System and method for compensating for nonlinear behavior of magnetic flux-based acoustic transducers - Patents.com

Abstract

To provide a system and method for compensating for non-linear behavior of an acoustic transducer based on magnetic flux.SOLUTION: In at least another embodiment, an audio amplifier system including a memory and an audio amplifier is provided. The audio amplifier includes the memory and is programmed to receive an audio input signal and to generate a target current signal based on the audio input signal and a velocity of a diaphragm of a loudspeaker. The audio amplifier is further programmed to generate a corrected current signal based at least on the target current signal and on a predicted position of a voice coil of the loudspeaker and determine the predicted position of the voice coil of the loudspeaker based on a flux density value. The flux density value corresponds to a product of magnetic flux of an air gap for the voice coil in the loudspeaker and a length of a voice coil wire in the loudspeaker.SELECTED DRAWING: None

Description

[Technical field]

[0001] CROSS-REFERENCE TO RELATED APPLICATIONS This application may be related to U.S. Application No. __________, entitled “SYSTEM AND METHOD FOR NON-LINEAR BEHAVIOR FOR AN ACOUSTIC TRANSDUCER”, Attorney Docket No. HARM0681PUS, and filed on _______________.

[0002] One or more aspects disclosed herein are generally systems and methods for compensating for nonlinear behavior of acoustic transducers. Aspects disclosed herein may provide active, sensorless, nonlinear compensation methods and apparatus for loudspeakers using moving coil acoustic transducers, enclosures, and passive radiators. For example, low machine instruction seconds (MIPS), sensorless models and algorithms are provided for designs of closed, vented, and passive radiators driven by current and voltage sources to compensate for distortions caused by diaphragm suspensions and voice coil motors as a function of voice coil position. These and other aspects are discussed in more detail herein. [Background technology]

[0003] French PCT / US2018 / 052336 ("the '336 publication") 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 stationary coil signal to a stationary coil of the acoustic transducer based on the first reference signal and to measure a current through the stationary coil after providing the stationary coil signal to the stationary coil. The controller is further configured to generate a first output indicative of a current through the stationary 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 to the moving coil that is 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 [Means for solving the problem]

[0004] In at least one embodiment, an audio amplifier system is provided that includes a memory and an audio amplifier. The audio amplifier includes a 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 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 in a speaker enclosure based on at least the first predicted position of the voice coil and determine a position of a passive radiator based on at least the pressure in the speaker enclosure. The audio amplifier is further configured to determine a magnetic flux density value of the speaker and generate a second predicted position of the voice coil based on the magnetic flux density value and at least the pressure in the speaker, the position of the passive radiator, and the first correction current signal.

[0005] In at least another embodiment, an audio amplifier system is provided that includes a memory and an audio amplifier. The audio amplifier includes 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 speaker diaphragm. The audio amplifier is further programmed to generate a correction current signal based on at least the target current signal and an expected position of the speaker voice coil, and to determine the expected position of the speaker voice coil based on a magnetic flux density value that corresponds to a product of a magnetic flux in an air gap of the speaker voice coil and a length of the speaker voice coil wire.

[0006] In at least another embodiment, a computer program product embodied in a non-transitory computer readable medium programmed to amplify an audio input signal is provided. 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 speaker and generating a first corrected current signal based on the target current signal and the first predicted position of the voice coil. The computer program product includes instructions for generating a first predicted position of a voice coil of a speaker and generating a first corrected current signal based on the target current signal and the first predicted position of the voice coil. The computer program product further includes instructions for determining a pressure in a speaker enclosure 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 in the speaker enclosure. The computer program product further includes instructions for determining a value of a magnetic flux density of the speaker and generating a second predicted position of the voice coil based on the value of the magnetic flux density and at least the pressure in the speaker, the position of the passive radiator, and the first corrected current signal.

[0007] While embodiments of the present disclosure are pointed out with particularity in the appended claims, other features of the various embodiments will become more apparent and will be best understood by reference to the following detailed description taken in conjunction with the accompanying drawings, in which: For example, the present application provides the following: (Item 1) 1. An audio amplifier system comprising: Memory, an audio amplifier; the audio amplifier includes the memory; receiving an audio input signal; generating a target current signal based on the audio input signal; 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 enclosure based on at least the first predicted position of the voice coil; determining a position of a passive radiator based at least on a pressure within the speaker enclosure; determining a value of magnetic flux density of said loudspeaker; generating a second predicted position of the voice coil based on the value of the magnetic flux density, at least the pressure within the speaker, the position of the passive radiator, and the first corrected current signal; It is programmed to The audio amplifier system as described above. (Item 2) The system described in the preceding item, wherein the value of the magnetic flux density corresponds to the product of the magnetic flux in the air gap in the speaker and the length of the voice coil wire in the speaker. (Item 3) The audio amplifier further comprises: generating a second correction current signal based on the target current signal and the second predicted position of the voice coil; sending the second correction current signal to the speaker to control a position of the voice coil based on the second correction current signal; The system according to any one of the preceding claims, programmed to: (Item 4) 13. The system of claim 12, wherein the audio amplifier includes a converter block programmed to store any number of the generated predicted positions of the voice coil and provide a history of the generated predicted positions of the voice coil. (Item 5) The transformation block further comprises: generating an average spring model signal corresponding to a spring stiffness of the loudspeaker based on the history of the generated predicted positions of the voice coil; generating an average attenuation model signal corresponding to attenuation of the loudspeaker based on the history of the generated predicted positions of the voice coil; The system according to any one of the preceding claims, programmed to: (Item 6) 2. The system of claim 1, wherein the spring stiffness of the speaker corresponds to the spring stiffness of the speaker surround and spider. (Item 7) 2. The system of claim 1, wherein the attenuation of the speaker corresponds to friction losses in the speaker surround and spider. (Item 8) The audio amplifier, a spring normalization block programmed to normalize the spring stiffness of the speaker to a rest position corresponding to the predicted position of the voice coil, the rest position being set to zero; an attenuation normalization block programmed to normalize the attenuation of the speaker to the rest position; 3. The system according to claim 1, further comprising: (Item 9) a first plurality of filters programmed 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; a second plurality of filters programmed 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; a divider circuit programmed to generate an impedance signal indicative of an impedance of the voice coil based on the first filter output and the second filter output; 2. The system of claim 1, further comprising: (Item 10) a first polynomial block programmed to determine a spring stiffness of the loudspeaker based on the impedance signal; a second polynomial block programmed to determine an attenuation of the speaker based on the impedance signal; 2. The system of claim 1, further comprising: (Item 11) 1. An audio amplifier system comprising: Memory, an audio amplifier; the audio amplifier includes the memory; receiving an audio input signal; generating a target current signal based on the audio input signal and a velocity of a speaker diaphragm; generating a correction current signal based on at least the target current signal and an expected position of a voice coil of the loudspeaker; determining the predicted position of the voice coil of the speaker based on the value of magnetic flux density; It is programmed to The audio amplifier system, wherein the magnetic flux density value corresponds to the product of the magnetic flux in the air gap of the voice coil of the speaker and the length of the voice coil wire of the speaker. (Item 12) 13. The system of claim 12, wherein the audio amplifier is further programmed to provide a correction voltage signal to the speaker to control a position of the voice coil based at least on the predicted position of the voice coil. (Item 13) The system of any one of the preceding claims, wherein the audio amplifier includes a first conversion block programmed to convert a voltage corresponding to the audio input signal into the target current signal based on the audio input signal and the velocity of the diaphragm of the speaker. (Item 14) 2. The system of claim 1, wherein the audio amplifier includes a second conversion block programmed to convert the corrected current signal into the corrected voltage signal based on the magnetic flux density and the velocity of the diaphragm of the speaker. (Item 15) The audio amplifier, a voice coil power estimation block programmed to determine an average power of the voice coil based on the correction voltage signal; a stationary coil power estimation block programmed to determine an average power of a stationary coil of the loudspeaker based on at least a current in the stationary coil; 3. The system according to claim 1, further comprising: (Item 16) The audio amplifier further comprises: 1. A subtractor circuit comprising: comparing the average power of the voice coil to the average power of the fixed coil; increasing a stationary coil current signal provided to the stationary coil of the speaker when the average power in the voice coil is greater than the average power in the stationary coil; The subtractor circuit is programmed to: 3. The system according to claim 1, further comprising: (Item 17) The audio amplifier, a composite source block programmed to provide a measurement current in a stationary coil of said loudspeaker; a flux transformation circuit programmed to provide a value corresponding to the magnetic flux in the air gap of the voice coil of the speaker based on the measured current in the stationary coil; 3. The system according to claim 1, further comprising: (Item 18) The audio amplifier, a composite source block programmed to provide a measurement current in a stationary coil of said loudspeaker; a pre-processing block programmed to determine an average resistance of the stationary coil based on the measured current of the loudspeaker; a flux transformation circuit programmed to provide a value corresponding to the magnetic flux in the air gap of the voice coil of the speaker based on the average resistance of the stationary coil; 3. The system according to claim 1, further comprising: (Item 19) The audio amplifier, a signal type determination block configured to compare a frequency of the audio input signal against a predetermined frequency; a slow average peak set block configured to provide an output to a flux transformer circuit that provides a value corresponding to the magnetic flux in the air gap of the voice coil of the speaker when the frequency of the audio input signal is below the predetermined frequency, and to eliminate reflected current from the voice coil of the speaker; a fast average peak set block configured to provide said flux transformer circuit with a value that provides an envelope of a current in a fixed coil when said frequency of said audio input signal is higher than said predetermined frequency; a composite source block configured to supply said current to said fixed coil for said fast average in conjunction with a peak set block; 3. The system according to claim 1, further comprising: (Item 20) 1. A computer program product comprising instructions embodied in a non-transitory computer readable medium programmed to amplify an audio input signal, the computer program product comprising: The above command: receiving an audio input signal; generating a target current signal based on the audio input signal; 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 enclosure based on at least the first predicted position of the voice coil; determining a position of a passive radiator based at least on a pressure within the speaker enclosure; determining a value of magnetic flux density of said loudspeaker; generating a second predicted position of the voice coil based on the value of the magnetic flux density, at least the pressure within the speaker, the position of the passive radiator, and the first corrected current signal; The above computer program product causes the computer to perform the above steps. (Summary) In at least another embodiment, an audio amplifier system is provided that includes a memory and an audio amplifier. The audio amplifier includes 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 speaker diaphragm. The audio amplifier is further programmed to generate a correction current signal based on at least the target current signal and an expected position of the speaker voice coil, and to determine the expected position of the speaker voice coil based on a magnetic flux density value that corresponds to a product of a magnetic flux in an air gap of the speaker voice coil and a length of the speaker voice coil wire. [Brief description of the drawings]

