resonance of transducer 106 (e.g., 50-100 Hz for a 6-inch mid-bass driver). This, in turn, can improve distortion in the passband of transducer 506 while providing accurate control of the average current of the fixed coil (or the average value of the fixed coil signal) and transient levels.

[0182] In order to realize the behavior of the impedance in the case of the complex source block 710, the compensation block 720 can be implemented as, for example, a proportional-integral-differential (PID) controller. For example, the compensation block 720 can include a proportional path with a gain "Kp" in the current feedback path, where the current of the fixed coil signal is measured by the current measurement circuit 707. The integral term and the differential term (i.e., Ki and Kd) can be, for example, zero. It is sufficient for the filter 725 to use a proportional current feedback K (i.e., Ki and Kd=0). The integral term Ki and the differential term Kd are stable because the 2nd order system generated by the inductor 726 and the capacitor 728 is simplified to a first order system by means of the current measurement and the proportional current feedback Kp performed by the current measurement circuit 707. By using the proportional current feedback Kp in the feedback path, this condition produces an effective current source.

[0183] In this arrangement, the inductance of inductor 726 is effectively cancelled (in a stability sense) by means of a current source generated by using current feedback. By selecting an appropriate gain Kp in the feedback path of compensation block 720, the frequency at which the natural impedance of capacitor 728 affects the output impedance can be regulated. The higher the gain of Kp, the higher the frequency will be. At high frequencies, the impedance provided by complex source block 710 is determined by the impedance of capacitor 728 and thus looks like a voltage source. To do this, the capacitance size of capacitor 728 should be large enough so that at a desired frequency above the resonance of transducer 506, the impedance of capacitor 528 is similar to or less than the impedance of transducer 506. At low frequencies where the impedance of capacitor 728 is high, the output current will be determined by the effective current source generated by using current feedback. Therefore, control block 702 can provide the characteristic impedance of a current source at low frequencies and the characteristic impedance of a voltage source at high frequencies. Where the higher frequency is typically 3 to 5 times the mechanical resonance of transducer 506, and the low frequency is typically any frequency below the high frequency. Finally, it is recognized that this same effect can be achieved by other control methods such as utilizing voltage sensing and adding an integral term Ki and a proportional term Kp and possibly a derivative term Kd to increase stability. The foregoing can be represented in either the s-domain or the z-domain.

[0184] Furthermore, in a system in which current is to be fed to more than one stationary coil 518, the loads of the stationary coils 518 may be connected in parallel to each other and one control loop and voltage source may be used. However, in order to prevent a malfunction at the input terminal arranged between the controller 702 and the stationary coil 518, the measured current at the current measurement circuit 707 in the feedback path mentioned above may be the higher current among the multiple currents of the stationary coil 518 at any time. In this way, the current of the stationary coil 518 is adjusted according to the load of the stationary coil 518 that provides the highest current.

[0185] Typically, the current level of the fixed coil 518 for optimizing the efficiency of the arrangement 700 is typically determined by the peak setting block 714. The peak setting block 714 receives the input audio signal 502. This arrangement helps avoid large variations in the desired current of the fixed coil 518 near the resonance of the transducer 506. At near resonance, less power is required to produce the same acoustic output level. For this reason, the arrangements 500 and 600 typically result in a reduction in the current of the fixed coil 518 at resonance to balance the power. However, when the current of the fixed coil 518 is reduced, the damping is reduced, thereby requiring even lower power for the moving coil 520, which results in a further reduction in the current of the fixed coil 518. Because the transducer 506 may be almost completely damped due to its mechanical losses, this result may result in errors in both sensitivity and frequency response at near resonance. Therefore, by providing the input audio signal 502 to the peak setting block 714, this condition circumvents the errors noted above. While this may result in that power balance between the stationary coil 518 and the moving coil 520 may not be maintained at near resonance, this aspect may not be a concern because the power levels of the stationary coil 518 and the moving coil 520 are low at near resonance.

