JPH06266373A - Noise cancellation system - Google Patents

Abstract

PURPOSE:To prevent a coefficient from vibrating owing to disturbance by calculating the mean values of the latest coefficients obtained by a coefficient update expression, tap by tap, and employing the mean value as a current real coefficient of each tap. CONSTITUTION:For adaptive signal processing, a filter 32c for filtered X signal generation performs arithmetic in each sampling cycle to generate a filtered X signal rn. The signal fetch part 41 of an adaptive signal processing part 32a inputs the filtered X signal rn and an error signal en and a coefficient arithmetic part 42 calculates coefficients Wj(n+1) (j=1, 2, 3, and n) of the respective taps on the basis of the coefficient update expression. The coefficient storage part of a vibration removal part 43 stores the latest coefficients obtained by the coefficient arithmetic part 42 as to the respective taps. A mean value arithmetic part calculates the mean value of the latest coefficients, tap by tap, and inputs it to an adaptive filter 32b. The adaptive filter 32b performs digital filter processing by using the coefficients from which vibration components are removed and outputs a noise canceling signal yn.

Description

Detailed Description of the Invention

[0001]

BACKGROUND OF THE INVENTION 1. Field of the Invention The present invention relates to a noise canceling system, and more particularly to a noise canceling system for improving a noise canceling effect by removing a vibration component contained in a coefficient of an adaptive filter.

[0002]

2. Description of the Related Art As a noise countermeasure, a method using a sound absorbing material (passive control) has been conventionally known. However, in the method using the sound absorbing material, it is troublesome to form a quiet area where noise is small, and there is a problem that the bass cannot be effectively eliminated. In particular, in order to prevent noise in the passenger compartment of an automobile, there is a problem that the weight of the automobile increases and the noise cannot be effectively eliminated. For this reason, a method (active control) of radiating a noise canceling sound having a phase opposite to that of the noise from the speaker to reduce the noise has been in the spotlight and is being put to practical use in a part of indoor space such as a factory or an office. Also, a method has been proposed in which noise is reduced by active control even in the passenger compartment of an automobile.

FIG. 3 is a block diagram of a conventional noise canceling device for canceling engine noise. Reference numeral 11 is an engine that is a noise source, 12 is a rotation speed sensor that detects the engine rotation speed R, and 13 is a reference signal generation unit that generates a sine wave signal of a constant amplitude having a frequency according to the engine rotation speed R as a reference signal xn. Is. When the noise source is an engine, noise generated by engine rotation has a periodicity, and its frequency depends on the engine speed. For example, in the case of a four-cylinder engine, the periodic noise generated in the vehicle compartment is dominated by the second harmonic of the engine speed, and the engine speed is 600r.
At pm (10 rps), the frequency of noise generated in the passenger compartment is 20 Hz, and the rotation speed is 6000 rpm (100 rp).
In the case of s), the frequency of noise generated in the vehicle compartment is 200H
z. The reference signal generation unit 13 stores the sine wave data of the second harmonic in the ROM, reads the data as needed, and outputs the data to generate the reference signal xn. The timing of reading / outputting this data is controlled according to the engine speed R, whereby a reference signal having the frequency of periodic noise generated according to the engine speed R is output.

Reference numeral 14 denotes a noise canceling controller, which receives the reference signal xn generated from the reference signal generating section 13 and outputs noise Sn at a noise canceling position in the passenger compartment (observation point, for example, near the driver's ear). The synthesized sound signal of the cancel sound Sc is input as an error signal en, and adaptive signal processing is performed so that the error signal is minimized, and a noise cancel signal yn is output. The noise cancellation controller 14 filters the adaptive signal processing unit 14a, the adaptive filter 14b having a digital filter configuration, and the reference signal xn by convolving the propagation characteristics (transfer function) of the cancellation sound propagation system from the speaker to the noise cancellation point. It has a filtered X signal producing filter 14c for producing an X signal (reference signal for signal processing) rn. 1
5 is a DA converter that converts the output of the adaptive filter (noise cancel signal yn) into an analog noise cancel signal, 16 is a power amplifier that amplifies the noise cancel signal, 17 is a cancel speaker that emits the noise cancel sound Sc, 18 Is placed at the noise cancellation point, and noise S
An error microphone that detects the synthesized sound of n and the cancel sound Sc and outputs the synthesized sound signal as an error signal en, 19 is an amplifier that amplifies the error signal en, and 20 is a noise signal outside the band of periodic noise. Reference numeral 21 denotes an AD converter that converts the output of the low-pass filter into a digital signal.

