WO2005078925A1 - デジタルフィルタの設計方法および装置、デジタルフィルタ設計用プログラム、デジタルフィルタ - Google Patents
デジタルフィルタの設計方法および装置、デジタルフィルタ設計用プログラム、デジタルフィルタ Download PDFInfo
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- G—PHYSICS
- G06—COMPUTING; CALCULATING OR COUNTING
- G06F—ELECTRIC DIGITAL DATA PROCESSING
- G06F17/00—Digital computing or data processing equipment or methods, specially adapted for specific functions
- G06F17/10—Complex mathematical operations
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- H—ELECTRICITY
- H03—ELECTRONIC CIRCUITRY
- H03H—IMPEDANCE NETWORKS, e.g. RESONANT CIRCUITS; RESONATORS
- H03H17/00—Networks using digital techniques
- H03H17/02—Frequency selective networks
- H03H17/06—Non-recursive filters
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- G—PHYSICS
- G06—COMPUTING; CALCULATING OR COUNTING
- G06F—ELECTRIC DIGITAL DATA PROCESSING
- G06F30/00—Computer-aided design [CAD]
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- H—ELECTRICITY
- H03—ELECTRONIC CIRCUITRY
- H03H—IMPEDANCE NETWORKS, e.g. RESONANT CIRCUITS; RESONATORS
- H03H17/00—Networks using digital techniques
- H03H2017/0072—Theoretical filter design
Definitions
- the present invention relates to a digital filter design program, a digital filter design program, and a digital filter design program.
- the present invention provides a delay line with a tap comprising a plurality of delay units, and an output signal of each tap. It is related to a type of FIR filter that multiplies each and then adds the multiplication results and outputs the result, and a design method thereof.
- IIR Infinite Impulse Response
- FIR Finite Impulse Response
- the FIR filter has the following advantages. First, since the pole of the transfer function of the FIR filter is only at the origin of the z-plane, the circuit is always stable. Second, if the filter coefficient is symmetric, it is possible to achieve completely accurate linear phase characteristics. In this FIR filter, an impulse response represented by a finite time length is used as it is as a filter coefficient. Therefore, designing an FIR filter means determining the filter coefficient so as to obtain the desired frequency characteristics.
- a filter coefficient is calculated based on a target frequency characteristic, and windowing is performed on the filter coefficient to obtain a finite number of coefficient groups.
- the obtained coefficient group was converted to frequency characteristics by performing FFT (Fast Fourier Transform), and the design was performed by a method to confirm whether or not this satisfied the characteristics of the giant target.
- FFT Fast Fourier Transform
- a method of adjusting the filter band by inserting a value of 1 or more between each tap (between each filter coefficient) of the tapped delay line is known (for example, see Table 6-6). 0 3.450 Publication No.)
- a method of realizing a steep frequency characteristic by cascade-connecting FIR filters is also known (for example, see Japanese Patent Application Laid-Open No. 5-243908). It was only possible to narrow the pass band of the filter, but it was not possible to accurately realize frequency characteristics of any shape with a small number of filters. Disclosure of invention ⁇
- the present invention has been made to solve such a problem, and it is an object of the present invention to provide an FIR digital filter capable of realizing a desired frequency characteristic with high accuracy on a small circuit scale and a design method thereof. Aim.
- one or more basic filters of the FIR type having a symmetrical numerical sequence having predetermined characteristics as filter coefficients are cascaded and connected in any combination. Processing for calculating the filter coefficient in the case, and rounding the lower few bits to the data of the calculated filter coefficient.
- the number of filter coefficient pits is reduced by performing a process of multiplying the output filter coefficient by a predetermined number and rounding off the decimal point. Coefficients are quantified.
- the unnecessary fill coefficient can be significantly reduced by rounding off the lower few bits of the fill coefficient.
- the number of taps required in the evening is very small, and the type of fill coefficient required for each evening output is very small. Therefore, the number of circuit elements (particularly, arithmetic units) can be greatly reduced, and the circuit scale can be reduced.
- the number of fill coefficients can be significantly reduced by the rounding process, it is possible to eliminate the need for windowing as in the past in order to reduce the number of fill coefficients.
- a filter coefficient having a value smaller than a predetermined threshold value is discarded by the rounding process for reducing the number of bits, most of the main filter coefficients that determine the frequency characteristics remain, which has a bad influence on the frequency characteristics. Almost no.