[0008] [Figure 1] 1 shows an example of a generally enclosed speaker system. [Diagram 2] In general, various aspects including a transducer are depicted. [Diagram 3] In general, various aspects including a passive radiator are presented. [Figure 4] 1 generally depicts a model of the transducer and passive radiator related elements in a speaker system. [Diagram 5] 1 illustrates generally a system for estimating Kms(x) and Rms(x) in a speaker system, according to one embodiment. [Figure 6] 1 illustrates generally an amplifier system for correcting distortion in a speaker system, according to one embodiment. [Figure 7] 7 illustrates the amplifier system of FIG. 6, further including a core correction block, according to one embodiment. [Figure 8] 1 illustrates a compensation system that functions as a voltage supply for driving a voice coil, according to one embodiment. [Figure 9] 1 illustrates a system that executes a model for providing Kms and Rms, according to one embodiment. [Figure 10] 13 illustrates a system that also implements the model to provide Kms and Rms, according to one embodiment. [Figure 11] 1 illustrates an example of an apparatus for measuring Kms and Rms values, according to one embodiment. [Figure 12] 13 illustrates corresponding values ​​of Rms and Kms as a function of impedance according to one embodiment. [Figure 13] 1 illustrates generally a system for providing Kms and Rms, according to one embodiment. [Figure 14] 1 illustrates generally an implementation of an acoustic transducer device according to one embodiment. [Figure 15] 1 illustrates generally one implementation for calculating magnetic flux density (BL), according to one embodiment. [Figure 16] 1 illustrates generally another implementation for calculating magnetic flux density (BL), according to one embodiment. [Figure 17]1 illustrates generally another implementation of an acoustic transducer device according to an embodiment. [Figure 18] 1 illustrates generally another implementation of an acoustic transducer device according to an embodiment. [Figure 19] 1 illustrates generally another implementation of an acoustic transducer device according to an embodiment. [Figure 20] 1 illustrates generally another implementation of an acoustic transducer device according to an embodiment. [Figure 21] 1 illustrates generally a method performed by an audio amplifier system, according to one embodiment. [Figure 22] 1 illustrates generally a method performed by an audio amplifier system, according to one embodiment. [Figure 23] 1 illustrates generally a method performed by an audio amplifier system, according to one embodiment. [Figure 24] 1 illustrates generally a method performed by an audio amplifier system, according to one embodiment. DETAILED DESCRIPTION OF THE PREFERRED EMBODIMENTS

[0009] Where necessary, detailed embodiments of the present invention are disclosed herein, but it should be understood that the disclosed embodiments are merely examples of the present invention that may be embodied in various and alternative forms. The figures are not necessarily to scale, and some features may be exaggerated or minimized to show details of specific components. Therefore, specific structural and functional details disclosed herein are not intended to be limiting, but should be construed merely as representative criteria for teaching those skilled in the art to variously use the present invention.

[0010] It is 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), and software that cooperate with each other to perform the operation(s) disclosed herein. In addition, such controllers disclosed utilize one or more microprocessors that execute computer programs embodied in non-transitory computer readable media that are programmed to perform any number of functions disclosed. Furthermore, the controller(s) provided herein include a housing and a variety of microprocessors, integrated circuits, and memory devices (e.g., FLASH, random access memory (RAM), read only memory (ROM), electrically programmable read only memory (EPROM), electrically erasable programmable read only memory (EEPROM)) disposed within the housing. The disclosed controller(s) also include hardware-based inputs and outputs for respectively transmitting and receiving data to and from other hardware-based devices as discussed herein.

[0011] As moving coil transducers (or moving coil speakers) increase their acoustic output, such transducers increase their distortion. This fundamental relationship results in transducer size, weight, cost, and inefficiency, all of which are undesirable. This is especially true for transducers used in automotive applications, where all of these performance issues are critical. At the same time, there is an ever-increasing need for systems that can achieve or provide high output, low distortion, and the desired active noise cancellation (ANC), engine order cancellation (EOC), individual sound zones (ISZ), and echo cancellation for voice recognition.

[0012] As a result, sensorless methods exist, such as those described by Klippel, that attempt to minimize transducer distortions through signal processing. This, in turn, if used properly, allows transducer designers to achieve smaller, lighter, lower cost, or more efficient solutions, depending on the desired tradeoffs. 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. Furthermore, these methods often require embedded microcontrollers and digital signal processors (DSPs). Thus, there is a need for low MIP algorithms (e.g., providing relatively low processing requirements) and hardware low-cost methods for correcting nonlinear distortions, such as those provided herein. Furthermore, the solution should be compatible with automotive hardware, which requires relatively low processing requirements.

[0013] Generally, fundamentally, once the control or compensation of the transducer nonlinearity is actively controlled or compensated, the transducer and system designer has flexibility regarding trade-offs that may be required in the speaker. This may improve size, weight, cost, and efficiency depending on the design goal. For example, the embodiments disclosed herein may provide better control over the transducer displacement or out-of-range and the voice coil current, allowing the transducer to be driven closer to its limits and thus provide more output. Furthermore, the embodiments disclosed herein may provide enhanced control over the transducer nonlinear performance and enhance the performance of acoustic algorithms that rely on the transducer linearity or response, such as ANC, RNC, EOC, ISZ, echo cancellation, etc.

[0014] The embodiments disclosed herein (i) are robust and inherently predictable in terms of stability, repeatability, and testability (i.e., not a black box); (ii) are computationally simple due to low to very low MIP, sensorless; (iii) accommodate simple current sensing; and (iv) may operate in a DSP environment, simplifying the algorithms and potentially eliminating the need to adapt the associated embedded controller.

[0015] 1 generally illustrates an example of a closed speaker system 100, according to one embodiment. System 100 generally includes an enclosure 101 that contains a speaker 102 (or transducer) (e.g., an active speaker or main driver) and a passive radiator 104 (or drone cone that does not receive electrical energy in the form of an audio input signal). Enclosure 101 generally represents a typical speaker enclosure for transmitting audio signals, and aspects related to transducer 102 and passive radiator 104 are discussed in more detail below.

[0016] 2 generally illustrates various embodiments including a transducer 102. For example, the transducer 102 generally includes a cone (or diaphragm) 110 and a voice coil 112. A surround (or suspension) 114 is attached to the end of the diaphragm 110. A former 116 surrounds the voice coil 112 and is disposed 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 former 116. A spider 122 surrounds a portion of the former 116.

[0017] Generally, 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 one another, and the audio input signal induces linear motion of the diaphragm 110 in a vertical axis based on the polarity of the audio input signal. The diaphragm 110 is generally flexible and displaces in both directions in the vertical axis in response to a magnetic field transmitted between the voice coil 112 and the magnet 120. The former 116 is attached to the diaphragm 110 and undergoes a displacement similar to that of the diaphragm 110 (or moves along the vertical axis). As a result of 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 spider 122 is generally configured to prevent the diaphragm 110 from moving horizontally while the diaphragm 110 is displaced vertically or linearly in the vertical axis.

[0018] FIG. 3 generally illustrates various aspects including a passive radiator 104. Generally, the passive radiator 104 may include all of the above components including the transducer 102, except for the voice coil 112 and the magnet 120. The passive radiator 104 may use the sound trapped within the enclosure 101 to generate resonance to emit 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 enclosure 101. The passive radiator 104 may be tuned to the enclosure 101 by changing its overall diaphragm mass (including the weight of the diaphragm 110 or cone). When the transducer 102 generates air pressure through linear displacement of the diaphragm 110, such air pressure moves the passive radiator 104.

[0019] 4 shows a model of the transducer 102 and passive radiator 104 related elements in the speaker system 100. In general, by mathematically modeling the behavior of the voice coil 112 (or moving coil of the transducer 102) and other mechanical elements of the speaker system 100, it is possible to calculate non-linear behavior and use amplifiers and signal processing in real time to correct for the non-linear behavior. These aspects are described in more detail herein.

[0020] There are many ways to model a speaker system. However, when, as in this case, the physical elements of the system are well understood in advance, a model that fits the elements may be computationally simplest and easiest to tune. The embodiments disclosed herein attempt to model the physical elements (e.g., transducer 102 and passive radiator 104) and their interactions in speaker system 100 in a manner that can be directly computed and adaptively tuned, and compensate when the elements behave in a nonlinear manner.

[0021] Speaker system 100 generally has four subsystems: (1) transducer 102, which converts an electrical signal from an amplifier (not shown) into a mechanical output (not shown) (e.g., the mechanical output can be thought of as motion, which in turn converts the mechanical output into an acoustic signal); (2) passive radiator 104, which resonates with enclosure 101 and transducer 102 to produce an acoustic output at low frequencies; (3) enclosure 101, which couples passive radiator 104 to transducer 102 (via pressure) and isolates the back pressure of both passive radiator 104 and transducer 102 from the front pressure; and (4) amplifier and signal processing (not shown). Two simplified subsets of speaker system 100 can also be used: a vented system, which replaces the passive radiator 104 with an acoustic mass created using a port in the enclosure 101, and a closed box system, which simply comprises a sealed enclosure without a vented radiator or passive radiator 104. Figure 4 shows the three mechanical subsystems, which are similar to a two-body resonant system.