The adaptive signal processing unit 14a receives the error signal en at the noise canceling point and the signal processing reference signal (filtered X signal) rn input through the filter 14c, and uses these signals to cancel the noise canceling point. Adaptive signal processing is performed so as to cancel noise, and the coefficient of the adaptive filter 14b is determined. For example, the adaptive signal processing unit 14a is a well-known filtered X LMS (Least Mean).
Square) According to the adaptive algorithm, the error microphone 18
The coefficient of the adaptive filter 14b is determined so that the error signal en input from (1) is minimized. Adaptive filter 14b
Cancels noise by performing digital filter processing on the reference signal xn according to the coefficient determined by the adaptive signal processing unit 14a and outputting a noise cancel signal yn. still,
The reference signal xn must be a signal having a high correlation with the noise Sn that is desired to be deleted, and a sound that is not correlated with the reference signal is not deleted.

The adaptive filter 14b, as shown in FIG.
It is composed of an N-tap FIR digital filter, and, for example, sequentially delays the input signal by one sampling time (N-
1) N delay elements DL, DL ... And N pieces for multiplying each delay element output by coefficients w ₁ (n), w ₂ (n), w ₃ (n) ... w _N (n) , And the addition units AD, AD, ... Which sequentially add the outputs of the respective multiplication units. That is, the reference signal at the current time n · Ts is xn, and the coefficient of each multiplier at that time is w ₁ (n), w ₂ (n), w ₃ (n), ... W.
_{If N} (n) and the output (noise cancellation signal) are yn, the adaptive filter 14b is

[0007]

[Equation 1] Then, the noise cancellation signal yn is output.

Filtered X signal producing filter 14c
As shown in FIG. 5, it is configured by an FIR type digital filter, for example, (M-1) delay elements DL, DL ... Which delay the input signal sequentially by one sampling time, and a coefficient for each delay element output. It is realized by M multiplication units ML, ML, ... For multiplying c ₁ , c ₂ , c _3, ... C _M , and addition units AD, AD, ... For sequentially adding the outputs of the multiplication units. . Coefficient c ₁ ,
c ₂ , c ₃ ... c _M are determined so as to simulate the propagation characteristics of the secondary sound propagation system (system from the speaker to the observation point).
If the reference signal at time n · Ts is xn and the output (filtered X signal) is r (n), the filter 14c is

[0009]

[Equation 2] And outputs the filtered X signal r (n).

The adaptive signal processing unit 14a has the coefficients w ₁ (n + 1), w ₂ (n + 1) and w ₃ of the adaptive filter 14b at the next time (n + 1) · Ts after one sampling time Ts. (n + 1) ・ ・ ・ w
_N (n + 1) is the coefficient at the current time n · T and the error signal en
And the filtered X signal rn are used to determine by the following equation (coefficient updating equation).

[0011]

[Equation 3]

However, the j-th filter coefficient updating formula is given by w _j (n + 1) = w _j (n) + μ · r (n-j + 1) · en (4). In equation (3), (n) is the value at the current sampling time, (n + 1) is the value one sampling time later, and (n-1) is 1
The value before the sampling time, (n-2) means the value before the two sampling times, .... Further, μ is a constant (step size parameter) of 1 or less that determines the step of updating the coefficient of the adaptive filter, and is set to an appropriate value according to the noise cancellation system. The above is the case where the adaptive signal processing is performed by the filtered X LMS adaptive algorithm, but in the case of the LMS adaptive algorithm that does not use the filtered X signal, the equation (3) is

[0013]

[Equation 4]

Thus, the j-th filter coefficient updating formula is given by w _j (n + 1) = w _j (n) + μ · x (n-j + 1) · en (4a). In the noise cancellation by the filtered XLMS adaptive algorithm, the operations of the above formulas (1) to (3) are performed within one sampling time, and the noise cancellation signal yn is output. Further, when the LMS adaptive algorithm is used, the calculation of the equations (1) and (3a) is performed within one sampling time, and the noise cancel signal yn is output.