- the digital filter can be designed without windowing, there is no occurrence of a truncation error in the frequency characteristic, the cutoff characteristic can be greatly improved, and the phase characteristic has a linear and excellent filter characteristic. Can be. That is, a desired frequency characteristic of the digital filter can be realized at one time.
- the numerical value of the filter coefficient is converted to an integer.
- Can 'simplify' This makes it possible to configure a coefficient unit with a bit shift circuit instead of a multiplier, and to further simplify the configuration of a mounted digital filter.
- FIG. 1 is a diagram showing filter coefficients of a basic low-pass filter L 4 an.
- FIG. 2 is a diagram illustrating the frequency characteristics of the basic one-pass filter L4a4.
- FIG. 3 is a diagram illustrating the frequency-gain characteristics of the basic low-pass filter L4an.
- FIG. 4 is a diagram showing filter coefficients of the basic one-pass filter L an.
- FIG. 5 is a diagram showing frequency characteristics of the basic low-pass filter L a4.
- FIG. 6 is a diagram showing the frequency-gain characteristics of the basic mouth one-pass filter L an.
- FIG. 7 is a diagram showing filter coefficients of the basic hyper filter H 4 sn.
- FIG. 8 is a diagram illustrating frequency characteristics of the basic high-pass filter H 4 s 4.
- FIG. 9 is a diagram illustrating frequency-gain characteristics of the basic high-pass filter H 4 sn.
- FIG. 10 is a diagram showing filter coefficients of the basic high-pass filter H sn.
- FIG. 11 is a diagram illustrating a frequency characteristic of the basic high-pass filter Hs4.
- FIG. 12 is a diagram showing a frequency-gain characteristic of the basic high-pass filter H sn.
- Fig. 13 is a diagram showing the filter coefficients of the basic bandpass filter B4sn.
- FIG. 14 is a diagram showing the frequency characteristics of the basic band-pass filter B 4 s 4.
- FIG. 15 is a diagram showing the frequency-gain characteristics of the basic node-pass filter B 4 sn
- Figure 16 shows the filter coefficients of the basic bandpass filter Bsn.
- FIG. 17 is a diagram illustrating the frequency characteristics of the basic node-pass filter Bs′4.
- Fig. 18 shows the frequency-gain characteristics of the basic bandpass filter Bsn.
- FIG. 19 is a diagram illustrating a frequency-gain characteristic of the basic high-pass filter H msn in which m is a noise.
- FIG. 20 is a diagram showing an optimum value of the parameter n with respect to the parameter m.
- FIG. 21 is a diagram showing the relationship between the parameter m and the optimum value of the parameter n for the parameter m, and the relationship between the parameter m and the parameter X for the parameter m.
- FIG. 22 is a diagram showing the impulse response of the basic high-pass filter H msn.
- FIG. 23 is a diagram illustrating frequency-gain characteristics of the basic low-pass filters L4a4 and L4a4 (1).
- Figure 24 shows the calculation of the filter coefficients when the basic filters are cascaded. It is a figure for explaining the contents.
- FIG. 25 is a diagram showing the frequency-gain characteristics of the basic low-pass filter (L4a4) M.
- FIG. 26 is a diagram illustrating frequency-gain characteristics of the basic high-pass filter (H4s4) M.
- Figure 27 is a diagram schematically showing a method of designing a bandpass filter by cascading basic filters.
- Fig. 28 is a diagram showing a specific design example of a bandpass filter formed by cascading basic filters.
- Fig. 29 is a diagram showing a specific design example of a bandpass filter by cascading basic filters.
- FIG. 30 is a diagram schematically showing a means for narrowing the band width by cascading different types of basic filters.
- FIG. 31 is a diagram schematically showing a means for expanding a band width by cascading basic filters of the same type.
- FIG. 32 is a diagram schematically showing a means for finely adjusting the band width.
- FIG. 33 is a graph showing filter coefficient values (before rounding) actually calculated with 16-bit operation accuracy.
- FIG. 34 is a diagram illustrating a frequency characteristic of the digital filter before the filter coefficient is rounded.
- FIG. 6 is a diagram showing a coefficient value obtained by converting an integer into an integer.
- FIG. 36 is a diagram showing a frequency-gain characteristic when a filter coefficient is calculated with 16-bit operation accuracy, and then rounded to 10 bits and further converted to an integer.
- FIG. 37 is a flowchart showing a procedure of a digital filter design method according to the second embodiment.
- FIG. 38 is a frequency characteristic diagram for explaining the concept of the design method of the digital filter according to the second embodiment.