[0186] The conversion circuit 712 may receive the measured current of the stationary coil 518 (i.e., the stationary coil signal) to determine the flux density in the air gap 536. The determined flux density in the air gap 536 is used to determine the varying acoustic frequency response and acoustic sensitivity of the transducer 506 based on the current of the stationary coil 518. If the measured current of the stationary coil 518 is used to determine the magnetic flux in the air gap 536, the adaptation block 184 may correct for the sensitivity of the stationary coil 518. However, if the measured current of the moving coil 520 is used directly to determine the magnetic flux in the air gap 536 and thus the sensitivity and frequency response, distortion may occur at some frequencies and levels.

[0187] Typically, the stationary coil 518 replaces a conventional magnet that is typically used to generate magnetic flux in the air gap 536 to enable the transducer 506 to output audio. However, when the transducer 506 is to output high peak audio (i.e., drum hits, etc.), the stationary coil 518 utilizes a high amount of current. Therefore, the controller 702 adjusts the current on Is based on the envelope of the input audio signal 502. The controller 702 reduces the current on Is when it is not necessary to output high levels of audio, and increases the current on Is when it is necessary to output high levels of audio (i.e., providing dynamic adjustment of the current).

[0188] The complex source 710 provides an output current Is to a conversion circuit 712, which provides a value corresponding to the magnetic flux in the air gap 536 of the moving coil 520 having a function B(i). The adaptive block 184 uses the output of the conversion circuit 712 to determine the correction BL of the model based on the changing "B" (or magnetic flux) through equation 19. The voltage conversion block 186 generates V_corrected, which is provided as an input to the moving coil 520 through the voltage source 708. The adaptive block 184 uses the flux value to provide the same frequency response for the input audio signal. The complex source 710 has the impedance characteristics of a voltage source or a current source. When the audio input signal has a large level, the complex source 710 forces the current of Is to climb quickly and quietly.

[0189] The stationary coil 518 and the moving coil 520 are transducer coupled through the magnetic material 512. Therefore, the current in the moving coil 520 will produce a transducer coupled or reflected current in the stationary coil 518. This distortion will be more prevalent or apparent at frequencies and signal levels where the reflected current of the moving coil 520 is large compared to the average value of the current of the stationary coil 518. When the stationary coil current is measured using the current measurement circuit 707, the measurement may include the current reflected from the current of the moving coil 520. However, the controller 702 may use the measured current to determine the acoustic sensitivity of the transducer 506. When the current phase relationship of the moving coil 520 is correct, the reflected current of the moving coil 520 to the stationary coil 518 may be subtracted from the average current of the stationary coil 518, thereby causing the conversion circuit 712 to calculate a lower flux density in the gap 536 and therefore a lower sensitivity. This, on the one hand, may reflect more current in the stationary coil 518, which will further subtract from the average current in the stationary coil 518, causing the conversion circuit 712 to calculate a still lower flux density in the gap 536 and ultimately increase the current of the moving coil 520. Thus, a positive feedback is established that causes the distortion mentioned above.

[0190] In the opposite phase, the current of the moving coil 520 may add to the average of the current of the fixed coil 518, causing the conversion circuit 712 to calculate a higher flux density in the gap 536. This provides the same positive feedback, which then causes distortion. The result is that at some frequencies, the generated output signal is asymmetrically distorted by large even-order distortion components. In one aspect, it may be advantageous to separate the effect of the current of the moving coil 520 reflected to the fixed coil 518, which is used to determine the sensitivity compensation and also determine the frequency compensation.

[0191] Fig.19Another implementation of an acoustic transducer arrangement (or audio amplifier system) 800 according to another embodiment is generally depicted. The acoustic transducer arrangement 800 includes a transducer 506 and an acoustic transducer controller (or controller) 802. The acoustic transducer controller 802 typically includes at least one digital processor 801 and a memory 803. The digital processor 801 typically performs the functions performed by the controller 802. In response to receiving an input audio signal 502, the acoustic transducer controller 802 generates and transmits a moving coil signal (Im) and a fixed coil signal (Is) to the moving coil 520 and the fixed coil 518, respectively.

