JPS6059811A - Digital filter with resonance characteristic - Google Patents

Abstract

PURPOSE:To obtain a filtering output with stable gain by providing a control means decreasing an input signal to a digital filter in response to a resonance coefficient and applying the result to the filter. CONSTITUTION:An ROM storing coefficients b1, b2 selected by the resonance coefficient R and a cut-off frequency fc is provided to a filter coefficient operating circuit (a). The coefficient information b1, b2 outputted from said circuit (a) is given to a control circuit 10. The control circuit 10 gives an input coefficient K to a multiplier 1. In such a digital filter, as the resonance coefficient R is increased, the input coefficient K given from the control circuit 10 is reduced linearly, resulting that the peak value of the frequency characteristic is made almost constant and the overflow in the filter internal operation is reduced.

Description

DETAILED DESCRIPTION OF THE INVENTION [Technical Field of the Invention j] The present invention relates to a digital filter with resonance characteristics that realizes custom-made resonance (custom-made resonance).

[Backbone of invention]

In recent years, digital filters consisting of multipliers, adders, delay circuits, etc. have been studied in various fields and have been put into practical use.

The present applicant has already filed several applications for such digital filters whose amplitude characteristics have a peak near the cutoff frequency (for example, Japanese Patent Application No. 55-94462, Special application 1982-
27912).

In this prior art, when the transfer function of a digital filter is taken as the transfer function, only the coefficient b2 or both coefficients #b1 and h2 are changed according to the resonance coefficient R, and as a result, the resonance coefficient When is large,
There was a problem in which overflow often occurred in the calculations inside the digital filter, making it impossible to obtain the correct characteristics. To improve this, the overall gain (determined by the coefficient K) was lowered and the resonance was maximized. However, in that case, the gain when the resonance is at its minimum would be too small, resulting in an unnatural sound, making this type of digital filter particularly difficult to use in the field of electronic musical instruments. In this case, there were problems such as the output sound being too low and becoming audibly unnatural.

[Purpose of the invention]

SUMMARY OF THE INVENTION An object of the present invention is to provide a digital filter with a resonance characteristic that can provide a filtering output with a stable gain regardless of the value of the resonance coefficient.

[Key points of the invention]

[Embodiment] An embodiment of the present invention will be described below.

First, a general explanation of digital filters with resonance characteristics will be given. That is, it is a secondary reference roper. Here, Q indicates the depth of resonance, and is 1 in a normal state without resonance.

Transfer function color of this reference low-pass filter.

Let us consider the transfer function of the resonant angular frequency WO colossal filter.

becomes.

If we apply bilinear Z transformation to this, we get becomes. Therefore, T8 is the sampling time.

Therefore, given the resonance depth Q and the resonance angular frequency wo, the coefficients h1, b2, K
Therefore, by storing all the respective values in, for example, 1'1OM, a digital filter with resonance characteristics can be realized.

However, setting all the coefficient information in the FIOM in this way requires a large amount of ROM capacity, which is not realistic.

Therefore, the above-mentioned patent application No. 55-94462 and patent application No. 56
-27912, the coefficient h2 alone or the coefficients kl and bz are calculated according to the resonance coefficient R (corresponding to the depth Q of the resonance, and takes 0 when the resonance is minimum and 1 when the resonance is maximum). It has gained. As a result, only the coefficient information corresponding to the resonance angular frequency wo is stored in the ROM, and there is no need to store data for each coefficient when changing the resonance depth.

By the way, in such a digital filter, if it is assumed that the gain is 1 (OdB) when the angular frequency w = Q, then the relationship expressed by equation (1) will be obtained, and therefore. By the way, if the input coefficient is set to 0 in this way, there is a possibility that an overflow will occur in the internal calculation of the digital filter when the resonance is at its maximum.

Therefore, in this embodiment, the resonance coefficient R is used, and the input coefficient K is assumed to be KO(1-Rα) . Here, α is a positive value determined from the frequency range of the filter and the number of internal calculation bits, and for example, 7/8 is appropriate in cases where extremely accurate characteristics are not required, such as in electronic musical instruments.

Therefore, in this embodiment, according to equation (10), the input coefficient K
The specific circuit configuration thereof will be explained below with reference to FIG. 1.