- FIG. 39 shows the frequency-gain characteristics of the original band-pass filter and one to three adjustments to this original band-pass filter; the frequency-gain characteristics obtained when the filters are cascaded.
- FIG. FIG. 40 is a diagram for explaining the principle of a change in frequency characteristics obtained when the adjustment filters according to the second embodiment are cascaded.
- FIG. 3 is a diagram showing frequency characteristics obtained.
- Figure 42 shows the frequency-gain characteristics of the original one-pass filter and the frequency-gain characteristics obtained when 1 to 5 adjustment filters are cascaded to this original one-pass filter.
- FIG. 43 is a flowchart showing a procedure of a digital filter design method according to the third embodiment.
- FIG. 44 is a flowchart showing a procedure of a method of generating a basic filter according to the third embodiment.
- FIG. 45 is a diagram illustrating the frequency-gain characteristics of the basic filter according to the third embodiment.
- FIG. 46 is a diagram illustrating frequency-gain characteristics of a basic filter according to the third embodiment and a plurality of frequency shift filters generated from the basic filter.
- FIG. 47 is a diagram illustrating an example of a frequency-gain characteristic of a digital filter generated by the filter design method according to the third embodiment.
- FIG. 4 is a frequency-gain characteristic diagram of FIG.
- FIG. 49 is a block diagram illustrating a configuration example of a digital filter design device according to the third embodiment.
- FIG. 50 is a block diagram illustrating a configuration example of a digital filter according to the first embodiment.
- FIG. 51 is a block diagram illustrating a configuration example of a digital filter according to the second embodiment.
- FIG. 52 is a block diagram illustrating a configuration example of a digital filter according to the third embodiment.
- Basic filters are broadly classified into three types: basic one-pass filters, basic high-pass filters, and basic bad-pass filters (including comb filters).
- the basic low-pass filter L man (where m and n are variables and n is a natural number)>
- the filter coefficient of the basic low-pass filter L man is a numerical sequence of "_ 1, m,-1" Is used as a starting point, a moving average calculation is performed by sequentially adding the original data before the calculation and the previous data only a predetermined delay before the calculation.
- the original data is (n — 1 )
- the data is (n ⁇ 1
- the filter coefficients of the m ports connected in cascade in two stages are 3.7 3 2, 1 4.92
- the filter coefficients of the basic mouth-pass filter Lman are all the sum of the numerical sequence is "1" and the sum of the numerical sequence is "1". It has the property that the sum is the same sign and equal to each other.
- the figure shows the frequency characteristics (frequency-gain characteristics and frequency-phase characteristics) obtained by performing FFT conversion on the numerical sequence of the filter coefficients in).
- the gain is represented by a linear scale, and the normalized gain is shown by multiplying it by 32, while the frequency is normalized by "1".
- the frequency-gain characteristics show that the passband is almost flat, and the slope of the cutoff region has a gradual waiting property.Also, the frequency-phase characteristics show almost linear characteristics. Has been. Thus, in the basic low-pass filter L4a4, there is no overshoot ringing.
- Figure 3 shows n of the basic low-pass filter L 4 an. It is a figure which shows the frequency-gain characteristic which made it into a lazy evening, and (a) expresses a gain by a straight line
- FIG. 4 shows the gain on a logarithmic scale. From Fig. 3, it can be seen that the slope of the cutoff area becomes steeper as the value of n increases.
- the basic mouth-to-pass filter L 4 an has a relatively sharp frequency characteristic when n ⁇ 5. It is appropriate to use a frequency characteristic that is relatively moderate when n ⁇ 5.
- the filter coefficient of the basic ⁇ one-pass fill Lan is“ 1 ”as the starting point, and the moving average is obtained by sequentially adding 7 pm and ⁇ te. It is obtained by calculation.
- Each of the filter coefficients of the basic low-pass filter L an shown in FIG. 4 has the property that its numerical sequence is symmetrical, and that the total value of each jump of the numerical sequence is the same sign and equal to each other (for example, the basic low-pass filer La
- FIG. 5 shows the numerical sequence of the filter coefficients of the basic low-pass filter La4 as F F
- FIG. 9 is a diagram illustrating frequency characteristics obtained by performing T conversion.
- the gain is represented by a straight line and a scaled gain is shown by 16 times.
- the frequency is normalized by "1".
- the almost flat passband in the frequency-gain characteristics is narrower than in Fig. 2, but the slope of the cutoff region has a gentle characteristic.