[0022] In general, the mechanical elements of the transducer 102 can be modeled as a spring with stiffness (e.g., Kms_TD), damping (e.g., Rms_TD), and 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 surround 114 and spider 122 combination. Kms_TD corresponds to the spring stiffness of the surround 114 and spider 122 combination. Similarly, the passive radiator 104 can be modeled as stiffness (e.g., Kms_PR), damping (e.g., Rms_PR), and moving mass (e.g., M_PR). The transducer 102 and the passive radiator 104 can be considered as the two bodies of the system 100. The body coupling force can be modeled by multiplying the pressure of the enclosure 101 (e.g., relative to the ambient pressure outside the enclosure 101) by the surface area of ​​the diaphragm 110 of the transducer 102 (e.g., Sd_TD) and the diaphragm 110 of the passive radiator 104. The compressibility of the air in the enclosure 101 can be modeled as a spring with stiffness of kappa "κ" (i.e., the insulating index of air, approximately 1.4) multiplied by the pressure of the box.

[0023] For the voice coil 112 (or the moving coil of the transducer 102), the driving force F_1 can be modeled by the magnetic field strength (e.g., "B") in the air gap 118 multiplied by the length "L" of the field conductor and the current in the conductor (e.g., the voice coil 112).

number

[0024] Reference frame x 1 (t) is defined relative to the position of the diaphragm 110 of the transducer 102. Similarly, in the reference frame x 2 (t) is defined relative to the position of the diaphragm 110 of the passive radiator 104. 1 The positive direction of (t) is defined as movement into the enclosure 101, x2 The positive direction of (t) is defined as movement away from the enclosure 101 .

[0025] Using the relationship that the force of a moving mass is mass times acceleration, the force of a spring is equal to the distance from rest, the stiffness of the spring, and the force of friction (or damping) is the product of velocity and friction.

[0026] The force on the moving mass of the transducer 102 (eg, MmsTD) can be expressed by:

number

[0027] Similarly, the force on the moving mass of the passive radiator 104 can be expressed as:

number

[0028] It may then generally be necessary to calculate a pressure "P" based on the position of the diaphragm 110 of the transducer 102 and the position of the diaphragm 110 of the passive radiator 104. This may then be accomplished by first calculating the volume of the enclosure 101 (e.g., Vol_1), which may be the volume of the enclosure 101 minus the volume obtained by the displacement of the diaphragm 110 of the transducer 102 and the passive radiator 104 from the rest position (e.g., Vol_0). It is known that the volume of air is proportional to the pressure, etc.

number

[0029] Then, by relating the relative pressure inside the enclosure "p" to the relative volume and pressure outside the enclosure, p_amb (ambient case), the new pressure resulting from the change in volume can be calculated by:

number

[0030] Note that "p" in the free body force diagram (i.e., FIG. 4) is p(x1,x2) in equation (5).

[0031] If Vol_0 is allowed to be the volume of the enclosure 101 with the diaphragm 110 (for both the transducer 102 and the passive radiator 104) at rest, the change in pressure relative to the ambient pressure can be shown by Equation 6 as follows:

[0032] Combining equations (4) and (5) to calculate the pressure within enclosure 101 relative to ambient as a function of X1 and X2 gives:

number

[0033] This system of ordinary differential equations can then describe the motion of the diaphragm 110 (i.e., of the transducer 102 and passive radiator 104) given the driving force from the voice coil 112. However, it does not yet take into account non-linear behavior.

[0034] Due to the shape of the magnetic field in the vicinity of the voice coil 112, BL is a non-linear function of the position X1 of the diaphragm 110 of the speaker 102. There can be several ways to model this aspect, but a simple one is to use an nth order polynomial. For example, the following equation can express BL as a function of the normalized position to the rest position multiplied by the nominal value at the rest position:

number

[0035] Although equation (7) shows a fourth order polynomial, it is recognized that nth order polynomials can be implemented for equation (7). Due to the physical properties of the diaphragm 110 suspension, Kms and Rms are nonlinear functions of the position X1. In terms of BL, Rms and Kms can be expressed as polynomials. The polynomials are decomposed into two sections: a normalized part and a scalar part with X1=0, which corresponds to the rest position. The advantage of this will become apparent in the following refinements.

number

number

[0036] Equations (8) and (9) can be expressed in terms of signal flow as shown in FIG. 5 through a first normalization circuit 130, a second normalization circuit 132, a first multiplier circuit 134, and a second multiplier circuit 136. The cR 4 .x 4 It will be appreciated that Equations (8) and (9) correspond, respectively, to first normalization circuit 130 and second normalization circuit 132. Each of first normalization circuit 130 and second normalization circuit 132 generally includes hardware and software for performing the calculations required by Equations (8) and (9).

[0037] In the case of Rms, it may also be a function of the velocity of the diaphragm 110, which may also be modeled as a polynomial, for example, as follows:

number

[0038] In equation (10), Rms(x) represents the Rms of equation (9).

[0039] These equations can then be solved using a numerical method, such as Euler's method, where the equations are repeated in small steps of time (small compared to the rate of change of position of any variable of system 100). In particular, solving the system of equations 1-10 yields the velocity of diaphragm 110, which is described in more detail below. Current source compensation

[0040] Now that models have been established to estimate the position and velocity of the diaphragm 110 of the transducer 102 and the passive radiator 104, these aspects can be inserted into a system (or audio amplifier system) 150 to correct distortions (see FIG. 6). The system 150 can be implemented as a current source amplifier (or audio amplifier) ​​and generally includes an equalization block 152, a core correction block 154, and a transducer prediction model block 156. The computationally simplest approach is to drive the voice coil 112 using a current source 158. Due to the nature of the current source 158, the system 150 eliminates the effects of resistance and inductance on the current in the voice coil 112, and therefore can be neglected. The current source 158, by definition, provides the desired current regardless of the load. With this approach, it may only be necessary to determine the correction current for the voice coil 112.

[0041] The equalization block 152 generates a current target (or I_target) corresponding to a desired current based on the audio input signal. The transducer model block 160 is generally provided with an input current I_vc (or I corrected) representing the current of the voice coil 112 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) per Equations 2, 3, 6, 7, 8, 9, and 10. The system 150 provides I corrected to the voice coil 112 to move the voice coil 112 to the predicted position 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 Equation 2, Equation 7, Equation 8, Equation 9, and Equation 10. The pressure model block 162 generally implements Equation 6, and the passive radiator model block 164 generally implements Equation 3. Given Kms_TD(X1), BL(x) from the respective polynomial and target current (I_target in equalization block 152), a correction current (e.g., I_current) to compensate for the nonlinearity of Kms_TD(x) and BL(x) can be calculated as follows:

number

[0042] In general, when BL(x) is less than BL(0), with an amount added to offset the force error due to changes in spring stiffness, the target current may be increased proportionately. However, in such a system, the frequency response may be inaccurate because the electrical damping caused by the resistance of the voice coil 112 may be countered by the amplifier 150 (or current source). This aspect can be compensated for by using a fixed equalization filter in the equalization block 152. FIG. 7 represents the amplifier 150 of FIG. 6, and further includes a core correction block 155 that can be improved in later implementations.

[0043] Voltage supply compensation FIG. 8 shows an audio amplifier system 180 that functions as a voltage source for driving the voice coil 112. The system 180 includes a current conversion block 182, an adaptation block 184, and a voltage conversion block 186. The system 180 provides a corrected 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. In general, the system 180 converts a target voltage (from an equalization block, not shown (the target voltage is generated based on the audio input signal)) to a target current (i.e., I_target) via the current conversion block 182. The core correction block 190 corrects the target current to generate a corrected current (i.e., I_corrected). The voltage conversion block 186 converts I_corrected to a corrected voltage (i.e., V_corrected) that is used to drive the voice coil 112. A voltage source amplifier (not shown) applies V_corrected to the voice coil 112. The system 180 ignores the effect of the inductance of the voice coil 112, which generally works if the correction is for the lower frequencies of the system 180. This can be useful because most of the motion and nonlinearity occurs at low frequencies.

[0044] The system 180 also utilizes the predicted velocity of the diaphragm 110 in addition to the diaphragm position X1 (see output from the transducer prediction model block 156). The current conversion block 182 utilizes the velocity of the diaphragm 110 to convert the audio signal (proportional to voltage) to a target current I_target and sends it to the core correction block 190. The voltage conversion block 186 also converts I_corrected to 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 predicted magnetic flux X and length of the air gap 118). The voltage conversion block 186 also requires the predicted BL to convert I_corrected to V_corrected as shown in equation 13 below.

[0045] In general, it is necessary to convert the target voltage (i.e., input to the current conversion block 182) to I_target for use in the transducer prediction model block 156. For example, the movement of the voice coil 112 transmits a current. This generates a voltage proportional to the velocity times "B" times "L". This corresponds to the length of the air gap and can be called 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 maintain a balance across the resistance of the voice coil resistance (e.g., Rvc). The linear target current (i.e., I_corrected) that matches the voice coil current when BL(x) is linear can be calculated as follows:

number

[0046] Similarly, once the target current is corrected as described above, it must be converted back to a corrected voltage (i.e., Vcorrected). Based on the same relationships, this can be done with the following equation:

number

[0047] Voice coil DC resistance variation (Rvc) In a simple approach, the resistance of the voice coil 112 may be assumed to be constant. If the resistance of the voice coil 112 is assumed to be constant, Rvc in equation (13) Avg Rvc nominal is set to I_corrected. Generally, the voice coil is made of copper or aluminum. These materials may experience resistance changes as the corresponding temperature changes. Therefore, to improve the implementation of the voltage supply of the system 180, a thermal model can be used to estimate the temperature rise of the voice coil 112, thereby calculating the temperature corrected resistance of the voice coil 112. The power of the voice coil 112 can be obtained as the current is predicted as I_corrected. There are several thermal models that can be used based on the accuracy. The simplest can be the RC model, where R represents the thermal resistance of the voice coil 112 to the surroundings and C represents the specific heat capacity of the voice coil 112. The RC model can also be solved iteratively using Euler's method.

