Patent Publication Number: US-9900691-B2

Title: Method and circuitry for protecting an electromechanical system

Description:
CROSS REFERENCE TO RELATED APPLICATION 
     This claims the benefit of commonly-assigned U.S. Provisional Patent Application No. 62/147,282, filed Apr. 14, 2015, which is hereby incorporated by reference herein in its entirety. 
    
    
     FIELD OF USE 
     Implementations of the subject matter of this disclosure generally pertain to apparatus and methods for protecting electromechanical systems from damage caused by being overdriven. In particular, implementations of the subject matter of this disclosure pertain to apparatus and methods for protecting speakers. 
     BACKGROUND 
     The background description provided herein is for the purpose of generally presenting the context of the disclosure. Work of the inventors hereof, to the extent the work is described in this background section, as well as aspects of the description that may not otherwise qualify as prior art at the time of filing, are neither expressly nor impliedly admitted to be prior art against the present disclosure. 
     Certain kinds of electromechanical systems are susceptible to damage if overdriven. For example, a loudspeaker may be damaged if overdriving causes the speaker membrane or cone, or the voice coil itself, to move beyond its designed excursion limit. Another source of potential damage to a loudspeaker may arise from high temperatures, which might result from ohmic heating. Such temperatures could cause adhesives used in the loudspeaker to melt, and could also cause the speaker membrane or cone to become brittle and ultimately fail. 
     As another example, an electric motor may be damaged if overdriving causes the motor to exceed its designed rotational speed limit. Similarly, stress resulting from frequent substantial speed changes could cause mechanical failure (e.g., of motor bearings). 
     Known techniques for controlling overdriving of electromechanical systems are either mechanically complex (which also results in greater cost), or computationally complex. 
     SUMMARY 
     A method, according to implementations of the subject matter of this disclosure, for limiting motion in an electromechanical system, includes filtering an input signal using an adaptive filter to yield a predicted motion, and attenuating the input signal by an amount controlled by the predicted motion. 
     The filtering may be performed using an adaptive infinite impulse response filter. 
     The filtering may further yield a predicted temperature, and the amount of attenuating may be further controlled by the predicted temperature. The attenuating may include clamping the input signal at a predetermined amplitude when the predicted temperature exceeds a threshold. 
     The electromechanical system may be a loudspeaker, in which case the motion may be displacement of a transducer of the loudspeaker, and the attenuating may include removing components of the input signal at selected frequencies. The attenuating may includes mixing a portion of the input signal from which the components have been removed with a portion of the input signal from which the components have not been removed. The mixing may be performed according to a combination of more than one mixing function. 
     The method may further include equalizing the portion of the input signal from which the components have not been removed with the portion of the input signal from which the components have been removed. The removing of components of the input signal at selected frequencies may include applying a notch filter, and the equalizing may include phase-adjusting the portion of the input signal from which the components have not been removed to account for phase delay introduced by the notch filter. Removing components of the input signal at selected frequencies may include applying a notch filter. The notch filter may operate at a resonant frequency of the loudspeaker. 
     The electromechanical system may be a motor, and the motion may be the rotational speed of the motor. 
     Circuitry, according to implementations of the subject matter of this disclosure, for limiting motion in an electromechanical system, may include an adaptive filter to yield a predicted motion from an input signal, control circuitry to attenuate the input signal by an amount controlled by the predicted motion. The adaptive filter may be an adaptive infinite impulse response filter. 
     The adaptive filter may further yield a predicted temperature, and the control circuitry may further attenuate the input signal based on the predicted temperature. The control circuitry may include a clamping circuit to clamp the input signal at a predetermined amplitude when the predicted temperature exceeds a threshold. 
     The electromechanical system may be a loudspeaker, in which case the motion may be displacement of a transducer of the loudspeaker, and the control circuitry may include a notch filter to remove components of the loudspeaker input signal at selected frequencies. The selected frequencies may be centered on a resonant frequency of the loudspeaker. The control circuitry may include a mixer to mix a portion of the input signal that passes through the notch filter with a portion of the input signal that does not pass through the notch filter. The mixer may operate according to a combination of more than one mixing function. 
     Circuitry according to implementations of the subject matter of this disclosure may further include a path equalizer to phase-adjust the portion of the input signal that does not pass through the notch filter to match phase delay introduced by the notch filter. 
     The electromechanical system may be a motor, in which case the motion may be the rotational speed of the motor. 
    