- the frequency-phase characteristics show almost linear characteristics.
- Figure 6 shows the frequency-gain characteristics of the basic low-pass filter with n as the parameter, where (a) represents the gain with a large linear scale and (b) the gain with a logarithmic scale. It is represented by From Fig. 6, it can be seen that the greater the value of ⁇ n, the steeper the slope of the cutoff region. It can be said that when n ⁇ 5, m is used for relatively steep frequency characteristics, and when n ⁇ 5, it is used for relatively smooth frequency characteristics.
- ⁇ Basic high-pass filter H msn (mn is a variable, n is a natural number)>
- the filter coefficient of the basic high-pass filter H msn starts from the numerical sequence of "1m, 1" and starts from the original data before the operation. It is obtained by a moving average calculation that sequentially subtracts the previous data by a predetermined delay amount.
- H 4 sn the basic high-pass filter
- the first numerical value "1" from the top of the basic high-pass filter H4s1 is obtained by subtracting the previous data "0" from the original data "1".
- the second number "3” is obtained by subtracting the previous data from the original data "4". Also, the third numerical value “1 3” is obtained by subtracting the previous data "4" from the original: r1 data 1 ", and the fourth numerical value 1
- the numerical sequence "1, m, 1" is generated based on the original numerical sequence "1, N".
- the basic unit filter that has this numerical sequence "1, N" as filter coefficients is an asymmetric type, so in order to make it a symmetric type, it is necessary to use an even-numbered cascade connection.
- the filter coefficients are "N, N + 1, N" due to the convolution of the numerical sequence "1, N".
- N (m + (m 2 — 4) 1/2 ) / 2.
- the filter coefficients of the basic high-pass filter H msn are each such that the sum of the numerical sequence is "0" and the sum of the jumps in the numerical sequence is one. Have the opposite sign and are equal to each other. .
- the gain is represented by a linear scale, and the normalized gain is shown by 32 times.
- the frequency is normalized by "1".
- the frequency-gain characteristics have a flat passband and a gentle slope in the cutoff region.
- almost linear characteristics are obtained in the frequency-phase characteristics.
- the basic high-pass filter H4s4 can obtain a good high-pass filter frequency characteristic without overshoot or ringing.
- Fig. 9 is a graph showing the frequency-gain characteristics of the basic high-pass filter H4sn, where n is a parameter, and (a) shows the gain by a linear scale.
- n is an odd number, the absolute value of the numerical sequence is symmetric, and the numerical sequence of the first half and the numerical sequence of the second half have a sign opposite to that of the ⁇ ⁇ numerical sequence.
- Fig. 11 shows the frequency characteristics obtained by performing an FFT transform on the numerical sequence of the filter coefficients of the basic high-pass filter Hs4.
- the gain is represented by a linear scale, and the normalized gain is shown by 16 times. Frequency is normalized by "1".
- FIG. 9 is a diagram showing frequency-gain characteristics in which n of the filter H s n is set to “n”, and (a) represents the gain by a linear scale;
- ⁇ Spherical H sn is suitable for applications with relatively steep frequency characteristics when n ⁇ 5, and is suitable for applications with relatively gentle frequency characteristics when n ⁇ 5. I can.
- the moving average is calculated by sequentially subtracting the previous ⁇ — evening from the original data.
- the original text refers to the j-th data from the top of the (n-1) -th column.
- the key is the (j-1 2) th data from the top of the (n-1) th column.
- the third number “3” is obtained by subtracting the previous data “1” from the original data "4", and the fifth number “_3” is obtained from the original data "1” and the previous data "4".
- N 2 + ⁇ 3.
- the coefficients of the basic unit filter are "1, 0, 3732" (three decimal places are displayed here).
- the filter coefficients are "3.732, 0, 14".
- the gain is represented by a linear scale, and the normalized gain is shown by 32 times.
- the frequency is normalized by "1".
- the frequency-gain characteristics are such that the passband is almost flat and the cutoff band has a gentle slope.
- almost linear characteristics are obtained in the frequency-phase characteristics.
- the basic band-pass filter B4s4 can obtain good frequency characteristics of the band-pass filter without overshoot or ringing.
- Fig. 15 shows the frequency-gain characteristics of the basic bandpass filter B4sn with n as a parameter.