[0048] One example of Euler's method of iteratively solving the system of equations is described below. By repeatedly looping through the code of the algorithm as shown below, the algorithm solves the various system equations in small time steps, allowing the equations to be moved over the small time steps and considered and treated as linear. For example, a time step of 200uS (for a sample rate of 5kHz) can adequately model a typical loudspeaker. This model may require downsampling or decimation at the input (e.g., audio input, which may be 48KHz, etc.) and Vcorrected output and Icorrected output, which may be 48KHz, and upsampling with an interpolation filter at the output (e.g., Vcorrected output and Icorrected output, which may be 48KHz). This approach may require approximately 5-6 MIPS per channel for a full passive radiator system and a minimum of 1-2 MIPS for a closed box system, for a fixed point full run. * / / / Transducer motion solution: / / dt is defined as the small time step of the sample system X1=X1+Velocity_TD*DT; Force_damping_TD=-Velocity_TD*Rms(X1)_TD; Force_spring_TD=-X1*Kms(X1)_TD Force_pressure_TD=-K*pressure*Sd_TD; Force_motor=BL(X1)*Ivc_corrected; Force_net_TD=Force_damping_TD+Force_spring_TD+Force_pressure_TD+Force_motor; Velocity_TD=Velocity_TD+Force_net_TD / M_TD*DT; Solution of the motion of the passive radiator 104: Force_damping_PR=-Velocity_PR*Rms(X2,Velocity_PR)PR; Force_spring_PR=-X2*Kms(X2)_PR; Force_pressure_PR=k*pressure*Sd_PR; Force_net_PR=Force_damping_PR+Force_spring_PR+Force_pressure_PR; Velocity_PR=Velocity_PR+Force_net_PR / M_PR*DT; X2=X2+Velocity_PR*DT; / / Solution for pressure change in enclosure 101: pressure=p_0*(Sd_TD*X1-Sd_PR*X2) / (VB+SD*X1+Sd_PR*X2); / / Solution for the correction current in voice coil 112: Ivc_corrected=Ivc_target*BL(0) / BL(X1)+(Kms(X1)-Kms(0)*X1 / BL(X1); / / For the voltage source algorithm, you can add the following C code: / / Solution of Ivc_target Ivc_target=(EQ_out-Velocity_TD*BL(X1)) / Rvoice_coil; / / Solution for correction voltage for voice coil 112: V_voicecoil=Ivc_corrected*Rvoice_coil+BL(X1)*Velocity_TD. Variation in Kms and Rms as a result of exercise history

[0049] The model also assumed that Kms and Rms during motion are defined by a single polynomial. In reality, these parameters may vary depending on the "history" of the motion. For example, when the diaphragm 110 is moved with significant speed and displacement, the suspension 114 of the diaphragm 110 may soften. This may change both Rms and Kms.

[0050] As an improvement, the Kms and Rms values ​​can be scaled using estimates of the changing values ​​of Rms(0) and Kms(0) over time. The Kms(x) and Rms(x) polynomials are normalized to the stationary position so that the time-varying parameters can be directly multiplied with the normalized position-varying parameters to determine more accurate Kms and Rms.

[0051] The softening and stiffening of the suspension 114 of the diaphragm 110 as a function of position can be predicted as an average over time that can be modeled as a sum of exponential decays, with the input to the averaging corresponding to the state values ​​of Kms and Rms that would occur if the magnitude of motion were applied indefinitely. This steady-state value of Kms can be expressed as Equation (14), which is a polynomial in the envelope of the changing position:

number

[0052] The exponential decay can take the form:

number

[0053] Next, the average Kms (or Kms Avg ) can be calculated by multiplying equation (15) by equation (14). This average Kms can then be substituted for Kms(0) in equation (8) to obtain:

number

[0054] The same form of equation can be used for the Rms steady state.

number

[0055] In terms of Kms, we can use equations (15) and (17) to relate the steady state Rms to the magnitude of motion. We can then multiply equation (15) by equation (17) to calculate the average Rms. This average Rms can then be substituted for Rms(0) in equation (9) to get:

[0056]

number

[0057] Kms shown in Equation 15 and Equation 16 Avg and Rms Avg obtains the history of predicted positions of the voice coil 112 by averaging X1 over its recent history.

[0058] FIG. 9 illustrates a system 200 that implements a model to provide Kms and Rms. The system 200 may be part of the transducer prediction model block 156. The system 200 generally includes a controller 202 that implements a model to provide 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 generally converts the history of X1 to an average value of Kms (or KMS avg ) (e.g., an average spring model signal). The conversion block 204 is generally configured to convert the history of X1 into an average Rms (or Rms avg ) (e.g., an average damping model signal). System 200 provides Kms as represented by Equation 16 above. System 200 also provides Rms as represented by Equation 18 above. Kms normalization block 206 (or spring normalization block) is programmed to normalize the spring stiffness of speaker 102 to a rest position corresponding to the predicted position of voice coil 112 set to zero. Rms normalization block (or damping normalization block) 208 is programmed to normalize the damping of speaker 102 to a rest position (e.g., predicted position of voice coil 112 set to zero).

[0059] FIG. 10 generally illustrates a system 250 that also implements a model for providing Kms and Rms. The system 250 includes a transformation block 204, a Kms normalization block 206, an Rms normalization block 208, a first multiplier circuit 210, and a second multiplier circuit 212. The transformation block 204 generally includes an envelope block 252, an averaging 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 improves the approximation of Kms_avg and Rms_avg by using different time constants as shown in, but not limited to, equation (15). The improved Kms_avg and Rms_avg may be based on whether the suspension 114 of the diaphragm 110 is soft or hard. In other words, the system 250 takes into account when the amount of movement of the diaphragm 110 increases or decreases compared to the recent movement. This aspect can be implemented by a selection operation that compares Kms_avg or Rms_avg to the 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 softening, and a softening time constant can be selected. Otherwise, if the average value is greater than the steady-state value, the suspension 114 of the diaphragm 110 may be stiffening, and then a stiffening time constant is selected. This is described in more detail below. In general, the transformation block 204 is configured to generate and store any number of predicted positions (e.g., X1) of the voice coil 112 to present a history of generated predicted positions of the voice coil 112. For example, the transformation block 204 is configured to perform integration, peak detection, and / or averaging over a period of time to determine a history of generated predicted positions of the voice coil 112. 6, which is part of the transducer prediction model block 156. It will be appreciated that the system 150 includes a memory (not shown) for storing a history of the generated predicted positions of the voice coil 112.

[0060] At its input, the envelope block 252 receives a position signal (e.g., X1) and presents the envelope of the position signal. The averaging block 254 applies a peak hold to the envelope generated by the envelope block 252. The output of the averaging block 254 represents the short-term average level of motion and therefore the softening potential of the suspension 114 motion. The Kms state block 256 presents a steady state Kms value (e.g., predicted steady state) and the Rms steady state block 258 presents a steady state Rms value (e.g., predicted steady state). The Kms selection circuit 264 and the Rms selection circuit 266 each contain a first or higher order polynomial that converts the average motion to the 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 steady-state Kms and steady-state Rms values, respectively, to model the time-dependent aspects of the softening behavior of the suspension 114 of the diaphragm 110.

[0061] As a refinement, a different exponential decay time constant may better fit the actual behavior of the suspension 114. In particular, the suspension 114 of the diaphragm 110 may soften (as the motion decreases or stops) faster than the suspension 114 stiffens. To model this aspect, the Kms selection circuit 264 compares the steady-state predicted softening (or the steady-state Kms value) to a modeled average of Kms, and if the steady-state Kms value (or predicted softening) is softer than the value of current predicted by the exponential decay (or the modeled average of Kms), then the softening τ is used in the exponential decay (or applied to the Kms exponential decay block 260). On the other hand, if the steady-state Kms value (or predicted softening) is stiffer than the value of current predicted by the exponential decay (or the modeled average of Kms), then the hardening τ is used in the exponential decay (or applied to the Kms exponential decay block 26). Similar aspects apply with respect to the steady-state Rms value (or predicted softening), the modeled average of Kms, comparison with each other, and the use of softening τ and hardening τ as applied by the Rms selection circuit 266 to the Rms exponential decay block 262.

[0062] Providing adaptive Kms and Rms In some cases, it may be possible to measure the current of the voice coil 112 used in connection with the system 180 (e.g., the system 180 acting as a voltage source to drive the voice coil 112 described in connection with FIG. 8) in real time while playing music or other audio. Similarly, it may be possible to measure the voltage of the voice coil 112 used in connection with the system 150 (e.g., the system 150 acting as a current source to drive the voice coil 112) in real time while playing music or other audio. When this is done, there are other implementations that measure the Kms and Rms values, or the average of Kms and Rms over time. FIG. 11 shows an example of such an implementation (or apparatus) 300 for measuring the Kms and Rms values, according to one embodiment. The apparatus 300 includes a first band pass filter 302, a second band pass filter 304, a first low pass filter 306, a second band pass 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 and second bandpass filters 302 and 304 bandpass filter the voltage and current at a predetermined frequency. The first lowpass filter 306 and the second lowpass 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 it to corresponding rms values ​​of the average voltage (e.g., Vvc_avg) 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).

[0063] In one method, 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), and converting the filtered results to rms values ​​and dividing them can provide the average impedance presented by the voice coil 112 at a given frequency. In general, the first polynomial block 312 and the second polynomial block 314 are configured to fit the curves of Kms(z) and Rms(z) as shown in Figure 12. It is recognized that Kms and Rms are part of the transducer model block 160 by Equation 2 and Equation 3, which include KmsTD and RmsTD, and then Kms_PR and Rms_PR, as shown in Figure 6. The transducer model block 160 provides X1 so that the voice coil 112 moves to a predicted position within the speaker 102 (e.g., the predicted position and correction block 154 for the voice coil 112 generates I_corrected, which corresponds to the current supplied to the voice coil 112).

[0064] FIG. 12 illustrates corresponding values ​​of Rms and Kms as a function of impedance, according to one embodiment. In general, many combinations of Rms and Kms may result in the same impedance at a particular frequency. For a speaker 102, the Rms and Kms of the speaker 102 tend to vary in a predictable manner, so 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 illustrated in FIG.