    
     
       BRIEF DESCRIPTION OF THE DRAWINGS 
       Further features of the disclosure, its nature and various advantages, will be apparent upon consideration of the following detailed description, taken in conjunction with the accompanying drawings, in which like reference characters refer to like parts throughout, and in which: 
         FIG. 1  shows a simplified representation of a loudspeaker system with which implementations of the subject matter of this disclosure may be used; 
         FIG. 2  is a graphical representation of voice coil displacement as a function of input voltage for a representative loudspeaker; 
         FIG. 3  is a representation of a lumped parameter model of a loudspeaker; 
         FIG. 4  is a schematic representation of an electromechanical protection system incorporating implementations of the subject matter of this disclosure; 
         FIG. 5  is a schematic representation of a loudspeaker protection system incorporating implementations of the subject matter of this disclosure; 
         FIG. 6  is a schematic representation of a loudspeaker protection system such as that of  FIG. 5 , with some components shown in more detail than in  FIG. 5 , and some components shown in less detail than in  FIG. 5 : 
         FIG. 7  shows an example of a frequency attenuation control transfer function that may be used in implementations of the subject matter of this disclosure; and 
         FIG. 8  shows an example of an amplitude attenuation control transfer function that may be used in implementations of the subject matter of this disclosure. 
     
    
    
     DETAILED DESCRIPTION 
     As noted above, electromechanical systems such as loudspeakers and electric motors are susceptible to damage by overdriving. Implementations of the subject matter of this disclosure may be used to control the driving signals of an electromechanical system to minimize such damage. Although the subject matter of this disclosure may be useful for different types of electromechanical systems, the description which follows will focus, for ease of discussion, on loudspeakers. However, that focus is not meant to limit the scope of this disclosure. 
     A simplified representation of a loudspeaker system  100  is shown in  FIG. 1 . Loudspeaker  101  includes a transducer such as a voice coil  111  and a membrane/cone  121 , and is driven by an amplified signal  122  from amplifier circuitry  102 , based on an input signal  112 . Input signal  112  is indicated as a time-varying voltage V in (t), but also may be characterized as a time-varying current (not shown). Under the control of amplified signal  122 , voice coil  111  causes membrane/cone  121  to vibrate, reproducing sound. The vibrations may be characterized as a time-varying physical displacement x(t), which can occur in both directions from a resting position. 
     If the displacement is too great (resulting from the loudspeaker being overdriven), which is sometimes referred to as “over-excursion,” the displacement can actually cause physical damage to components of the loudspeaker. This is shown in  FIG. 2 , which shows positive and negative voice coil displacement x(t) as a function  200  of input voltage V in (t) for a representative loudspeaker. In region  201 , at low input voltages, the relationship of displacement to voltage is approximately linear. As the absolute value of the voltage increases, the absolute value of displacement becomes nonlinear and increases more slowly than the voltage. Loudspeaker damage begins to occur at a displacement that varies depending on parameters (including size) of the particular loudspeaker. For the example in  FIG. 2 , over-excursion regions  202  begin at a displacement of about ±0.25 mm from the resting position. 
     Known techniques for preventing damaging over-excursion of a loudspeaker suffer from various disadvantages. For example, in a first technique, the actual voice coil displacement is directly measured. Although this first technique provides an accurate measurement of the voice coil displacement, with low computational overhead, this first technique requires expensive and electromechanically complex hardware. 
     According to a second technique, the driving voltage and/or current are measured, and a linear model of the loudspeaker is used to determine the voice coil displacement corresponding to the measured voltage and/or current. This second technique does not require any special hardware, and has low computational overhead. However, as may be appreciated from the displacement function  200  represented in  FIG. 2 , above, this second technique produces inaccurate results that are at best an approximation of the actual voice coil displacement. Indeed, because damaging over-excursion occurs only outside the linear region  201  of loudspeaker operation, in the example of  FIG. 2  this second technique will be least accurate precisely in the operating region in which it is needed. In fact, this technique will overestimate the voice coil displacement, and therefore unnecessarily reduce the loudspeaker output. The inaccuracy is compounded because voice coil displacement is frequency-dependent, being greatest at the resonant frequency of the loudspeaker, but because the actual loudspeaker system is non-linear, the resonant frequency itself changes with amplitude. 
     According to a third technique, the driving voltage and/or current are measured, and the voice coil displacement corresponding to the measured voltage and/or current is determined by a linear model of the loudspeaker that is augmented by a lumped parameter model (also known as a lumped component model or lumped element model) for the non-linear parameters. This third technique does not require any special hardware, and produces a relatively accurate result, but is computationally intensive, and therefore power-intensive as well. 
     A lumped parameter model  300  of a loudspeaker, as described by W. Klippel, “Prediction of Speaker Performance at High Amplitudes”, Convention Paper 5418, 111th Convention of the Audio Engineering Society (2001), is shown in  FIG. 3 , where: 
     R e =electrical resistance R ms =mechanical resistance 
     L e =electrical inductance F m =reluctance force 
     BL=force factor Z m =lossy inductance 
     R 2 , L 2 =Eddy factor parameters M ms =mechanical mass 
     K ms =stiffness 
     The lumped parameters may be derived computationally from the lumped parameter model, giving rise to the computational burden referred to above for this third technique. 
     In accordance with implementations of the subject matter of this disclosure, the computational burden of determining the lumped parameters may be reduced by using an adaptive filter to match the loudspeaker current. The filter coefficients resulting from that adaptation can be used to directly predict voice coil displacement as discussed below. The filter may be an infinite impulse response (IIR) filter, which responds quickly to input changes. 
     A subset of the lumped parameters may be identified as functions of the voice coil displacement x:
         R e (x), K ms (x), M ms (x), R ms (x), BL (x)
 