[0192] The controller 802 includes a current transformation block 182, an adaptation block 184, a voltage transformation block 186, a voltage source 808, a complex source block 710, a conversion circuit 712, and a peak setting block 714, as well as a pre-processing block 804. In the case of the arrangement 800, the measured current of the fixed coil 518 is not used directly to determine the magnetic flux of the air gap 536 of the moving coil 520. Instead, the pre-processing block 804 pre-processes the measured current of the fixed coil 518. For example, the pre-processing block 804 takes the long-term average voltage amplitude of the fixed coil 518 measured at the current measurement circuit 707 to determine the average resistance of the fixed coil 518. The average resistance of the fixed coil 518 is used in the L / R model of the fixed coil 518 to predict the effective average current of the fixed coil 518, which will then be free of reflected current from the moving coil 520. The L / R model is similar to the model of the fixed coil building block 618 (see Fig.17 ), but where the resistance R of the stationary coil 518 has been measured and is therefore more accurately included. In this case, the resistance and internal temperature of the moving coil 520 are accurately known, which helps predict the effective average current of the stationary coil 518. It is recognized that when the current of the stationary coil 518 is low, it may be difficult to calculate the resistance of the moving coil 520, and that the inductance is generally an ideal approximation of the actual inductance of the stationary coil 518, which may include all non-ideal aspects of the inductor, such as residual magnetism in steel, saturation, and other effects.

[0193] The pre-processing block 804 is configured to employ a fast averaging and peak setting function of the measured current. The pre-processing block 804 first obtains the peak detection value of the measured current of the fixed coil 518, and then averages the peak detection value using the low-pass filter therein. The filtering removes most of the reflected current from the moving coil 520 from the measured current of the fixed coil 518. In order to respond to the transient, fast rising current of the fixed coil 518, the value passing through the filter can be forced to be a peak value during the fast rise of the fixed current. This may be optimal for eliminating the distortions noted above, except for pure sine waves with frequencies below the low-pass filter cutoff frequency (where the filter can no longer estimate the average fixed current).

[0194] Similar to the above Fig.18 , the control block 710 provides the output current Is to the conversion circuit 712, which provides a value corresponding to the magnetic flux in the air gap 536 of the moving coil 520 having the function B(i). The adaptive block 184 uses the output of the conversion circuit 712 to determine the correction BL of the model based on the changing "B" (or magnetic flux) through equation 19. The voltage conversion block 186 generates V_corrected, which is provided as an input to the moving coil 520 through the voltage source 808.

[0195] Fig. 20 Generally depicted is an acoustic transducer arrangement (or audio amplifier system) 900 according to another embodiment. The acoustic transducer arrangement 900 includes a transducer 506 and an acoustic transducer controller (or controller) 902. The acoustic transducer controller 902 typically includes at least one digital processor 901 and a memory 903. The digital processor 901 typically performs the functions performed by the controller 902. In response to receiving an input audio signal 502, the controller 902 generates a moving coil signal (e.g., Im) and a fixed coil signal (e.g., Is) and transmits them to the moving coil 520 and the fixed coil 518, respectively.

[0196] The controller 902 includes a current transformation block 182, an adaptation block 184, a voltage transformation block 186, a voltage source 908, a complex source block 910, a conversion circuit 912, a peak setting block 1014, a preprocessing block 954, a delay block 904, a signal type discriminator block 906, and a signal scaling block 911. The feedback path 923 and the feedforward path 924 are shown as providing inputs to the preprocessing block 954. For example, the preprocessing block 954 includes a peak setting block (or a slow average and peak setting block) 914, a switch 925, and a fast average peak setting block 916. The signal type discriminator block 906 is configured to select the switch 925 so that the feedback path 923 or the feedforward path 924 is selected to provide the input from the peak setting block 914 or the fast average peak setting block 916 to the conversion circuit 912.

[0197] The signal type discriminator block 906 determines when the input audio signal 502 is below the cutoff frequency (or predetermined frequency) of the low pass filter of the pre-processing block 954 or is primarily sinusoidal in nature (e.g., a single tone or test signal having a single frequency). If this condition is true (i.e., the input audio signal 502 is a sine wave), the peak setting block 914 can be used as an input to the conversion circuit 912 in conjunction with the switch 925 in the feedforward path 924. As noted above, the peak setting block 914 provides a target current for the fixed coil 518. Since the feedback path is not used, this mode eliminates the effect of the reflected current of the moving coil 520 by eliminating the feedback path 923. In addition, the peak setting block 914 includes a fast peak setting function in the same manner as the fast average peak setting block 916 to allow fast transients to set the output of the block 514 to eliminate the delay associated with the averaging filter.