1 in the figure is a multiplier, which calculates the input coefficient K for the input signal.
The result of the calculation is sent to the adder 2. This adder 2 is given the output of the multiplier 3.4, subtracts the output of the multiplier 3 and the output of the multiplier 4 from the output of the multiplier 1, and sends the result of the operation to the adder. 5 and the sampling time to a delay circuit 6 having a delay time of the sampling time.

The output of the delay circuit 6 is given to the multiplier 3, multiplied by a coefficient b1, and sent to the adder 2, as well as to a delay circuit 7 having a delay time equal to the sampling time and a multiplier 8. .

In this multiplier 8, the input signal is multiplied by a constant 2, and then outputted to the adder 5. In addition, the output of the delay circuit 7 is not only given to the adder 5, but also given to the multiplier 4. The multiplier 4 multiplies the result by a coefficient b2 and outputs the result to the adder 2.

As described above, the adder 5 is supplied with the output of the adder 2, the output of the multiplier 8, and the output of the delay circuit 7, respectively, and after adding them together, outputs the output as a digital filter output.

It is clear that the transfer function of the digital filter composed of each of the circuits 1 to 8 is as shown in equation (1).

Then, the coefficients h1 and h for this digital filter are
2 is output from the filter coefficient calculation circuit 9. The filter coefficient calculation circuit 9 includes a coefficient hl selected by the resonance coefficient R and the cutoff frequency fC (this cutoff frequency fC corresponds to the resonance angular frequency W0);
It is equipped with a ROM that stores b2. Of course, as described above, it is also possible to store in the ROM only the coefficients that are selected and turned on according to l and the cutoff frequency fC, and calculate the coefficient when adding resonance by calculation using the resonance coefficient H.0 This filter Coefficient information h output from the coefficient calculation circuit 9
x is supplied to the multiplier 3, and coefficient information b2 is supplied to the multiplier 4.

Further, the coefficient information @b1, h2 of Sowara is given to the control circuit 10.

This control circuit 10 performs a calculation corresponding to the formula (Iυ), and sends an input coefficient K to the multiplier 1. That is, the control circuit 10 receives the output b of the filter coefficient calculation circuit 9.
1, h2 is provided, and the adder 11 is provided with the numerical value "1", so the calculation output is (
1+61+b2).

Then, the output is multiplied by 1/4 by a multiplier 12 (actually, it can be composed of a shift circuit) and becomes information KO. That is, this KO is a value given by equation (9). This value KO is then applied to the multiplier 13, multiplied by RcL, and sent to the adder 14. Moreover, the above value K O is
It is also given directly to the adder 14, so that the adder 14 multiplies F! -Apply 11 times.

Therefore, in the digital filter of this embodiment, as the resonance coefficient iR increases, the input coefficient K decreases linearly by 1 in the control circuit 10, and as a result, the gain and angular frequency are shown in FIG. As shown in the relationship, the frequency Ifj becomes p.

In the above example, the present invention was applied to a digital low-pass filter according to equation (1), but it can also be applied to digital filters with different characteristics, such as similar bypass filters, and even higher-order filters. The same applies to digital filters.

Further, the method of decreasing the input coefficient K as the resonance coefficient H increases is not limited to linear change.

〔Effect of the invention〕

As detailed above, according to the present invention, even if deep resonance is applied, a stable gain can be obtained, and overflow inside the digital filter is reduced, which is advantageous.
This has the advantage that a natural-looking resonance characteristic can be achieved.

[Brief explanation of the drawing]

The drawings show an embodiment of the present invention; FIG. 1 is a circuit I of a digital filter, and FIG. 2 is a diagram for explaining resonance characteristics. 1.3.4.8... Multiplier, 2.5... Adder,
6.7...Delay circuit, 10...Control circuit. Patent applicant Casio Computer Co., Ltd.

Claims (1)

[Scope of Claims] (Li) A digital filter that realizes resonance characteristics according to a given resonance coefficient, characterized in that a control means is provided to reduce and supply an input signal to the digital filter according to the resonance coefficient. (2) The transfer function of the digital filter is given by, and the control means is simplified according to the equation. 1
7sonance% note digital filter.