JP3055562B2 - How to change filter coefficient of digital filter - Google Patents

Description

Description: TECHNICAL FIELD The present invention relates to a method for changing a filter coefficient of an IIR filter (a filter having a feedback loop) by digital signal processing.

Thanks to the emergence of digital signal processors (hereinafter referred to as DSPs) that can execute multiply-accumulate operations at high speed,
Digital signal processing by a digital filter or the like realized by software processing has become a reality, and has become widespread in the fields of communication equipment, audio equipment, and the like.

The present invention relates to a method of changing a filter coefficient in order to change a characteristic of an IIR filter having a feedback loop among the digital filters.

[Problems to be solved by the prior art and the invention]

In FIG. 4 (a), a secondary filter 10 is shown as an example of an IIR filter. The transfer function of this filter is Given by Then, the filter difference equation _{y [n] = a 0 x} [n] + a 1 x [n-1] + a 2 x [n-
_{2] + b 1 y [n} -1] + b 2 y [n-2] is realized by performing the operation represented by (2). Here, x [i] (i = 1... N) is the value of the input signal for each sampling period (operation period), and y [i] (i = 1.
n) is the value of the output signal. The coefficients a ₀ , a ₁ , a ₂ , b ₁ , and b ₂ are constants that determine the characteristics of the filter, and by sequentially changing these, the characteristics of the filter are changed.

Incidentally, these constant changes are generally transferred from a control device external to the DSP, for example, a microcomputer. In that case, the transfer speed is generally slower than the operation cycle of the filter operation, so that the coefficient before the change and the coefficient after the change are mixed during the filter coefficient change period. In that case, depending on the combination of each filter coefficient and signal input at that time and the value existing on each tap t ₁ , t ₂ , t ₄ , t ₅ , an unexpected abnormal value appears at the output, which is a feedback loop. Hopping may cause pop noise. If the value is the maximum positive value or the maximum negative value that can be expressed digitally, no matter how small the number is multiplied as a coefficient, the sign will only change, so from the positive maximum value to the negative maximum value, and so on A signal that changes abruptly in the opposite direction circulates and is output, resulting in a so-called oscillation state. This phenomenon is a troublesome phenomenon that the state is maintained even after all the filter coefficient changes are completed.

If all filter coefficients can be changed within one operation cycle (sampling cycle), pop noise and oscillation can be prevented.
The software or hardware of P must be added, resulting in a complicated configuration.

Mute circuit 20 shown in column (b) or (c) of FIG.
Is provided before the filter 10 (column (b)) or after the filter 10 (column (c)), and during the period when the filter coefficient is being changed, the signal is reduced to 0 by the mute circuit 20 so that the transition period is reduced. Can be considered a method of suppressing the generation of noise. However, in the method shown in column (c), the filter
Since the signal input to 10 and the state of the filter 10 are the same as those without the mute circuit 20, the problem of oscillation is not solved. According to the method shown in the column (b), the taps t ₁ and t _2, that is, the value of the feedforward section can be set to 0 after the elapse of a predetermined sampling period, so that the output can be suppressed to a very small value, and noise during the transition period is generated. It is considered that this has an effect of suppressing oscillation. However, in the feedback section of the taps t ₄ and t ₅ , past values circulate, and the possibility of oscillation cannot be completely denied.

Accordingly, an object of the present invention is to propose a method of changing a filter coefficient which completely prevents oscillation of a filter by a simple method.

[Means for solving the problem]

FIG. 1 is a view for explaining a filter coefficient changing method of a digital filter according to the present invention. In this figure,
The method of the present invention comprises the following steps: at least a primary feedback signal operation unit 40, at least a primary feedforward signal operation unit 44, an output of the feedback signal operation unit 40, an output of the feedforward signal operation unit 44, and an input signal. A filter coefficient changing method for a digital filter including an adder 42 for adding, wherein (i) all filter coefficients are sequentially set to 0, and (ii) the filter coefficients in the feedback signal operation unit 40 are sequentially changed. (Iii) updating the filter coefficients in the feedforward signal calculation unit 44 sequentially to the changed coefficients.

(Operation)

By sequentially setting all the filter coefficients to 0, all the values at the taps in the feedforward signal operation unit 44 and the feedback signal operation unit 40 become 0 after a lapse of a predetermined time after the change. After that, even if the filter coefficient of the feedback signal unit 40 is sequentially updated to the value after the change, each coefficient of the feedforward signal calculation unit 44 remains 0, so the value at each tap remains 0,
There is no danger of oscillation. After that, the coefficients of the feedforward signal calculation unit 44 are sequentially updated to the changed values, and the change of the filter coefficients is completed.