[0198] In the event that the signal type discriminator block 906 determines that the input audio signal 502 is above the cutoff frequency of the low pass filter of the pre-processing block 954 (or the input audio signal 502 is not primarily sinusoidal in nature), then the fast average peak setting block 916 can be used with the switch 925 in the feedback path 923 to provide the output from the fast average and peak setting block 916 to the conversion circuit 912. The fast average and peak setting block 916 plays a similar role to the average and peak setting block 714. However, the cutoff frequency of the average low pass filter of the fast average peak setting block 916 is comparable to the resonance of the transducer 506. This replicates the operation of the pre-processing block 954 on the input audio signal. When the envelope of the current through the fixed coil 518 corresponds to the rapid changes in the measured current, the fast average peak setting block 916 provides the envelope. The complex source block 910 is configured to provide the current of the fixed coil 518 to the fast average peak setting block 916.

[0199] The signal scaling block 911 scales the level of the target current of the stationary coil 518 based on the properties of the input audio signal 502 as detected by the signal type discriminator block 906. In this way, an optimal current of the stationary coil 518 is provided for balancing the power of the sine wave, and a different optimal current that can better maintain a noise or music signal, the sine wave having a lower peak-to-average ratio than the noise or music. In addition, the delay block 904 provides additional time for the current of the stationary coil signal to rise to the target current of the stationary coil 518, particularly during fast transients.

[0200] The magnitude of the delay employed by delay block 904 may depend on the voltage available for driving the current of stationary coil 518, the inductance and resistance of stationary coil 518, the bandwidth of arrangement 900 and thus the transition rate of the reproduced transient, and secondary factors such as amplifier headroom, as determined by the power electronics employed (such as the power supply for voltage source 724). In some cases, no delay may be required.

[0201] The adaptive block 184 uses the output of the conversion circuit 912 to determine the corrected BL of the model based on the changing "B" (or magnetic flux) through Equation 19. The adaptive block 184 provides the corrected BL to the voltage transformation block 186. The voltage transformation block 186 generates V_corrected, which is provided as an input to the moving coil 520 through the voltage source 808. The voltage transformation block 186 generates V_corrected in response to the corrected BL.

[0202] Fig.21 A method 1000 performed by the audio amplifier system 150 is generally depicted according to one implementation.

[0203] In operation 1002, the audio amplifier system 150 receives an audio input signal. In operation 1004, the audio amplifier system 150 generates a target current signal based on the audio input signal. In operation 1006, the audio amplifier system 150 generates a first predicted position of the voice coil 112 for the speaker 102. In operation 1008, the audio amplifier system 150 generates a first correction current signal based on the target current signal and the first predicted position of the voice coil.

[0204] In operation 1010, audio amplifier system 150 determines a pressure within speaker housing 101. In operation 1012, audio amplifier system 150 determines a position of passive radiator 104 based on at least the pressure within speaker housing 101. In operation 1014, audio amplifier system 150 generates a second predicted position of voice coil 112 based on at least the pressure within speaker 102, the position of passive radiator 104, and the first correction current signal.

[0205] Fig. 22 A method 1020 performed by the audio amplifier system 180 is generally depicted according to one implementation.

[0206] In operation 1022, the audio amplifier system 180 generates a target current signal based on the audio input signal and the velocity of the diaphragm of the speaker 102. In operation 1024, the audio amplifier system 180 generates the target current signal based on the audio input signal and the velocity of the diaphragm 110 of the speaker 102. In operation 1026, the audio amplifier system 180 generates a first correction current signal based on at least the target current signal and based on a first predicted position of the voice coil 112 of the speaker 102.

[0207] Fig.23 A method 1040 performed by the audio amplifier system 500 is generally depicted according to another implementation.