which also may be referred to as R e , K t , m t , c t , and Φ, respectively. These t parameters may be calculated directly from the IIR filter coefficients adaptation without knowing the displacement x. Therefore, instead of deriving these parameters from x, x can be derived from these parameters.
       

     An adaptive IIR Filter is used to minimizing the error between the measured loudspeaker current and the estimated loudspeaker current. The IIR transfer function (in the z-transformed domain) is:
 
 Y ( z )=( b   0   +b   1   z   −1   +b   2   z   −2 )/(1+ a   1   z   −1   +a   2   z   −2 )
 
where the vector [b0 b1 b2 a1 a2] represents the adapted coefficients. During normal sound reproduction, the adapted coefficients “move” trough the solution space, in order to minimize the adaptive error, thereby adapting the internal filter to the external loudspeaker. After the adaptive filter coefficients have been adapted, they track loudspeaker variations in real time. Therefore, they can be used to predict the non-linear lumped parameters and the voice coil displacement.
 
     Starting from the IIR Adapted Admittance:
 
 Y ( z )=( b   0   +b   1   z   −1   +b   2   z   −2 )/(1+ a   1   z   −1   +a   2   z   −2 )  1)
 
This can be rewritten in a more canonical form:
 
 Y ′( z )=( b   0   z   2   +b   1   z+b   2 )/( z   2   +a   1   z+a   2 )  2)
 
     Applying the z2s Direct Transform (from the discrete-time z domain to the continuous-time Laplace s domain):
 
 z= 1+ sT   3)
 
gives the s-Admittance:
 
 Y ′( s )=((( b   0   +b   1   +b   2 )/ T   2 )+((2 b   0   +b   1 ) s/T )+ b   0   s   2 )/(((1+ a   1   +a   2 )/ T   2 )+((2+ a   1 ) s/T )+ s   2 )  4)
 
     Rewriting the s-Admittance using the lumped parameters and applying Eq. 3:
 
 Y ( s )=( k   t   /R   eb   m   t   +c   t   s/R   eb   m   t   +s   2   /R   eb )/( k   t   /m   t +( c   t   /m   t +Φ 2   /R   eb   m   t ) s+s   2 )  5)
 
from which the relationships between the lumped parameters and the z-Admittance coefficients may be determined:
 
 R   eb =1/ b   0   6)
 
 k   t   =m   t ((1+ a   1   +a   2 )/ T   2 )  7)
 
 c   t =( m   t   /b   0 )((2 b   0   +b   1 )/ T )  8)
 
Φ 2 =( m   t   /b   0 )(2+ a   1   /T )− c   t   /m   t )  9)
 
