1. Field of The Invention
The present invention relates generally to an apparatus for actively reducing noise for interior of enclosed space. The present invention, particularly, relates to the apparatus for actively reducing noise sound for a vehicular compartment or for a cabin of a fuselage, and so on, the noise sound being generated and propagated from a noise source, e.g., a vehicular or aircraft power source and the apparatus using an adaptive signal processing filter.
2. Description of The Background Art
A previously proposed active noise reduction apparatus is exemplified by a British Patent Application Publication No. GB 2 149 614 A published on Jun. 12, 1985.
FIG. 1 shows a circuit block diagram of the previously proposed active noise reduction apparatus described above.
In FIG. 1, an enclosed space 101 is provided with a plurality of, i.e., three loud speakers 103a, 103b, and 103c and a plurality of, i.e., four microphones 105a, 105b, 105c, and 105d. Each loud speaker 103a, 103b, 103c, and 103d generates a controlling sound which interferes with the noise sounds and each microphone 105a, 105b, 105c, and 105d measures a residual signal at an observing point of location of the enclosed space 101.
These loud speakers 103a, 103b, and 103c and microphones 105a, 105b, 105c, and 105d are connected to a signal processing unit 107. The signal processing unit 107 receives basic frequencies of the respective noise sources measured by basic frequency measuring means and input signals derived from the respective microphones 105a, 105b, 105c, and 105d and output drive signals to the loud speakers 103a, 103b, and 103c so that a sound pressure level in the enclosed space 101 gives a minimum value.
Although, in the enclosed space 101, three loud speakers 103a, 103b, and 103c and four microphones 105a,, 105b, 105c, and 105d are installed, suppose now that one loud speaker 103a and one microphone 105a are individually installed therein for easiness in explanation.
Suppose, then, that a transfer function established between the single noise source and the single microphone 105a is denoted by H, a transfer function established between the loud speaker 103a and microphone 105a is denoted by C, and a sound source information generated by the single noise source is denoted by X.sub.p.
At this time, a noise signal E as the residual noise sound observed by the microphone 105a is expressed below: EQU E=X.sub.p .multidot.H+X.sub.p .multidot.G.multidot.C
In the above equation, G denotes a transfer function required to extinguish or cancel the noise sound. Theoretically, at a sound extinguishing (canceling) point (at a position at which the microphone is disposed), when the noise is completely canceled, E=0. At this time, G is derived from the above-equation. EQU G=-H/C
Filter coefficients in the signal processing unit 107 are adaptively updated on the basis of G derived so that the power of microphone detection signal becomes minimum. A technique of deriving the filter coefficients so that the power of microphone detection signal E becomes minimum includes an LMS (Least Mean Square) algorithm which is a kind of a steepest descent method.
As shown in FIG. 1, in a case where the plurality of microphones are disposed, the control for the output signals for the loud speakers is such that a total sum of the powers of signals detected by, e.g., respective microphones 105a , 105b, 105c, and 105d becomes the minimum.
A Multiple Error Filtered-X LMS algorithm (hereinafter, LMS is referred to as Multiple Error Filtered-X LMS algorithm) will specifically be explained below.
That is to say, suppose now that a noise signal is denoted by e.sub.l (n) detected by an l number microphone 105a (105b, 105c, . . . ), a noise signal is denoted by e.sub.pl (n) detected by the l number microphone 105a (105b, 105c, . . . ) when no control sound is present from any one of the loud speakers 103a, 103b, and 103c, a filter coefficient is denoted by C.sub.lmj when a j number term of j=0, 1, 2, . . . , J.sub.c -1) a transfer function (a finite form of an impulse response function) established between an m number loud speaker 103a (103b, . . . ) and an l number microphone (evaluating point), i.e., working position is represented by a digital filter, a reference signal, i.e., sound source information signal x.sub.p (n), and a coefficient of the i number (i=0, 1, 2, 3, . . . , I.sub.k -1) of an adaptive processing filter which drives the m number of loud speaker 103a (103b, 103c, . . . ), inputting the reference signal x.sub.p (n) is denoted by W.sub.mi.
At this time, the equation (1) of attached Table 1 of mathematical equations is established.
Next, suppose furthermore that a performance function (a variable to make the noise signal e.sub.l (n) minimum) Je is expressed as in the equation (2) of attached Table 1 of the mathematical equations, the performance function being based on the equation of (1).