- A shows the gain on a linear scale
- (b) shows the gain. Is represented on a logarithmic scale. From Fig. 15 it can be seen that the greater the value of n, the steeper the slope of the cutoff region. It can be said that this basic bandpass filter B4sn is suitable for applications with relatively steep frequency characteristics when n ⁇ 5, and is suitable for applications with relatively gradual frequency characteristics when n ⁇ 5. ⁇
- the basic band-pass filter B sn shown in Fig. 16 when n is an even number, the numerical sequence of any of the filter coefficients is symmetric, and the sum of the three values of the numerical sequence is the opposite sign.
- Figure 17 shows a numerical sequence of the filter coefficients of the basic bandpass filter Bs4.
- FIG. 3 is a diagram illustrating frequency characteristics obtained by FFT conversion.
- the gain is represented by a linear scale, and the normalized gain is shown by 16 times.
- the passband that is almost flat in the frequency-gain characteristic is narrower than that in Fig. 14, but the slope of the cut-off region has a gradual characteristic.
- almost linear characteristics are obtained, and even in the basic band-pass filter B s4, it is possible to obtain a good band-pass filter frequency characteristic in which neither omission nor ringing exists.
- Fig. 18 shows the frequency-gain characteristics of the basic band-pass filter B sn with n as a parameter.
- A shows the gain on a linear scale
- (b) shows the gain.
- the gain is shown on a logarithmic scale. It can be seen from FIG. 18 that the slope of the cutoff area becomes steeper as the value of n increases.
- This basic bandpass filter Bsn is suitable for applications with relatively steep frequency characteristics when n ⁇ 5, and is suitable for applications with relatively gentle frequency characteristics when n ⁇ 5.
- FIG. 19 is a diagram showing frequency-gain characteristics of a certain high-pass filter H msn using m as a parameter. From Fig. 19, it can be seen that the smaller the value of m, the steeper the slope of the cutoff band and the narrower the band width of the passband. Although not shown here, the same can be said for the basic mouth one-pass filter Lman and the basic bandpass filter Bmsn.
- Figure 20 shows this in an easy-to-understand graph.
- the M value of the parameter for a given m over a period of m is larger as the value of m is smaller.
- FIG. 9 is a diagram showing, in a tabular form, a relationship between evening m and an optimum value of parameter n.
- FIG. 21 also shows the relationship between the parameter m and the parameter z.
- the optimal value of the parameter n for the parameter m increases as the value of m decreases.
- m 2
- the number of stages of the moving average calculation may be one. Therefore, it is preferable that the parameter m is used under the condition of 2 ⁇ m ⁇ l0.
- the value of the parameter n is determined by using an arbitrary value selected within a certain range before and after the optimum value shown in Fig. 21 as the center, as shown in Figs. 3, 9, and 15. Can be adjusted.
- FIG. 22 is a diagram showing impulse responses of the four types of basic high-pass filters H msn shown in FIG.
- the impulse response having the waveform shown in Fig. 22 has a finite value other than "0" only when the sample position along the horizontal axis is constant, and the value in other regions Are all "0", that is, a function whose value converges to "0" at a given sampling position.
- the basic high-pass filter H sn, the basic low-pass filter L man, L an, and the basic band-pass filter B msn, B sn also have a finite impulse response.
- the band width of the pass band of the basic filter can be adjusted. is there.
- the delay between taps was one clock.
- this is (k + 1) clocks (when k “0s” are inserted between each filter coefficient)
- the frequency axis (period in the frequency direction) of the frequency-gain characteristic is 1 ( k + 1), and the band width of the passband becomes narrow.
- L man L man
- FIG. 9 is a diagram showing the frequency-gain characteristics of the basic low-pass filter L4a4 (1) generated by inserting one "0"'between numbers, and (a) shows the gain.
- (B) represents the gain in logarithmic scale. As can be seen from o in Fig. 23, assuming that the number of "0" inserted between the filter coefficients is k, the frequency The frequency axis (period in the frequency direction) of the gain characteristic is 1 / (k + 1), and it is possible to narrow the band width of the passband.
- FIG. 24 is a diagram for explaining the calculation contents of the filter coefficient that is connected / connected. As shown in FIG. 24, when two basic filters are cascaded, one filter coefficient is configured (
- Figure 25 is a diagram showing the frequency-gain characteristics of the basic low-pass filter Otsu 4 & 4, (L4a4), (L4a4) 4 , and (L4a4) 8 .
- ) Represents the gain on a linear scale
- (b) represents the gain on a logarithmic scale. If there is only one basic low-pass filter L4a4, the amplitude will be 0.5
- the clock at the right position is 0.25.