Eq.9 can be rewritten as:
 
Φ=(1/ b   0 )( m   t ( a   1   b   0   −b   1 )/ T ) 0.5   10)
 
The resonant frequency f 0  also may be calculated from the z-Admittance
 
 f   0 =(1/2π)( k   t   /m   t ) 0.5 =(1/2π)((1+ a   1   +a   2 )/ T ) 0.5   11)
 
     The voice coil displacement in the s-domain is:
 
 X ′( s )=(Φ/ R   eb   m   t )/( k   t   /m   t +( c   t   /m   t +Φ 2   /R   eb   m   t ) s+s   2 )  12)
 
Substituting Eqs. 6-9 yields voice coil displacement in the z-domain in terms of the filter coefficients [b0 b1 b2 a1 a2]:
 
 X ( z )=(1/ b   0 )( m   t ( a   1   b   0   −b   1 )/ T ) 0.5   z   −2 )/(1+ a   1   z   −1   +a   2   z   −2 )  13)
 
The estimated voice coil displacement can be obtained by applying the input voltage signal:
 
 X   e ( z )= X ( z ) V   in ( z )  14)
 
     Estimated loudspeaker temperature also can be derived from the filter coefficients. The estimated loudspeaker temperature can be predicted from electrical resistance lumped parameter R e (T 0 ) (T)=R e (T 0 ) (1+α(T−T 0 )), where α=3.86×10 −3 /° K for copper. This can be rewritten as T=T 0 +(1/α)(((R e (T))/(R e (T 0 ))−1), where the R e  terms may be derived from the filter coefficients as described above. 
     An electromechanical protection system  400  based on motion estimated in this way is shown in  FIG. 4 , and includes an excursion compressor control (ECC) block  401  and an adaptive system model  402 . The incoming voltage signal V is input to both ECC block  401  and adaptive speaker model  402 . An adjusted voltage V* is output by ECC block  401  and drives electromechanical system  403 , and also is fed back to adaptive system model  402 . Adaptive system model  402  outputs a current I′ based on input voltage V, and on error signal e, which results from subtracting I′ from the adjusted current I* and which is fed back to adaptive system model  402 . Adaptive system model  402  also outputs estimated motion x′ and estimated temperature T′, which are input to ECC block  401  to generate adjusted voltage V* and adjusted current I*. 
     As discussed above, system  403  may be loudspeaker. Thus, adaptive system model  402  may be an adaptive speaker model, in which estimated motion x′ is estimated voice coil displacement. As shown in  FIG. 5 , adaptive speaker model  500  may include lumped speaker model  300  and adaptive IIR filter  501 , which operates as described above in connection with Equations 1-14. Adaptive IIR filter  501  receives, as inputs, error signal e and adjusted voltage V*, and provides the lumped parameters R e , K t , M t , c t , and Φ, to lumped speaker model  300 . Lumped speaker model  300  also receives, as inputs, input voltage V and adjusted voltage V*. 
     In practice, as shown in  FIG. 6 , adaptive speaker model  500  is collapsed to a single block that provides estimated displacement x′ and estimated temperature T′ based on the IIR coefficients as described above, without explicitly deriving the parameters R e , K t , m t , c t , and Φ. 
       FIG. 6  also shows an implementation  600  of ECC block  401  in more detail. As shown, implementation  600  of ECC block  401  includes an adaptive notch filter  601 , a path equalizer  602 , an attenuation control mixer  603 , and a temperature control block  604 . 
     Attenuation control mixer  603  selects equalized voltage V ˜  from path equalizer  602 , or attenuated voltage V′ from adaptive notch filter  601 , or a mix of voltage V ˜  and voltage V′, depending on the value of estimated displacement x′ from adaptive speaker model  402 . As noted above, voice coil displacement is greatest at the resonant frequency of loudspeaker  101 , and the resonant frequency is amplitude dependent. If the estimated displacement is large enough to potentially damage loudspeaker  101 , attenuation control mixer  603 , under control of estimated displacement x′, will select more of voltage V′ from adaptive notch filter  601 . Adaptive notch filter  601  will have adapted to the instantaneous resonant frequency based on the adjusted voltage V* and the adjusted current I*, reducing the amplitude of the input voltage component at the resonant frequency in voltage V′. Therefore, selection of voltage V′ from adaptive notch filter  601  will reduce the voice coil displacement from damaging levels by removing from adjusted voltage V* the greatest contribution to those damaging levels. 
     On the other hand, if the estimated displacement is not large enough to potentially damage loudspeaker, attenuation control mixer  603 , under control of estimated displacement x′, will select more of voltage V ˜  from path equalizer  602 . Path equalizer  602  does not affect the magnitude of voltage V ˜ , but adjusts for any phase delay introduced by adaptive notch filter  601 , so that there is no phase mismatch between the portion of the signal that passes through adaptive notch filter  601 , and the portion of the signal that does not pass through adaptive notch filter  601 . 
     In one implementation, attenuation control mixer  603  may be a mixed-mode attenuation control (MMAC) mixer, using a combination of frequency attenuation control (FAC) and amplitude attenuation control (AAC) to select the relative amounts of voltage V ˜  and voltage V′ to pass. 
       FIG. 7  shows an example  700  of an FAC transfer function that may be used by MMAC mixer  603 . The relative amounts of voltage V ˜  and voltage V′ selected by MMAC mixer  603  are shown in  FIG. 7  as a function of normalized estimated displacement x′=x/x max . At a low level of displacement, the unfiltered, but phase-equalized, input signal (V ˜ ) passes without attenuation while the notch-filtered signal (V′) is blocked. At higher levels of displacement, more of the notch-filtered signal (V′) is passed, and the output V″ contains very little, or none, of the original signal from frequencies close to resonant frequency. FAC transfer function  700  may be described mathematically as the linear combination of a function  701  of input signal V ˜ , denominated φ 1 (x′), and a function  702  of notch-filtered signal V′, denominated φ 2 (x′):
 