In order to derive the filter coefficients W.sub.mi which makes the performance function Je minimum, the LMS algorithm is adapted. That is to say, the filter coefficient W.sub.mi is updated with a value of a partial differential of Je with respect to each filter coefficient W.sub.mi.
Then, from the equation (2), the partial differential is calculated as in the equation (3) of attached Table 1 of the mathematical equations.
On the basis of the equation (1), the equation (4) of Table 1 of the mathematical equations is established.
If a right side of the equation (4) is substituted by r.sub.lm (n-i), an updating equation of the filter coefficients can be derived according to the equation (5) of attached table 1 of the mathematical equations including a weight coefficient of .gamma..sub.l.
As appreciated from the equation of (5), a stability and divergence of the LMS algorithm are predominated in an equation (6) of attached Table 1 of the mathematical equations, a convergence coefficient .alpha., and the weight coefficient .gamma..sub.l.
Although the above-equation (6) is dependent on a system characteristic to be controlled and a setting method of the microphones in the system, such a transfer function (finite impulse response) C.sub.lm as established from one of the loud speakers to one of the microphones is treated as constant.
However, aging effects of each microphone 103a, 103b, - - - and each loud speaker 105a, 105b, - - -cause phase characteristics of the respective speakers and loud speakers to be varied so that the transfer function C.sub.lm is accordingly varied. Consequently, a convergence characteristic of the updating equation of (5) becomes extremely unstable. If surrounding conditions of the equation (5) becomes worsened, a rise in a sound pressure level at the evaluating point may occur and, so called, a divergence phenomenon may occur at the evaluating point.
In this case, it may be possible for the convergence coefficient .alpha. to become smaller so as to suppress the divergence. As the convergence coefficient .alpha. becomes significantly smaller, the number of times that calculations of the equation (5) is carried out until reaching the convergence becomes larger. Consequently, the convergence characteristic may become moderate or dull.
Therefore, an algorithm in which an alternative performance function Jm is used has been proposed in an English paper of IEEE TRANSACTIONS 0N ACOUSTICS SPEECH AND SIGNAL PROCESSING, VOL. ASSP-35, No. 10, October 1987.
That is to say, drive signals for the speakers are added to the old minimizing performance function and .beta. is multiplied by the speaker drive signals to establish the alternative performance function as in the equation (7) of attached Table 1 of the mathematical equations.
It is noted in all equations from (1) to (7 ) that x(n) denotes the reference signal at a sampling time of n, e.sub.pl (n) denotes a residual noise detection signal (primary sound) detected by the l number microphone when no control sound (secondary sound) is received from any one of the loud speakers, C.sub.lmj denotes a filter coefficient when a j number term of the transfer function between the l number microphone and m number loud speaker is represented by a digital filter, y.sub.m (n) denotes the output of the m number loud speaker, e.sub.l (n) denotes an error signal detected by the l number microphone, W.sub.mi denotes the i number adaptive filter coefficient for the m number loud speaker, L denotes a number of microphones, M denotes a number of speakers, .alpha. denotes a convergence factor (coefficient), and .beta. denotes an effort coefficient.
In the way described above, when the term of the speaker driver signal is added into the performance function Jm , the coefficient (effort coefficient .beta.) to determine a length of a vector which serves to try to keep the adaptive filter coefficient not go far away from an origin 0 can be given since the performance function makes the speaker drive signal smaller.
That is to say, as shown in FIGS. 2 and 3, a point determined by the adaptive filter coefficients W.sub.mi tries to return to the origin, with the vector which tries to return to the origin given to the vector based on the convergence coefficient .alpha.. Hence, when the divergence phenomenon occurs, the performance function can be approached to a minimum.
FIG. 3 shows a control algorithm in a case where the adaptive filter has two variable filter coefficients W.sub.0, W.sub.1.
In FIG. 3, J.sub.1 denotes a first term of .SIGMA.e.sup.2 in the performance function of J.sub.m, J.sub.2 denotes a second term of .beta.y.sup.2, W.sub.opt denotes optimum filter coefficients of W.sub.0 and W.sub.1 according to the performance function J, .DELTA.W.sub.y denotes a resultant vector of .beta.y.sup.2 and .DELTA.W.sub.e denotes a resultant vector of .beta.y.sup.2.
However, even in the case where, as described above, the noises are controlled by means of the algorithm having the term multiplied by the effort coefficient .beta. when the transfer function C.sub.lm is varied, the performance function cannot always be returned to the minimum position since the effort coefficient .beta. is fixed, as shown in FIGS. 2 and 3, and a slight deviation may occur. Thus, the insufficient noise control may result.