[0208] In operation 1042, the audio amplifier system 500 receives an audio input signal. In operation 1044, the audio amplifier system 500 generates a target current signal based on the audio input signal. In operation 1046, the audio amplifier system generates a first predicted position of the voice coil 520 of the speaker 102. In operation 1048, the audio amplifier system 500 determines a pressure within the speaker housing 101 based on the first predicted position of the voice coil 112. In operation 1050, the audio amplifier system 500 determines a position of the passive radiator 104 based on at least the first predicted position of the voice coil 520. In operation 1052, the audio amplifier system 500 determines a flux density value of the speaker 506. In operation 1054, the audio amplifier system 500 generates a position of the voice coil 520.

[0209] Fig.24 A method 1060 performed by any of the audio amplifier systems 600 , 700 , 800 , and 900 is generally depicted.

[0210] In operation 1062, any one or more of the systems 600, 700, 800, and 900 generates an audio input signal. In operation 1064, any one or more of the systems 600, 700, 800, and 900 generates a target current signal based on the audio input signal and a velocity of the diaphragm 514 of the speaker 506. In operation 1066, any one or more of the systems 600, 700, 800, and 900 generates a correction current signal based on at least the target current signal and based on a predicted position of the voice coil 520 of the speaker 506. In operation 1064, any one or more of the systems 600, 700, 800, and 900 determines a predicted position of the voice coil 520 of the speaker 506 based on the flux density value.

[0211] Although exemplary embodiments are described above, it is not intended that these embodiments describe all possible forms of the present invention. In fact, the words used in the specification are descriptive rather than restrictive, and it should be understood that various changes can be made without departing from the spirit and scope of the present invention. In addition, the features of various implementation embodiments can be combined to form other embodiments of the present invention.

Claims

1. An audio amplifier system, comprising: Memory; as well as an audio amplifier, the audio amplifier comprising the memory and being configured to: receiving an audio input signal; generating a target current signal based on the audio input signal, wherein the target current signal corresponds to the audio input signal before correction and indicates at least a current to be supplied to a voice coil of a speaker before the correction; generating a first predicted position of a voice coil of a loudspeaker; generating a first correction current signal based on the target current signal and the first predicted position of the voice coil; determining a pressure within a speaker housing based on at least the first predicted position of the voice coil; determining a position of a passive radiator included in the speaker housing based on at least the pressure within the speaker housing; determining a flux density value of the speaker, wherein the flux density value corresponds to a product of a magnetic flux of an air gap in the speaker and a length of a voice coil wire in the speaker; generating a second predicted position of the voice coil based on the flux density value and based on at least the pressure within the loudspeaker, the position of the passive radiator, and the first correction current signal, and A second correction current signal is generated based on the target current signal and a second predicted position of the voice coil.

2. The system of claim 1, wherein the audio amplifier is further configured to: The second correction current signal is transmitted to the speaker to control the position of the voice coil based on the second correction current signal. 3 . The system of claim 1 , wherein the audio amplifier comprises a conversion block configured to store any number of the generated predicted positions of the voice coil to provide a history of the generated predicted positions of the voice coil.

4. The system of claim 3, wherein the conversion block is further configured to: generating an average spring model signal corresponding to a spring stiffness of the loudspeaker based on the history of generated predicted positions of the voice coil; and An average damping model signal corresponding to damping of the loudspeaker is generated based on the history of generated predicted positions of the voice coil.

5. The system of claim 4, wherein the spring stiffness of the loudspeaker corresponds to the spring stiffness of a surround and a spider of the loudspeaker.

6. The system of claim 4, wherein the damping of the loudspeaker corresponds to friction losses of a surround and a spider of the loudspeaker.

7. The system of claim 4, wherein the audio amplifier comprises: a spring normalization block configured to normalize the spring stiffness of the loudspeaker to a rest position corresponding to setting the predicted position of the voice coil to zero; as well as A damping normalization block is configured to normalize the damping of the loudspeaker to the rest position.

8. The system of claim 1, further comprising: a first plurality of filters configured to receive a first voice coil signal indicative of a measured voltage across the voice coil and to generate a first filter output based on the first voice coil signal; and a second plurality of filters configured to receive a second voice coil signal indicative of a measured current across the voice coil and to generate a second filter output based on the second voice coil signal; as well as A divider circuit is configured to generate an impedance signal indicative of an impedance of the voice coil based on the first filter output and the second filter output.