[0015]

In addition to the noise signal to be canceled, there is a disturbance signal at the noise cancellation point. The disturbance signal has frequency components other than the frequency component to be canceled (= frequency component of the reference signal xn) and is mixed in the error signal en. When the disturbance signal is included in the error signal en, a signal having a frequency component completely different from that of the noise signal is generated by the product-sum calculation in the adaptive algorithm, and a signal other than the frequency component to be canceled is output from the adaptive filter. . For example, the frequency of the noise signal to be canceled is f ₁ (ω ₁ = 2πf ₁ ), and the frequency of the disturbance signal is f ₂ (ω ₂ = 2
πf ₂ ), and in order to simplify the explanation, assuming that the amplitude is 1 and the phase is 0, the signal processing reference signal rn and the error signal en are rn = sinω ₁ t en = sinω ₁ t + sinω ₂ t .

Therefore, the second term on the right side of the coefficient updating equation is sinω ₁ t (sinω ₁ t + sinω ₂ t) = (sinω ₁ t) ² + (1/2)  {cos (ω ₁ -ω ₂ ) t- cos (ω ₁ + ω ₂ ) t} (5), which includes the components of frequencies f ₁ , (f ₁ −f ₂ ), and (f ₁ + f ₂ ). When the coefficient of the adaptive filter 14b converges to a substantially constant value, the sinω ₁ t component included in the error signal en becomes small, so that the frequency components in the respective coefficients of the adaptive filter are (f ₁ −f ₂ ), (f ₁ + f ₂ ) becomes dominant, and the coefficient w _j of the j-th tap oscillates with time as shown in FIG. That is, the coefficient of the adaptive filter has a component that slowly oscillates at the frequency (f ₁ −f ₂ ), and a frequency (f 1
_It has a component that vibrates finely at ₁ + f ₂ ). Since the adaptive filter 14b multiplies the reference signal xn (= sin ω ₁ t) by the equation (5), more noise components are added to the noise canceling signal yn, and the noise canceling effect is reduced. That is, if the coefficient of the adaptive filter fluctuates due to the disturbance, noise cannot be canceled properly, and the silencing effect becomes small. From the above, an object of the present invention is to provide a noise canceling method capable of preventing the coefficient of the adaptive filter from vibrating due to disturbance.

[0017]

According to the present invention, the above object is to calculate an average value of a plurality of latest coefficients obtained by a coefficient update formula for each tap, and to calculate the average value for each tap. This is achieved by taking the true coefficient of the tap.

[0018]

For each tap, the average value of the latest plurality of coefficients obtained by the coefficient updating formula is calculated, and the average value is set as the true coefficient of each tap this time. This makes it possible to prevent the coefficient of the adaptive filter from vibrating due to disturbance.

[0019]

DETAILED DESCRIPTION OF THE PREFERRED EMBODIMENTS FIG. 1 is a block diagram of an example of the whole of the present invention.
Is the reference signal xn according to the second harmonic of the engine speed R
Is a reference signal generator, 32 is a noise cancel controller that outputs a noise cancel signal yn by adaptive signal processing (for example, a filtered X LMS algorithm), and 33 is a synthesized sound signal of the noise Sn and the cancel sound Sc at the noise cancel position. Is an error microphone for outputting as an error signal en. The noise canceling controller 32 includes an adaptive signal processing unit 32a, an adaptive filter 32b having an N-tap digital filter configuration, and a reference signal xn by convolving the propagation characteristics of the canceling sound propagation system from the speaker to the noise canceling point to the filtered X. It has a filtered X signal creation filter 32c for creating a signal (reference signal for signal processing) rn. Adaptive filter 32b,
Each of the filtered X signal generating filters 32c has the structure shown in FIGS.

The adaptive signal processing section 32a has a filtered X
A signal fetching unit 41 for fetching the signal rn and the error signal en at a predetermined sampling period Ts, a coefficient computing unit 42 for performing the computation of the coefficient updating equation of the equation (3), and a vibration removing unit for eliminating the vibration component from the coefficient. Has 43. As shown in FIG. 2, the vibration removal unit 43 has a coefficient storage unit 43a and an average value calculation unit 43b. The coefficient storage unit 43a, for each tap of the adaptive filter 32b, obtains the latest (M + 1) coefficients w _j (n + 1), w _j (n), and w _j (n-) obtained by the coefficient calculation unit 42. 1), ... w _j (n-M + 1) (j = 1,
Memorize 2,3, ... N). The coefficient storage unit 43a has a ring buffer configuration and sequentially stores the latest (M + 1) coefficients. Note that w _j (n + 1) is the coefficient calculated this time. The average value calculator 43b uses the latest (M + 1) of each tap.
The average value wjm of the coefficients is

[Equation 5] Calculated by, wjm → w adaptation as _j (n + 1) filter 32
Enter in b. In this way, if the average value of the latest (M + 1) coefficient values is calculated to be the current coefficient value w _j (n + 1), the positive and negative vibration components cancel each other out, and the influence of the vibration components is reduced. It can be lost.