〔Example〕

FIG. 2 is a diagram showing an example in which the present invention is applied to an FM / AM radio having a sound quality adjusting function by digital signal processing.

In the figure, an audio signal obtained by tuning, detecting, and amplifying a radio wave received by an antenna in an FM / AM tuner 700 is converted into a digital signal in an analog-to-digital (A / D) converter 702, and Various digital signal processing, digital-analog (D / A)
The signal is converted back to an analog signal in the converter 704, amplified by the amplifier 706, and sounds the speaker 708. The microcomputer 710 has a well-known configuration using a CPU, a ROM, and a RAM input / output interface connected to a bus. To send to. The DSP 610 incorporates various filters and attenuators realized by software processing, and performs processing to adjust the sound quality and volume of input / output digital signals based on various constants sent from the microcomputer 710. .

A secondary digital filter is shown in the box of the DSP 610 as one of them. This configuration is also well-known. A delay unit 606 that delays an input signal by one sampling period and outputs a primary delay signal, a delay unit 608 that further delays the output and outputs a secondary delay signal, A delay device 400 that outputs a delayed primary feedback signal;
A delay unit 402 for further delaying and outputting as a secondary feedback signal, a multiplier 600 for multiplying the input signal by a predetermined coefficient b ₀ and outputting the same, and a predetermined coefficient a for the primary delay signal
Multiplier 602 and outputting the multiplying _1, and outputs the multiplied by a predetermined coefficient b ₁ against multiplier 604, primary feedback signal output by multiplying a predetermined coefficient a ₂ to the secondary delayed signal multiplier 404, the multiplier 406 and outputs the multiplied by a predetermined coefficient b ₂ to the secondary feedback signal, and a multiplier 404,4
It comprises an adder 420 which adds all the outputs of 06, 600, 602 and 604 to produce an output signal. This digital filter has the same configuration as that shown in FIG. 4 (a).

Next, the coefficients of the digital filter having the configuration shown in FIG.
A microcomputer for changing a ₀ , a ₁ , a ₂ , b ₁ , b ₂
The flow of the process of 710 will be described with reference to the flowchart of FIG. First, in order to set all the filter coefficients to 0, a code representing each parameter and a value 0 are paired to the DSP 610.
Send to (Step a). Next, values are sent to update the coefficients b ₁ and b ₂ to new values (step b). Coefficient 1
The time required to send the pieces is more than several times the sampling period, and the coefficients a ₀ , a ₁ , a ₂ all remain 0,
By the time the processing of step b is completed, taps t ₀ , t ₁ , t ₂ , t ₄ ,
value in t ₅ are all set to 0. Then a ₀ , a ₁ , a ₂
Is sent to update the value to a new value, and the coefficient change process is completed (step c).

As described above, after the coefficients b ₁ and b ₂ in the feedback loop are updated, the values of the taps are all 0, so that the oscillation is completely prevented.

〔The invention's effect〕

As described above, according to the present invention, it is possible to completely prevent the oscillation of the filter with a simple configuration in which only the processing on the coefficient sending side is added without changing the processing on the DSP side.

[Brief description of the drawings]

FIG. 1 is a diagram for explaining a method of the present invention, FIG. 2 is a diagram showing an embodiment to which the present invention is applied, FIG. 3 is a flowchart of a filter coefficient changing process according to the present invention, FIG. FIG. 4 is a diagram for explaining a conventional method. In the figure, 404,406,600,602,604... Multipliers, 400,402,606,608... Delayers, 42,420.

────────────────────────────────────────────────── ─── Continuation of front page (72) Inventor Masatsugu Uemura Fujitsu Ten Co., Ltd. 1-2-28, Goshodori, Hyogo-ku, Kobe-shi, Hyogo Prefecture (56) References JP-A-61-7716 (JP, A) JP-A-60-198911 (JP, A) JP-A-57-44322 (JP, A) (58) Fields investigated (Int. Cl. ⁷ , DB name) H03H 17/00-17/08

Claims (1)

(57) [Claims] An at least primary feedback signal operation unit (40) for performing an operation on an output signal of a digital filter and outputting the result, and an at least primary feedforward unit for performing an operation on an input signal of the digital filter and outputting the result. A signal operation unit (44), an output of the feedback signal operation unit (40), and the feedforward signal operation unit (44)
And an adder (42) for adding the output of the digital filter to the output signal of the digital filter, wherein all the filter coefficients of the digital filter are sequentially set to 0, Digitally updating the filter coefficients in the feedback signal operation section (40) sequentially to the changed coefficients, and thereafter changing the filter coefficients in the feedforward signal operation section (44) to the sequentially changed coefficients. How to change the filter coefficient of the filter.