- M 8
- the clock at the position where the amplitude is 0.5 is 0.125.
- the basic single-pass filter L4a4 has a feature that the cutoff frequency portion of the frequency characteristic has a steep slope.
- the basic characteristic of the low-pass filter (L4a4) M has a characteristic that the passband becomes narrower as the number of cascaded connections M increases, and that the characteristic drops down to a very low rate even in the low-frequency range.
- FIG. 26 is a diagram showing the frequency-gain characteristics of the basic high-pass filters H 4 s 4, (H 4 s 4), (H 4 s 4) 4 , and (H 4 s 4) 8 .
- Gain is represented by a linear scale, and (b) represents gain by a logarithmic scale.
- the basic high-pass filter H 4 s 4 has the feature that the slope of the cutoff frequency portion of the frequency characteristic is steep.
- the frequency-gain characteristic of the basic high-pass filter (H4s4) M is such that the passband becomes narrower as the number M of cascade connections increases, and a characteristic is obtained in which, even in a high-frequency range, the frequency falls very deeply into a straight line.
- the configuration of the filter can be further simplified by optimizing the frequency sampling conditions. .
- the relationship between the center frequency F c of the band-pass filter and the sampling frequency F s of the signal is
- FIGS. 27A and 27B are diagrams schematically showing the design method of the above-described bandpass filter.
- the band width of the bandpass filter is adjusted by the number of cascade connections (number of M ) of the basic highpass filter (H4s4 (k)) M or the basic lowpass filter (L4a4 (k)) M. It is possible to do.
- FIG. 28 shows the frequency characteristics of the basic high-pass filter (H4s4 (8)) 8 and the basic single-pass filter (L4a4 (5)) 8 in an overlapping manner. By cascading evenings, only the overlapping parts can be taken out.
- Fig. 29 shows the extraction of the passband by LPF 1 or LPF 2. LPF 1 or LPF 2 is applied to the three band paths extracted as shown in Fig. 28. Thus, only the passbands at both ends can be extracted.
- FIG. 30 is a diagram schematically showing the method.
- Fig. 30 (a) is the same as Fig. 27 (b). If you want to obtain a smaller width, as shown in Fig. 30 (b),
- H 4 s 4 a basic high-pass filter H 4 s 4 (1
- the basic high-pass filter H 4 s 4 (1 4) has a passband in which the center frequency F c is 450 KH similarly to the basic octa-pass filter H 4 s 4 (8), and Band width is basic high pass fill evening H 4 s
- the band width can be efficiently narrowed without increasing the number of cascaded stages of the filter. Also, the basic high-pass filter H 4 s 4 (1 4)
- inverted basic filters shown in # 3 are connected in cascade.
- the slope of the frequency-gain characteristic obtained becomes steeper as shown in # 4, and the band width further narrows (the clock position of _6 dB moves to the higher frequency side).
- the number of inverting basic filters connected in cascade is two, the same as in # 2.By increasing the number, the amount of movement to the high frequency side is smaller than the amount of movement to the low frequency side. Can be larger.
- Equation 1 gives the basic filter of # 1 and the inverted basic filter of # 3
- Equation 1 gives the basic filter of # 1 and the inverted basic filter of # 3
- Equation 2 Equation 2 below.
- a and b are coefficients (a> b), M1 ⁇ M2, and * represents a cascade connection.
- FIG. 32 is a frequency-gain characteristic diagram for explaining a fine frequency adjustment method. As shown in Fig. 32, within the relatively wide pass band of the basic high-pass filter H4s4 (8), the eight-pass filter (HPF) and the low-pass filter (LPF) are set so that the pass bands overlap each other. ) And design
- the operation of narrowing the passband as shown in FIG. 25 and FIG. 26 or FIG. 30 for one or both of the eight-pass filter HPF and the one-pass filter LPF can be finely adjusted arbitrarily by performing the operation of expanding the passband as shown in FIG.
- FIG. 32 '(a) shows an example in which only one side of the zone-pass filter is shifted to the high-frequency side by performing an operation of widening the passband with respect to the low-pass filter LPF. Also, in Fig. 32 (b), eight paths
- both sides of the band pass filter are shifted to the low frequency side without changing the band width by performing an operation to narrow the pass band for the LPF.
- FIG. 33 shows a graph of the fill coefficient values (before rounding) actually obtained with 16-bit operation accuracy.