[0065] Therefore, if the impedance at a particular frequency is known, it is possible to predict the values ​​of Kms and Rms. One method of calculating Kms and Rms from the impedance at a particular frequency may involve fitting the respective Kms and Rms curves using polynomials, as shown in Figure 12.

[0066] In some cases, the suspension 114 of the diaphragm 110 may soften faster than can be detected using a measured impedance implementation (e.g., system 300) as described in connection with FIG. 11. This can be due to system 300's reliance on taking time-consuming averages. If this is a problem, a hybrid approach can be used in which the stiffening τ and measured Kms and Rms are selected when the suspension 114 of the diaphragm 110 is stiffening, and the softening τ and estimated steady-state Kms and Rms are selected when the suspension 114 of the diaphragm 110 is softening, as described above in connection with system 250 of FIG. 10.

[0067] FIG. 13 generally illustrates a system 320 for providing Kms and Rms, according to one embodiment. The system 320 includes the device 300 in addition to the first normalization circuit 130, the second normalization circuit 132, the first multiplier circuit 134, and the second multiplier circuit 136, as described above in connection with FIG. 5. As described above, the device 300 provides a measured average of Kms and a measured average of Rms. To model the nonlinear Kms and Rms, including the effects of position and motion history, the average stationary values ​​(e.g., the measured average Kms and the measured average Rms) can be multiplied by the normalized polynomial models of Kms(x1) and Rms(x1) of the first normalization circuit 130 and the second normalization circuit 132, respectively, via the first multiplier circuit 134 and the second multiplier circuit 136, respectively, to provide Kms and Rms.

[0068] It will be appreciated that systems 200, 250, 300, and 320 may be implemented within transducer prediction model block 156 as shown in Figures 6, 7, and 8. It will be appreciated that while systems 200, 250, 300, and 320 in Figures 9, 10, 11, and 13 show an input designated as X1, systems 200, 250, 300, and 320 utilize X1 as feedback as generated by transducer prediction model block 156.

[0069] In some cases, Rms and Kms may have similar behavior with both position and / or history. If this is the case, as a computational simplification, it is possible to only calculate Kms and provide Rms with a scaler that is a multiple of the value of Kms. Similarly, the Kms and Rms of the passive radiator 104 may be appropriately scaled to the Kms and Rms of the transducer 102.

[0070] Providing an adaptation model for the Rdc of the voice coil Furthermore, by selecting a frequency above the resonance of the speaker 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 value that is approximately close to the resistance of the voice coil 112 can be measured by measuring the impedance at mid-band frequencies in a similar manner as performed to determine Kms and Rms. The measured impedance can be close to the DC resistance of the voice coil 112. Since the DC resistance can be a function of temperature, the DC resistance can be used to determine the temperature of the voice coil 112 to determine whether the voice coil 112 requires thermal protection, etc. Thus, the model can be further adapted using the measured DC resistance instead of the voice coil resistance (e.g., Rvc_avg) calculated in equation (13) based on the thermal model.

[0071] For vent systems For various reasons, some speakers are designed with a tuned port or vent instead of a passive radiator. This vent has an acoustic mass that can resonate with the system to produce a low frequency output, similar to the mass of the passive radiator 104. However, the vent may not have an equivalent suspension stiffness Kms_PR of the passive radiator 104. Therefore, to obtain a solution for the motion of the acoustic mass of the vent, the value of Kms_PR can be set to zero. It is a characteristic of the air movement in the vent that friction losses can occur as well, especially at high speeds. Therefore, it may be more accurate to include Rms_vent, which is also a function of speed, as described for the Rms of the transducer 102.

[0072] For closed boxes Some speakers can be designed without passive radiators or tuned vents. Such speakers are frequently referred to as closed box systems. In this case, it is possible to set the location of the passive radiator (x2) or vents to zero. To simplify the calculations, it may not be necessary to solve for the diaphragm motion of the passive radiator since no passive radiator is present in the system.

[0073] Infinite baffle A system in which the transducer 102 is mounted in a very large enclosure 101 (e.g., the back shelf of a vehicle, which may include the entire trunk as the enclosure 101, or a car door that is acoustically open to the outside, etc.) can be considered an infinite baffle system. In this case, p(x1,x2) in equation (2) is set to zero and equation (3) can be neglected.

[0074] Modeling a Passive Radiator The same method for predicting 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 when the suspension of the passive radiator 104 behaves similarly to that of the transducer 102. However, as mentioned above, it may be appropriate 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 bring about improvements. This can be applied to Kms_PR and Rms_PR in equation (2).

[0075] Adaptation for non-permanent magnet based moving coil transducers In a non-permanent magnet based moving coil transducer (or SAM driver), "B" (the magnetic field) is continually changing as the magnetizing current (i.e., the current in the fixed coil) is continually adjusted. The magnetic field effect can be compensated for using either fixed equalization (e.g., similar to that of the current source compensation method described above) if a current source amplifier is used for the voice coil 112, or by dynamic equalization where the frequency response is adjusted based on B as it changes over time if a voltage source amplifier is used. The equalized signal is then further compensated for by multiplying it by B_nominal / B.

[0076] When a nonlinear correction approach is adapted to a SAM driver, a similar method for determining the targeted (or magnetizing) current relative to the current of the fixed coil may be applied. However, the equalization function for the algorithm in the SAM driver may be replaced by that described herein. For example, a current source implementation (e.g., system 150) as described in connection with FIG. 6 above may be used in connection with a current source implementation for a non-permanent magnet based moving coil transducer. However, a voltage source implementation (e.g., system 180) as described in connection with FIG. 8 may be used in connection with a voltage source implementation for a non-permanent magnet based moving coil transducer. This may suggest that the dynamic equalization function of the voltage source implementation of the SAM driver may be replaced with a voltage to target current conversion. Thus, dynamic equalization is no longer required and may in fact be improved by the embodiments disclosed herein.

[0077] The function that multiplies the equalized signal by B_nominal / B can be replaced by replacing BL(0) in equation (7) with BL(I_stationary), which replaces equation (7) with:

number

[0078] SAM drivers provide a function that converts the excitation current to "B" versus the magnetic flux density in the voice coil gap. Simply multiplying this B by the length of the stationary voice coil wire gives L, BL(I_stationary) (i.e. the magnetic flux density and length of the stationary voice coil wire).

[0079] 14 illustrates an implementation of an acoustic transducer device (or audio amplifier system) 500, according to one embodiment. The acoustic transducer device 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 provides a moving coil control signal (e.g., I m ) and fixed coil control signal (e.g., I S ) The transducer 506 includes a magnetic material 512, a diaphragm 514, a former 516, a stationary coil 518, and a moving coil (or voice coil) 520. The moving coil 520 is attached to the former 516.

[0080] The magnetic material 512 is generally toroidal and has a toroidal cavity. The stationary coil 518 is disposed within the cavity. In various embodiments, the magnetic material 512 can be formed from one or more pieces, which allows the stationary coil 518 to be more easily inserted or formed within the cavity. The magnetic material 512 is magnetized in response to a stationary coil signal, thereby generating a magnetic flux in the magnetic material 512. The magnetic material 512 includes a toroidal air gap 536 within a magnetic path 538, with the magnetic flux flowing within and near the air gap 536.

[0081] The magnetic material 512 may be formed from any material capable of being magnetized in the presence of a magnetic field. In various embodiments, the magnetic material 512 may be formed from two or more such materials. In some embodiments, the magnetic material 512 may be formed from laminations. In some embodiments, the laminations may be assembled radially and formed into a wedge shape such that a composite magnetic material is formed with no gaps between the laminations.

[0082] The moving coil 520 is attached to the former 516 and receives a moving coil signal from the control block 503. The diaphragm 514 is attached to the former 516 such that the diaphragm 514 moves with the former 516 and the moving coil 520. The former 516 and the moving coil 520 move within the air gap 536 in response to the moving coil signal and the magnetic flux in the air gap 536. In general, the various components of the acoustic transducer 506 that move with the former 516 may be referred to as moving components. Components that are stationary when the former 516 is moving may be referred to as stationary components. The stationary components of the acoustic transducer 506 generally include the magnetic material 512 and the stationary coil 518.

[0083] In various embodiments, the acoustic transducer 506 can be adapted to vent the air space between the dust cap 532 and the magnetic material 512. For example, an opening can be formed in the magnetic material 512 or an opening can be formed in the former 516 to allow for venting of the air space, thereby reducing or preventing air pressure from affecting the movement of the diaphragm 514.

[0084] 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 apparatus 500 utilizes the first current source 558 and the second current source 560 to replace a voltage source enabling the transducer apparatus 500 suitable for automotive applications.

[0085] Generally, what is needed is a control method for providing current to the fixed coil 518 and the moving coil 520, incorporating the advantages of both voltage and current sources, improving transient response, providing improved latency, accuracy, and proper protection and diagnostics. The control block 503 controls the moving coil signal (e.g., I m) (or voice coil signal) of the moving coil 520. 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) is not dependent on the impedance of the moving coil 520. Thus, the frequency response of the transducer 506 is no longer dependent on the moving coil signal (i.e., current) but is instead fixed. This aspect allows for a single fixed, non-time-varying second order filter 552 to be used to compensate the frequency response.

[0086] Optimal efficiency of the transducer 506, including the power of both the fixed coil 518 and the moving coil 520, is achieved when the power of the fixed coil 518 and the moving coil 520 are balanced (i.e., until the steel of the motor assembly for the transducer 506 begins to saturate and there is no benefit to further increasing the current in the fixed coil 506). In the transducer device 500, the power of the resistor is approximated via a square root circuit 562 of the peak of the audio signal level detected by a peak detector circuit 564 which sets the current in the fixed coil 518 since the power of the resistor is proportional to the square of the current. To directly use the output 504 from the square root circuit 562, a proportional current source (i.e., a second current source 560) is used to drive the fixed coil 518.