FAC=φ 1 ( x ′) V   ˜ +φ 2 ( x ′) V′.  
 
This results in relatively smooth control of voice coil displacement.
 
     However, at a very high volume level, FAC transfer function  700  may not be sufficient to limit excessive voice coil displacement. Therefore, in addition to FAC transfer function  700 , MMAC mixer  603  may also use an AAC transfer function such as the AAC transfer function  800  shown in  FIG. 8  as φ 3 (x′). At a low level of displacement, AAC transfer function  800  does not attenuate the input signal. But as displacement increases, the input signal is attenuated by an increasing amount, limiting voltage V″ and therefore the voice coil displacement. The overall transfer function of MMAC  603  in this example is therefore:
 
 V″=φ   3 ( x ′)[(φ 1 ( x ′) V   ˜ +φ 2 ( x ′) V′ ].
 
     Temperature control block  604  receives the estimated or predicted temperature T′ from adaptive speaker model  402  and the voltage V″, and adjusts the voltage V″ to yield voltage V*. One example of a temperature control transfer function  614  is shown in block  604 , where the output voltage V* is clamped at a certain maximum voltage, which transfer function  614  reaches at a certain maximum temperature T max , which may be predetermined according to the temperature at which the speaker (or other system) may be damaged. 
     Thus it is seen that without complex hardware for measuring actual voice coil displacement, and without a high computational burden, a loudspeaker protection method and system according to implementations of the subject matter of this disclosure will pass through loudspeaker input signals causing low levels of voice coil displacement, but for loudspeaker input signals causing higher levels of voice coil displacement, the signal will be attenuated by increasing amounts as voice coil displacement approaches a loudspeaker damage threshold, based on transfer functions such as those illustrated in  FIGS. 7 and 8 . Similarly, loudspeaker input signals that may cause a small temperature increase would be allowed to pass, while loudspeaker input signals that may cause potentially damaging temperature increase would be clamped at a non-damaging level according to a transfer function such that illustrated in  FIG. 6 . 
     The systems and methods described above can be used in any fixed or portable system that includes a loudspeaker for reproducing audio signals, such as a mobile telephone or analog or digital music player. Such systems also can be used to control any electromechanical system in which excessive motion or temperature is an issue, such as an electric motor. 
     It will be understood that the foregoing is only illustrative of the principles of the invention, and that the invention can be practiced by other than the described embodiments, which are presented for purposes of illustration and not of limitation, and the present invention is limited only by the claims which follow.