- Fig. 34 is a diagram showing the frequency-gain characteristics of the digital filter before the filter coefficient is rounded. (A shows the gain on a linear scale, and b) shows the gain on a logarithmic scale. I have.
- the value of the filter coefficient obtained by the design method of the present embodiment is maximum at the center (coefficient H.). Also, the difference between the values of the filter coefficients is extremely large compared to the filter coefficients obtained by the conventional filter design method. That is, the distribution of each filter coefficient obtained by the design method of the present embodiment has a larger value in a local region near the center, a smaller value in other regions, and a filter coefficient value near the center. The distribution has a high sharpness such that the difference between the coefficient and the surrounding coefficient value becomes extremely large. Therefore, even if a filter coefficient having a value smaller than a predetermined threshold value is discarded by the rounding process, most of the main filter coefficients that determine the frequency characteristic remain, and the frequency characteristic is hardly affected.
- the out-of-band attenuation of the frequency characteristic is restricted by the number of pits of the filter coefficient.
- the frequency characteristic obtained by the filter design method of this embodiment has a very deep attenuation. Therefore, even if the number of bits is slightly reduced, the desired attenuation can be secured.
- the number of filter coefficients designed based on this filter is smaller than before, and it is possible to use it as it is without performing rounding processing. In order to reduce the number of bits, it is preferable to perform a rounding process for reducing the number of bits.
- the sharpness is not so large in the distribution of the required filter coefficients obtained by the conventional filter design method
- the main filter coefficients that determine the frequency characteristics are also discarded. Many.
- it is difficult to obtain frequency characteristics with extremely deep out-of-band attenuation so if the number of filter coefficient bits is reduced, the necessary out-of-band attenuation cannot be secured. Therefore, conventionally, rounding processing to reduce the number of bits could not be performed, and the number of filter coefficients had to be reduced by windowing. Therefore, a truncation error occurs in the frequency characteristics, and it has been extremely difficult to obtain a desired frequency characteristic.
- the filter can be designed without windowing, there is no occurrence of a truncation error in the frequency characteristic. Excellent fill characteristics can be obtained.
- FIG. 7 is a diagram showing fill coefficient values corresponding to 4'1 steps (the number of steps including the zero value is 46 steps) remaining as a coefficient, and coefficient values obtained by converting them into integers.
- the value of the filter coefficient obtained by the cascade connection of the basic filters as described above is a decimal number.
- the number of digits can be reduced by rounding 0 bits, but it is a random set of values.
- This numerical sequence may be used as it is as a filter coefficient, but a multiplier used when implementing a digital filter In order to further reduce the number of, the numerical value of the filter coefficient may be further rounded and simplified.
- 1 0 the numerical sequence of filter coefficients rounded bit Bok 2 1 0-fold to, to integer coefficient values Do, the 1 to 6 lower 1 0 bit h of filter coefficients comprising a bit Bok the further 2 1 0 times the filter coefficients rounded to 1 0-bi Tsu Bok after rounding an example has been described in which integer, 1 6 directly 2 1 (1 multiplies its binding filter coefficients consisting of bits By rounding (rounding down, rounding up, or rounding down) the resulting value, the integer 10-bit filter coefficient may be directly obtained,
- the output from each evening of the extended line with a tap consisting of a plurality of delay units (D-type flip-flops) 1 is obtained. individually multiplies the integer fill evening engagement by a plurality of coefficient 2 to the signal, together 1/2 1 0 in one shift computing unit 4 after every pressing the respective calculated power by a plurality of adders 3 It can be configured to double.
- the integer filter coefficient is 2 '
- FIG. 9 is a diagram showing frequency-gain characteristics when the result is further converted to an integer.
- the ripple of the flat portion in the frequency-gain characteristic is extremely small, and ⁇ 0- It is well within the range of 3 dB, and the out-of-band attenuation after rounding is about 44 dB, but this amount of out-of-band attenuation is handled by the implementation of the noise detector. Limited by the number of possible bits. Therefore, if there is no restriction on the scale of 8-dwell, it is possible to obtain an out-of-band attenuation characteristic with a deeper attenuation by increasing the number of bits after rounding.
- the processing of rounding the data of y bits to X bits by truncating the least significant bits from the data of the filter coefficients has been described. Not limited. For example, if the value of each filter coefficient is compared with a predetermined threshold value and the filter coefficient smaller than the threshold value may be discarded, the remaining filter coefficient remains the original y bit. To convert this to an integer, multiply it by 2 y
- the numerical value sequence of the filter coefficient may be multiplied by N (N is a value other than a power of 2) and the decimal part may be rounded (rounded down, rounded up, rounded, etc.).