[0087] To compensate for the varying sensitivity of the output 504, which is directly proportional to the stationary coil signal (or current in the stationary coil 518), a conversion circuit 554 can be used to calculate the magnetic flux in the air gap 536 of the moving coil 520 as a function B(i). The conversion circuit 554 provides the magnetic flux density (B(i)) to the core compensation block 154. The magnetic flux density generally corresponds to the magnetic flux density in the air gap 536 of the moving coil 520 (or voice coil) as a function of the magnetization or steady-state coil current.

[0088] The transducer prediction model block 156 replaces BL(0) with BL(I_stationary) as shown in equation (7) to calculate BL as described above. FIG. 15 generally illustrates one 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. The 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 (while the moving coil 520 is stationary)) is provided to the multiplier 582, and the position of the moving coil 520 (X1) is provided to the normalization circuit 584. The multiplier 582 obtains the product of BL(0) and the output from the normalization circuit 584, providing the product of magnetic flux density B and the length of the moving coil wire (or BL(x1)). As described above, BL can be obtained by equation 7.

[0089] FIG. 16 generally illustrates a 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. In this case, BL(0) is expressed as BL(I stationary_coil )(or BL(I S )) to provide BL(X1) or BL, which is a function of the fixed coil 518 current and the voice coil (or moving coil) 512 position. For example, "B" from the core compensation block 154 can be replaced by "L", the length of the voice coil (a constant), to provide a new scalar part at x=0, which is a function of the fixed coil 518 current and the voice coil 520 position. This can be written as:

number

[0090] FIG. 17 generally illustrates another implementation of an acoustic transducer device (or audio amplifier system) 600 according to one embodiment. The transducer device 600 generally includes a transducer 506 and a control block 602. The control block 602 generally includes a current conversion block 182 (from FIG. 8), an adaptation block 184 (from FIG. 8), a voltage conversion block 186 (also from FIG. 8), a first voltage source 607, a moving coil power estimation block 610, a fixed coil power estimation block 612, a subtractor circuit 614, a second voltage source 616, a fixed coil modeling block 618, and a conversion circuit 620. Generally, the first voltage source 608 and the second voltage source 616 take the place of the first current source 558 and the second current source 560, respectively, as generally illustrated in connection with FIG. 14. In this case, the filter 552 of the control block 503 of FIG. 14 can no longer be fixed because the impedance of the moving coil 520 may no longer be countered by the first current source 558. With the apparatus 600, the conversion circuit 620 provides an output representing the magnetic flux in the air gap 536 of the moving coil 520 in function B(i). As in FIG. 8, the adaptation block 184 utilizes the output of the conversion circuit 620 to determine the correct BL of the model based on the changing "B" (or magnetic flux density) * length "L" of the voice coil 520 according to Equation 19. The output of the conversion circuit 620 represents BL, which is BL(I_stationary), which represents BL(Is) (or BL(I_stationary)) in Equation 19. In general, 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 of 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 .

[0091] However, since the second voltage source 616 has been replaced by the second current source 560, the target output current 605 is no longer directly proportional to the actual current in the fixed coil 518 due to the impedance of the fixed coil 518. To compensate for this aspect, the fixed coil modeling block 618 (e.g., inductance model) assumes that the resistance of the fixed coil 518 is known. In reality, it is not due to temperature effects that can change the resistance by more than 50%. These errors lead to errors in both frequency compensation and sensitivity compensation in both steady state and transient conditions when the current in the fixed coil 518 is changing.

[0092] It is recognized that optimal efficiency of the transducer 506 may be achieved when the power of the fixed coil 518 and the moving coil 520 are balanced. Therefore, the apparatus 600 utilizes a square root approximation method. For example, the moving coil power estimation block 610 determines the average power of the moving coil 520, and 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 moving coil 520 with the power of the fixed coil 518. If the average power of the moving 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 sounds). This condition causes a decrease in the power to the moving coil 520, balancing the power between the fixed coil 518 and the moving coil 520. When this condition occurs, it essentially requires a slight decrease in efficiency, but generally this condition may be unavoidable since it requires forcing a rapid decrease in current in the stationary coil 518. This aspect may generate an audible sound, so it is more advantageous to continue with the slight decrease in efficiency as a tradeoff. The stationary coil modeling block 618 is generally configured to model the stationary coil inductance behavior. For example, in the implementation of FIG. 17, no current measurements are made. Rather, the stationary coil modeling block 618 is configured to predict the inductance behavior of the stationary coil 518 with a model (e.g., an L / R inductance / resistance model). Additionally, 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.

[0093] However, the average power should be estimated over a fairly long period to avoid distortion, which may be between 0.1 and 1 second. This entails that in transient conditions, when the level of the audio signal increases rapidly, the fixed coil signal (or the current in the fixed coil 518) does not track quickly. As a result, during transient conditions, the sensitivity of the transducer 506 remains low for a long period of time, thus requiring very high transient moving coil 520 amplifier peak power or limiting the output SPL of the transducer 506 during transient conditions. Furthermore, if the level of the input audio signal 502 has a high dynamic content, the slow tracking of the fixed coil signal to the level of the input audio signal 502 may impair efficiency, since the power balance may not be maintained.

[0094] 18 generally illustrates another implementation of an acoustic transducer apparatus (or audio amplifier system) 700 according to one embodiment. The acoustic transducer apparatus 700 includes a transducer 506 and an acoustic transducer controller (or controller) 702. The acoustic transducer controller 702 generally includes at least one digital processor 701 and a memory 703. The digital processor 701 generally performs the functions performed by the controller 702. In response to receiving an input audio signal 502, the controller 702 generates and transmits moving coil and fixed coil signals to the moving coil 520 and fixed coil 518, respectively.

[0095] The apparatus 700 is generally configured to balance power between the fixed coil 518 and the moving coil 520 while achieving fast transient response without introducing distortion and without relying on a current source to generate the fixed coil and moving coil signals, improve accuracy of frequency and sensitivity compensation in the presence of a changing current for the fixed coil 518 (i.e., a changing fixed coil signal), and improve efficiency of the transducer 506. Additionally, as will be described in more detail, the apparatus 700 is generally configured to provide protection and diagnostics for electronics used in association with the fixed coil 518.

[0096] The controller 702 includes a current conversion block 182, an adaptation block 184, a voltage conversion block 186, a voltage source 708, a composite source block 710, a conversion circuit 712, and a peak set block 714. The composite source block 710 is provided to replace the second current source 560 as described in connection with Figure 14 and to replace the second voltage source 616 as described in connection with Figure 17. Generally, the composite source block 710 is configured for control or regulation of its impedance and generates a stationary coil signal for transmission to the stationary coil 518.

[0097] The average in the peak set block 714 takes the peak values ​​of the input audio signal 502 and uses a low pass filter to remove the ripple associated with a simple peak detector (similar to the peak detector circuit 564 of FIG. 14). During transient conditions, the slowly changing low pass filter is forced to respond immediately to the transient condition by setting the value of the low pass filter directly to the instantaneous maximum absolute value of the input audio signal. In this way, a minimal ripple input audio signal, a clean envelope of the input audio signal 502, can be produced that can respond to transient increases in the level of the audio signal and is provided as the reference signal 705 to the composite source block 710.

[0098] In this device 700, the level of the input audio signal 502 can be used without the square root circuit 562 (or square root function) (of FIG. 14), since the output voltage of the source of the fixed coil 518 is a known voltage whose power is proportional to the square of the voltage. This means that the power of the moving coil 520 is proportional to the power of the fixed coil 518, and the power of the moving coil 520 is proportional to the square of the current of the fixed coil 518, since the fixed coil signal (or the current provided to the fixed coil 518) is proportional to the input audio signal 502 via the reference signal 505. Since this approach ignores the effect of the frequency-dependent impedance of the moving coil 520, the above approach may not be as accurate as the power balance performed by the moving coil power estimation block 610 and the fixed coil power estimation block 612. However, for music and noise signals, an average scaling value can be selected to adequately approximate the effect of the impedance of the moving coil 520 with music and noise.

[0099] The composite source block 710 measures the current (or stationary coil signal) provided to the stationary coil 518 using a current measurement circuit 707. The current measurement circuit 707 may be a resistor, a current transformer, a Hall effect sensor, or the like. The measured current (i.e., the stationary coil measurement signal) is provided as feedback to the compensation block 720, which provides an error signal to a summer circuit 722. The summer circuit 722 compares a reference signal 705 to 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 may be implemented as a pulse width modulated ("PWM") (or other modulation scheme) buck (or other topology) regulator along with a filter 725. The filter 725 generally includes an inductor 726 and a capacitor 728 for filtering the voltage output from the voltage source 724. The compensation block 720 and filter 725 generally output an impedance such that the composite source block 710 appears to be a current source, a voltage source, or a mixed frequency based source if desired. In particular, it may be desirable for the composite source block 710 to behave as a current source at low frequencies and as a voltage source at frequencies above the mechanical resonance of the transducer 106 (e.g., 50-100 Hz for a 6 inch intermediate bus driver). This aspect can improve distortion in the passband of the transducer 506 while providing precise control over the average fixed coil current (or average fixed coil signal) and transient levels.

[0100] To achieve the impedance behavior in the composite source block 710, the compensation block 720 may be implemented, for example, as a proportional-integral-derivative (PID) controller. For example, the compensation block 720 may include a proportional path with gain "Kp" in the current feedback path where the current of the steady coil signal is measured by the current measurement circuit 707. The integral and derivative terms (i.e., Ki and Kd) may be, for example, zero. Using a proportional current feedback K (i.e., Ki and Kd=0) is sufficient for the filter 725. The integral term Ki and the derivative term Kd are stable because the second order system created by the inductor 726 and the capacitor 728 is reduced to a first order system by the measurement of the current by the current measurement circuit 707 and the proportional current feedback Kp. By using the proportional current feedback Kp in the feedback path, this condition creates an effective current source.