In the adaptive signal processing, the filtered X signal generating filter 32c is (2) every sampling period Ts.
A filtered X signal rn is created by performing an equation calculation. The signal acquisition unit 41 of the adaptive signal processing unit 32a acquires the filtered X signal rn and the error signal en, and the coefficient calculation unit 42 calculates the coefficient w _j (n + 1) of each tap based on the coefficient update expression of the expression (3). ) (j = 1,2,3, ..., N) is calculated. For the coefficient storage unit 43a is each tap of the vibration removal unit 43, the latest obtained by the coefficient calculation unit 42 (M + 1) pieces of coefficients _{w j (n + 1),} w j (n), w j (n -1), ... w _j (n-M + 1) (j = 1,
Memorize 2,3, ... N).

The average value calculation unit 43b provides the latest value for each tap.
The average value wjm (j = 1, 2, 3, ..., N) of the (M + 1) coefficients is calculated by the equation (6), and the adaptive filter 32b is calculated as wjm → w _j (n + 1).
To enter. The adaptive filter 32b digitally filters the reference signal xn using the coefficient w _j (n + 1) (j = 1,2,3, ..., N) from which the vibration component has been removed, thereby canceling noise. The signal yn is output to cancel the noise. After that, the above operation is repeated for each sampling period. As described above, since the adaptive filter processing is performed based on the coefficient value w _j (n + 1) in which the vibration component is removed for each sampling cycle, noise can be canceled correctly and the silencing effect can be improved. Note that the above is the case of updating the coefficient according to the filtered X LMS algorithm, but it is also applicable to the case of updating the coefficient according to the LMS algorithm or the like that does not use the filtered X signal. Although the present invention has been described above with reference to the embodiments, the present invention can be variously modified according to the gist of the present invention described in the claims, and the present invention does not exclude these.

[0023]

As described above, according to the present invention, the average value of the latest plurality of coefficients of the predetermined tap obtained from the coefficient updating formula is calculated, and the average value is set as the true coefficient of the predetermined tap this time. Since it is configured to, the vibration component can be canceled by plus and minus of the vibration component, and the influence of the vibration component can be eliminated from the coefficient.
The muffling effect can be improved.

[Brief description of drawings]

FIG. 1 is an overall configuration diagram of the present invention.

FIG. 2 is a configuration diagram of a vibration removing unit of the present invention.

FIG. 3 is a configuration diagram of a conventional noise canceling device.

FIG. 4 is a configuration diagram of an adaptive filter.

FIG. 5 is a configuration diagram of a filter for generating a filtered X signal.

FIG. 6 is an explanatory diagram of vibration of coefficients.

[Explanation of symbols]

 31 .. Reference signal generation unit 32 .. Noise cancellation controller 32a .. Adaptive signal processing unit 32b .. Adaptive filter 41 .. Signal acquisition unit 42 .. Coefficient calculation unit 43 .. Vibration elimination unit

─────────────────────────────────────────────────── ─── Continuation of the front page (51) Int.Cl. ⁵ Identification code Office reference number FI technical display location H03H 21/00 7037-5J

Claims (1)

[Claims] 1. A speaker for outputting a cancel sound for canceling noise at a noise cancel point, a sensor for detecting a composite sound of the noise and the cancel sound at the noise cancel point as an error signal, and the error signal and the cancel signal. The reference signal corresponding to the noise is input, the coefficients of all taps of the adaptive filter are updated so as to cancel the noise at the noise cancellation point according to a predetermined coefficient update formula using these signals and the step size parameter, and the reference signal is set. In a noise canceling system of a noise canceling device equipped with a noise canceling controller for generating a noise canceling signal by inputting to the adaptive filter and inputting the noise canceling signal to a speaker, the coefficient updating formula is used for each tap of the adaptive filter. The average of the latest multiple coefficients obtained It calculates the value, the noise cancellation system, characterized by the average value and the true coefficients of the current each tap of.