9. The system of claim 8, further comprising: a first polynomial block configured to determine a spring stiffness of the speaker based on the impedance signal; as well as A second polynomial block is configured to determine damping of the speaker based on the impedance signal.

10. An audio amplifier system, comprising: Memory; as well as an audio amplifier, the audio amplifier comprising the memory and being configured to: receiving an audio input signal; generating a target current signal based on the audio input signal and a velocity of a diaphragm of a speaker; generating a correction current signal based on at least the target current signal and based on a predicted position of a voice coil of the loudspeaker; and determining a predicted position of the voice coil of the loudspeaker based on the flux density value, wherein the flux density value corresponds to the product of the magnetic flux of the air gap of the voice coil in the speaker and the length of the voice coil wire in the speaker, The audio amplifier is further programmed to: providing a correction voltage signal to the speaker to control the position of the voice coil based on at least the predicted position of the voice coil, The audio amplifier comprises: A voice coil power estimation block is configured to determine an average power of the voice coil based on the correction voltage signal.

11. The system of claim 10, wherein the audio amplifier comprises a first transformation block configured to convert a voltage corresponding to the audio input signal into the target current signal based on the audio input signal and based on a velocity of the diaphragm of the speaker.

12. The system of claim 10, wherein the audio amplifier comprises a second transformation block configured to convert the correction current signal into the correction voltage signal based on the flux density and based on a velocity of the diaphragm of the loudspeaker.

13. The system of claim 10, wherein the audio amplifier comprises: A stationary coil power estimation block is configured to determine an average power of a stationary coil of the loudspeaker based on at least a current of the stationary coil.

14. The system of claim 13, wherein the audio amplifier further comprises: A subtractor circuit, the subtractor circuit being programmed to: comparing the average power of the voice coil to the average power of the stationary coil; and When the average power of the voice coil is greater than the average power of the fixed coil, a fixed coil current signal provided to the fixed coil of the speaker is increased.

15. The system of claim 10, wherein the audio amplifier comprises: a complex source block configured to provide a measurement current of a stationary coil in the speaker; as well as A flux conversion circuit is configured to provide a value corresponding to the magnetic flux of an air gap of the voice coil in the speaker based on the measured current of the stationary coil.

16. The system of claim 10, wherein the audio amplifier comprises: a complex source block configured to provide a measurement current of a stationary coil in the speaker; a pre-processing block configured to determine an average resistance of the stationary coil based on the measured current of the speaker; as well as A flux conversion circuit is configured to provide a value corresponding to the magnetic flux of an air gap of the voice coil in the speaker based on the average resistance of the stationary coil.

17. The system of claim 10, wherein the audio amplifier comprises: a signal type discriminator block configured to compare a frequency of the audio input signal with a predetermined frequency; a slow average peak setting block configured to provide an output to a flux conversion circuit providing a value corresponding to the magnetic flux of the air gap of the voice coil in the speaker to eliminate a reflected current from the voice coil of the speaker if the frequency of the audio input signal is lower than the predetermined frequency; a fast average peak setting block configured to provide an output to the flux conversion circuit providing the value to provide an envelope of a fixed coil current if the frequency of the audio input signal is above the predetermined frequency; as well as A complex source block is configured to provide the current of the stationary coil to the fast average peak setting block.

18. A computer program product, embedded in a non-transitory computer readable medium, the computer program product configured to amplify an audio input signal, the computer program product comprising instructions for: receiving an audio input signal; generating a target current signal based on the audio input signal, wherein the target current signal corresponds to the audio input signal before correction and indicates at least a current to be supplied to a voice coil of a speaker before the correction; generating a first predicted position of a voice coil of the loudspeaker; generating a first correction current signal based on the target current signal and the first predicted position of the voice coil; determining a pressure within a speaker housing based on at least the first predicted position of the voice coil; determining a position of a passive radiator included in the speaker housing based on at least the pressure within the speaker housing; determining a flux density value of the speaker, wherein the flux density value corresponds to a product of a magnetic flux of an air gap in the speaker and a length of a voice coil wire in the speaker; generating a second predicted position of the voice coil based on the flux density value and based on at least the pressure within the speaker, the position of the passive radiator, and the first correction current signal; as well as A second correction current signal is generated based on the target current signal and a second predicted position of the voice coil.

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