[0101] In this arrangement, the inductance of the inductor 726 is effectively eliminated (in a stability sense) by the current source created by using current feedback. By selecting an appropriate gain Kp in the feedback path of the compensation block 720, the frequency at which the natural impedance of the capacitor 728 affects the output impedance can be adjusted. The higher the gain of Kp, the higher the frequency. At high frequencies, the impedance presented by the composite source block 710 is dominated by the impedance of the capacitor 728 and therefore looks like a voltage source. For this to be true, the size of the capacitance of the capacitor 728 should be sufficient so that at the desired frequency above the resonance of the transducer 506, the impedance of the capacitor 528 is similar to or smaller than the impedance of the transducer 506. At low frequencies, where the impedance of the capacitor 728 is high, the output current is dominated by the effective current source created using current feedback. Thus, the 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. The higher frequency is typically 3-5 times the mechanical resonance of the transducer 506, and the low frequency is typically any frequency below the high frequency. Finally, it should be recognized that this same effect can be achieved with other control techniques, such as, for example, using voltage sensing and adding an integral term Ki and a proportional term Kp for stability, and possibly a derivative term Kd. The foregoing may be expressed in the s-domain or the z-domain.

[0102] Furthermore, in a system where multiple stationary coils 518 are supplied with current, it is possible to connect the loads of the stationary coils 518 in parallel with each other and use one control loop and voltage supply. However, to make the device fail-safe at the input between the controller 702 and the stationary coils 518, the current measured by the current measurement circuit 707 in the feedback path described above may be the higher of the multiple currents in the stationary coils 518 at any instant. In this way, the current in the stationary coils 518 is regulated to the load of the stationary coils 518 that provides the maximum current.

[0103] In general, the level of the current in the fixed coil 518 to optimize the efficiency of the device 700 is generally determined by the peak set block 714. The peak set block 714 receives the input audio signal 502. The device encourages to avoid large fluctuations in the desired current in the fixed coil 518 near the resonance of the transducer 506. At near resonance, less power is needed to produce the same acoustic output level. For this reason, the device 500 and the device 600 may generally result in the current in the fixed coil 518 being reduced at resonance to balance the power. However, as the current in the fixed coil 518 is reduced, the damping is reduced and even less power is needed for the moving coil 520, which leads to a further reduction in the current in the fixed coil 518. This result may lead to errors near resonance in both sensitivity and frequency response, because the transducer 506 may be almost entirely damped by its mechanical losses. Thus, by providing the input audio signal 502 to the peak set block 714, this condition avoids the above errors. This may entail that the power balance between the fixed coil 518 and the moving coil 520 may not be maintained near resonance, but this aspect may not be important because the power levels of the fixed coil 518 and the moving coil 520 are low near resonance.

[0104] The conversion circuit 712 can receive the measured current of the fixed coil 518 (i.e., the fixed coil signal) to determine the magnetic flux density in the air gap 536. The determined magnetic flux density in the air gap 536 is used to determine the changing acoustic frequency response and acoustic sensitivity of the transducer 506 as a function of the current in the fixed coil 518. If the measured current of the fixed coil 518 is used to determine the magnetic flux in the air gap 536, the adaptation block 184 can correct the sensitivity of the fixed coil 518. However, if the measured current of the moving coil 520 is used to directly determine the magnetic flux in the air gap 536, and therefore the sensitivity and frequency response, distortions can occur at some frequencies and levels.

[0105] Generally, the stationary coil 518 replaces a conventional magnet that is typically used to generate a magnetic flux in the air gap 536 to enable the transducer 506 to output audio. However, the stationary coil 518 utilizes a large amount of current when the transducer 506 outputs high peaks in audio (i.e., drum rolls, etc.). Thus, 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 a high level of audio does not need to be output, and increases the current on Is when a high level of audio needs to be output (i.e., dynamically adjusts the current).

[0106] The composite 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 with a function B(i). The adaptation block 184 utilizes the output of the conversion circuit 712 to determine the correct BL of the model based on the changing "B" (or magnetic flux) via Equation 19. The voltage conversion block 186 generates V_corrected, which is provided as an input to the moving coil 520 via the voltage source 708. The adaptation block 184 uses the value of the magnetic flux to provide the same frequency response to the input audio signal. The composite source 710 has the impedance characteristics of a voltage source or a current source. The composite source 710 raises the current Is quickly and quietly when the audio input signal has a large level.

[0107] The stationary coil 518 and the moving coil 520 are coupled to a transformer through the magnetic material 512. As a result, the current in the moving coil 520 generates a transformer coupling or reflected current in the stationary coil 518. This distortion is more prevalent or noticeable at frequencies and signal levels where the reflected current in the moving coil 520 is large compared to the average of the current in the stationary coil 518. When the stationary coil current is measured with the current measurement circuit 707, the measurement may include a current reflected from the current in the moving coil 520. However, the controller 702 may use the measured current to determine the acoustic sensitivity of the transducer 506. If the phase relationship of the current in 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 in the stationary coil 518, thereby causing the conversion circuit 712 to calculate a lower magnetic flux density in the gap 536, resulting in reduced sensitivity. This embodiment reflects more current in the fixed coil 518 being further subtracted from the average current in the fixed coil 518, causing the conversion circuit 712 to calculate a lower magnetic flux density in the gap 536, ultimately increasing the current in the moving coil 520. Thus, a positive feedback is established which creates the aforementioned distortion.

[0108] In the opposite phase, the moving coil 520 current adds to the average of the fixed coil 518 current, causing the conversion circuit 712 to calculate a higher magnetic flux density in the gap 536. This in turn provides the same positive feedback that results in distortion. As a result, at some frequencies, the resulting output signal becomes asymmetrically distorted with even higher order distortion components. In one aspect, it may be advantageous to isolate the effect of the moving coil 520 current reflected back to the fixed coil 518, which is used to determine the sensitivity compensation and frequency compensation.

[0109] 19 generally illustrates another implementation of an acoustic transducer device (or audio amplifier system) 800 according to another embodiment. The acoustic transducer device 800 includes a transducer 506 and an acoustic transducer controller (or controller) 802. The acoustic transducer controller 802 generally includes at least one digital processor 801 and a memory 803. The digital processor 801 generally 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.

[0110] The controller 802 includes a current conversion block 182, an adaptation block 184, a voltage conversion block 186, a voltage source 808, a composite source block 710, a conversion circuit 712, and a peak set block 714, as well as a pre-processing block 804. In the apparatus 800, the measured current in the stationary coil 518 is not used directly to determine the magnetic flux in the air gap 536 of the moving coil 520. Rather, the pre-processing block 804 pre-processes the measured current in the stationary coil 518. For example, the pre-processing block 804 takes the long-term average voltage amplitude of the stationary coil 518 measured by the current measurement circuit 707 and determines the average resistance of the stationary coil 518. The average resistance of the stationary coil 518 is used in an L / R model of the stationary coil 518 to predict the effective average current in the stationary coil 518, which is free of reflected current from the moving coil 520. The L / R model is similar to the model in the fixed coil modeling block 618 (see FIG. 17), except that the resistance R of the fixed coil 518 is measured and therefore more accurately included. In this case, the resistance and internal temperature of the moving coil 520 are known on an accurate basis which aids in predicting the effective average current in the fixed coil 518. It is recognized that when the current in the fixed 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 fixed coil 518, and may include all of the non-ideal aspects of the inductor, such as residual magnetism, saturation, and other effects of the steel.

[0111] The pre-processing block 804 is configured to obtain a fast average with a peak set function of the measured current. The pre-processing block 804 first obtains a peak detection value of the measured current of the fixed coil 518, and then utilizes a low-pass filter therein to average the peak detection value. 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 and fast rising current of the fixed coil 518, the value of the low-pass filter can be forced to a peak value during the fast rising of the steady current. This can be optimal to eliminate the distortion problem noted above, except for pure sine waves with frequencies below the cutoff frequency of the low-pass filter, when the filter cannot estimate the average steady current.

[0112] 18 above, the control block 710 provides the output current Is to a transformation circuit 712, which provides a value corresponding to the magnetic flux in the air gap 536 of the moving coil 520 with a function B(i). The adaptation block 184 utilizes the output of the transformation circuit 712 to determine the correct BL for the model based on the changing "B" (or magnetic flux) via Equation 19. The voltage transformation block 186 generates V_corrected, which is provided as an input to the moving coil 520 via a voltage supply 808.

[0113] 20 generally illustrates an acoustic transducer device (or audio amplifier system) 900 according to another embodiment. The acoustic transducer device 900 includes a transducer 506 and an acoustic transducer controller (or controller) 902. The acoustic transducer controller 902 generally includes at least one digital processor 901 and a memory 903. The digital processor 901 generally performs the functions performed by the controller 902. In response to receiving an input audio signal 502, the controller 902 generates and transmits a moving coil signal (e.g., Im) and a fixed coil signal (e.g., Is) to the moving coil 520 and the fixed coil 518, respectively.

[0114] The controller 902 includes a current conversion block 182, an adaptation block 184, a voltage conversion block 186, a voltage source 908, a composite source block 910, a conversion circuit 912, a peak set block 1014, a pre-processing block 954, a delay block 904, a signal-based discriminator block 906, and a signal scaling block 911. A feedback path 923 and a feedforward path 924 are shown as providing inputs to the pre-processing block 954. For example, the pre-processing block 954 includes a peak set block (or slow average with peak set block) 914, a switch 925, and a fast average peak set block 916. The signal-based discriminator block 906 is provided to select the switch 925 so that the feedback path 923 or the feedforward path 924 is selected to provide an input to the conversion circuit 912 coming from either the peak set block 914 or the fast average peak set block 916.

[0115] The signal type discriminator block 906 determines when the input audio signal 502 is below the cutoff frequency (or a predetermined frequency) of the low pass filter of the pre-processing block 954 or is predominantly sinusoidal in nature (e.g., a single tone or test signal of a single frequency). If this condition is true (i.e., the input audio signal 502 is sinusoidal), the peak set block 914 can use the switch 925 in the feed forward path 924 as an input to the conversion circuit 912. As described above, the peak set block 914 provides a target current for the fixed coil 518. This mode eliminates the effect of the reflected current in the moving coil 520 by eliminating the feedback path 923 since no feedback path is used. Additionally, the peak set block 914 includes a fast peak set function in the same manner as the fast average peak set block 916 does, allowing fast transients to set the output of the block 514 to eliminate the delay associated with the averaging filter.

[0116] If the signal type discriminator block 906 determines that the input audio signal 502 exceeds 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), the fast average peak set block 916 can be used in the feedback path 923 using the switch 925 to provide an output from the fast average by the peak set block 916 to the conversion circuit 912. The fast average by the peak set block 916 functions similarly to the average in the peak set block 714. However, the cutoff frequency of the averaging low pass filter of the fast average peak set block 916 is comparable to the resonance of the transducer 506. This replicates the operation of the pre-processing block 954 with the input audio signal. The fast average peak set block 916 provides the envelope of the current through the fixed coil 518 if the envelope corresponds to the rapid changes in the measured current. The composite source block 910 is configured to provide the current of the fixed coil 518 to the fast average peak set block 916.

[0117] The signal scaling block 911 scales the level of the target current in the fixed coil 518 based on the nature of the input audio signal 502 detected by the signal type discriminator block 906. In this way, an optimal current in the fixed coil 518 is provided to balance the power of the sine wave and the various optimal currents of the noise or music signal that can be well maintained, with the sine wave having a lower peak-to-average value than the noise or music. Furthermore, the delay block 904 provides additional time for the current of the fixed coil signal to rise to the target current in the fixed coil 518, especially during fast transients.

[0118] The magnitude of the delay used by delay block 904 may depend on the voltage available to drive the current in stationary coil 518, as determined by the power electronics employed, such as the power supply for voltage supply 724, the inductance and resistance of stationary coil 518, the bandwidth of device 900 and therefore the slew rate of the transients being reproduced, and secondary factors such as amplifier headroom. In some cases, no delay may be necessary.

[0119] The adaptation block 184 utilizes the output of the transformation circuit 912 to determine the correct BL for the model based on the changing "B" (or magnetic flux) via Equation 19. The adaptation block 184 provides the correct 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 via a voltage supply 808. The voltage transformation block 186 generates V_corrected in response to the correct BL.

[0120] FIG. 21 generally illustrates a method 1000 performed by an audio amplifier system 150, according to one embodiment.

[0121] 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.

[0122] In operation 1010, the audio amplifier system 150 determines a pressure inside the speaker enclosure 101. In operation 1012, the audio amplifier system 150 determines a position of the passive radiator 104 based on at least the pressure inside the speaker enclosure 101. In operation 1014, the audio amplifier system 150 generates a second predicted position of the voice coil 112 based on at least the pressure inside the speaker 102, the position of the passive radiator 104, and the first corrected current signal.

[0123] FIG. 22 generally illustrates a method 1020 performed by an audio amplifier system 180, according to one embodiment.

[0124] In operation 1022, the audio amplifier system 180 generates a target current signal based on the audio input signal and a velocity of the diaphragm of the speaker 102. In operation 1024, the audio amplifier system 180 generates a target current signal based on the audio input signal and a velocity of the diaphragm 110 of the speaker 102. In operation 1026, the audio amplifier system 180 generates a first corrected current signal based on the at least one target current signal and based on a first predicted position of the voice coil 112 of the speaker 102.

[0125] FIG. 23 generally illustrates a method 1040 performed by an audio amplifier system 500 according to another embodiment.

[0126] 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 inside the speaker enclosure 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 one of the first predicted positions of the voice coil 520. In operation 1052, the audio amplifier system 500 determines a value of the magnetic flux density of the speaker 506. In operation 1054, the audio amplifier system 500 generates a position of the voice coil 520.

[0127] FIG. 24 generally illustrates a method 1060 performed by any one of the audio amplifier systems 600, 700, 800, and 900.

[0128] In operation 1062, any one or more of the systems 600, 700, 800, and 900 generate an audio input signal. In operation 1064, any one or more of the systems 600, 700, 800, and 900 generate 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 generate a corrected current signal based on at least the target current signal and 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 determine a predicted position of the voice coil 520 of the speaker 506 based on the value of the magnetic flux density.

[0129] Although exemplary embodiments have been described above, it is not intended that these embodiments describe all possible forms of the present invention. Rather, the words used herein are words of description rather than of limitation, and it is understood that various changes may be made without departing from the spirit and scope of the present invention. In addition, features of the embodiments may be combined in various implementations to form further embodiments of the present invention.

Claims

1. 1. An audio amplifier system comprising: Memory, an audio amplifier; the audio amplifier includes the memory; Receiving an audio input signal; generating a target current signal based on the audio input signal; 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 enclosure based on at least the first predicted position of the voice coil; determining a position of a passive radiator based at least on the pressure within the speaker enclosure; determining a value of magnetic flux density of the loudspeaker; generating a second predicted position of the voice coil based on the value of the magnetic flux density, at least the pressure within the speaker enclosure, the position of the passive radiator, and the first corrected current signal; It is programmed to Audio amplifier system.

2. 2. The system of claim 1, wherein the magnetic flux density value corresponds to the product of an air gap magnetic flux in the speaker and a length of a voice coil wire in the speaker.

3. the audio amplifier further comprising: generating a second correction current signal based on the target current signal and the second predicted position of the voice coil; transmitting the second correction current signal to the speaker to control a position of the voice coil based on the second correction current signal; The system of claim 1 , programmed to:

4. 2. The system of claim 1, wherein the audio amplifier includes a transform block programmed to store any number of generated predicted positions of the voice coil and provide a history of the generated predicted positions of the voice coil.

5. The transformation block further comprises: generating an average spring model signal corresponding to a spring stiffness of the loudspeaker based on the history of the generated predicted positions of the voice coil; generating an average damping model signal corresponding to damping of the loudspeaker based on the history of the generated predicted positions of the voice coil; The system of claim 4 , programmed to:

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

7. The system of claim 5 , wherein the attenuation of the speaker corresponds to friction losses in the speaker surround and spider.

8. The audio amplifier, a spring normalization block programmed to normalize the spring stiffness of the speaker to a rest position corresponding to a predicted position of the voice coil which is set to zero; an attenuation normalization block programmed to normalize the attenuation of the speaker to the rest position; The system of claim 5 , comprising:

9. a first plurality of filters programmed 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; a second plurality of filters programmed 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; a divider circuit programmed to generate an impedance signal indicative of an impedance of the voice coil based on the first filter output and the second filter output; The system of claim 1 further comprising:

10. a first polynomial block programmed to determine a spring stiffness of the loudspeaker based on the impedance signal; a second polynomial block programmed to determine an attenuation of the speaker based on the impedance signal; The system of claim 9 further comprising:

11. 1. An audio amplifier system comprising: Memory, an audio amplifier; the audio amplifier includes the memory; Receiving an audio input signal; generating a target current signal based on the audio input signal and a velocity of a speaker diaphragm; generating a correction current signal based on at least the target current signal and an expected position of a voice coil of the loudspeaker; determining the predicted position of the voice coil of the speaker based on the value of magnetic flux density; It is programmed to An audio amplifier system, wherein the magnetic flux density value corresponds to the product of the magnetic flux in the air gap of the voice coil of the speaker and the length of the voice coil wire of the speaker.

12. 12. The system of claim 11, wherein the audio amplifier is further programmed to provide a correction voltage signal to the speaker to control a position of the voice coil based at least on the predicted position of the voice coil.

13. 13. The system of claim 12, wherein the audio amplifier includes a first conversion block programmed to convert a voltage corresponding to the audio input signal into the target current signal based on the audio input signal and the velocity of the diaphragm of the speaker.

14. 13. The system of claim 12, wherein the audio amplifier includes a second conversion block programmed to convert the corrected current signal to the corrected voltage signal based on the value of the magnetic flux density and the velocity of the diaphragm of the speaker.

15. The audio amplifier, a voice coil power estimation block programmed to determine an average power of the voice coil based on the correction voltage signal; a stationary coil power estimation block programmed to determine an average power of a stationary coil of the loudspeaker based on at least a current in the stationary coil; The system of claim 12 , comprising:

16. The audio amplifier further comprises:

1. A subtractor circuit comprising: comparing the average power of the voice coil to the average power of the fixed coil; increasing a stationary coil current signal provided to the stationary coil of the loudspeaker when the average power in the voice coil is greater than the average power in the stationary coil; a subtractor circuit programmed to The system of claim 15 , comprising:

17. The audio amplifier, a composite source block programmed to provide a measurement current in a stationary coil of the loudspeaker; and a flux transformation circuit programmed to provide a value corresponding to the magnetic flux in the air gap of the voice coil of the speaker based on the measured current in the stationary coil.

18. The audio amplifier, a composite source block programmed to provide a measurement current in a stationary coil of the loudspeaker; a pre-processing block programmed to determine an average resistance of the stationary coil based on the measured current of the speaker; a flux transformation circuit programmed to provide a value corresponding to the magnetic flux in the air gap of the voice coil of the loudspeaker based on the average resistance of the stationary coil; The system of claim 12 , comprising:

19. The audio amplifier, a signal type determination block configured to compare a frequency of the audio input signal against a predetermined frequency; a slow average peak set block configured to provide an output to a flux transformer circuit that provides a value corresponding to the magnetic flux in the air gap of the voice coil of the speaker when the frequency of the audio input signal is below the predetermined frequency, and to eliminate reflected current from the voice coil of the speaker; a fast average peak set block configured to provide an output to the flux transformer circuit that provides a value that provides an envelope of a current in a fixed coil when the frequency of the audio input signal is greater than the predetermined frequency; a composite source block configured to supply the current in the fixed coil to the fast average peak set block; The system of claim 12 , comprising:

20. 1. A computer program for amplifying an audio input signal, the computer program, when executed by a processor, causing the processor to: Receiving an audio input signal; generating a target current signal based on the audio input signal; 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 enclosure based on at least the first predicted position of the voice coil; determining a position of a passive radiator based at least on the pressure within the speaker enclosure; determining a value of magnetic flux density of the loudspeaker; generating a second predicted position of the voice coil based on the value of the magnetic flux density, at least the pressure within the speaker enclosure, the position of the passive radiator, and the first corrected current signal; A computer program for causing a computer to carry out a method comprising the steps of:

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