- N is a value other than a power of 2
- the decimal part may be rounded (rounded down, rounded up, rounded, etc.).
- the digital fill becomes as shown in Fig. 51, from each tap of the tapped delay line consisting of multiple delay units (D-type flip-flops) 1.
- the output signal is individually multiplied by an integer filter coefficient by a plurality of coefficient units 2, and the respective multiplied outputs are all added by a plurality of adders 3, and then a single multiplier 5 collectively outputs 1 N It can be configured to double.
- the filter coefficient of the integer is a binary number such as., 2 1 + 2 + ... (i and j are arbitrary integers). Can be expressed by addition of This makes it possible to configure a coefficient unit using a bit shift circuit instead of a multiplier, thereby simplifying the configuration of a digital filter to be mounted.
- Bit-by-bit rounding means that, for example, when a coefficient value is multiplied by 2 x and the fractional part is rounded down, all numbers in the range 2 x to 2 x + 1 are rounded to 2 x. The process of setting the value to an integral multiple of 1/2 X.
- rounding between bits means that, for example, when a coefficient value is multiplied by N (for example, 2 X — 1 and N ⁇ 2 X ) and the fractional part is truncated, it belongs to the range of N to N + 1.
- N for example, 2 X — 1 and N ⁇ 2 X
- the process of making the coefficient value an integer multiple of 1 ZN such as rounding all numbers to N.
- the value of the filter coefficient to be converted to an integer can be adjusted to an arbitrary value other than a power of two. In this way, the number of filter coefficients (the number of taps) used in the digital filter can be finely adjusted.
- the digital filter applies the output signal from each tap of the tapped delay line consisting of a plurality of delay units (D-type flip-flops) 1 as shown in Fig. 52.
- integer filter coefficients can be represented by binary addition, such as 2 1 + 2 1 + ⁇ ⁇ ⁇ (. I, j are arbitrary integers). .
- a coefficient unit with a bit shift circuit instead of a multiplier, and to simplify the configuration of a digital filter to be mounted.
- the number of filter coefficients (the number of taps) can be significantly reduced, and the number of bits is larger than that of X bits. Since a filter coefficient with (X + X) bits with high precision can be obtained, a better frequency characteristic can be obtained.
- An apparatus for realizing the above-described method of measuring and measuring anthalophile according to the present embodiment can be realized by any of a hardware configuration, DSP, and software.
- the filter design apparatus of the present embodiment may be
- M or hard disk - ⁇ -V This can be realized by running the L'feed program.
- Filter coefficients related to B msn and Bsn are stored as data in a storage device such as RAM, ROM, or hard disk. Then, the user selects any combination and connection order of the basic filers Lman, Lan, Hmsn, Hsn, Bmsn, Bsn, the number k of zero values inserted between each filter coefficient, and the basic filter.
- the CPU uses the filter coefficient data stored in the storage device to calculate the filter coefficient corresponding to the specified content by the above-described calculation. It is possible to In this case, the storage device corresponds to the basic filter coefficient storage means of the present invention, and the CPU corresponds to the calculating means of the present invention.
- the user sets each basic filter Lm-an, Lan, Hms, n, Hsn, Bm
- the user interface for designating the combination of sn and BS n and the connection order, the number k of opening values, the number M of cascade connections, and the like can be arbitrarily configured.
- the basic filter type (L man, L an
- H msn H sn, B msn, or B sn can be selected by operating the mouse from the keypad and the parameters m, n, k, and Input the value of ⁇ by keyboard or mouse operation. Then, the input order when the type selection and the parameter input are performed one by one is input as the connection order of the basic filter.
- the CPU obtains the information thus input, and obtains the filter coefficient corresponding to the content specified by the input information by the above-described calculation.
- the obtained filter coefficient is automatically FFT-transformed, and the result is It may be displayed on a display screen as a 'single gain characteristic' diagram. In this way, the frequency characteristics of the designed filter can be visually confirmed, and the filter can be designed more easily.
- an FIR filter having a numerical sequence finally obtained by the filter design device as a filter coefficient may be configured. . Ie
- the number of obtained filter coefficients is greatly reduced by rounding 10 bits, and is converted to a simple integer by doubling. Therefore, the number of steps is very small, and basically it is not necessary to provide a multiplier in the part of the coefficient unit 2 and a bit shift circuit can be used, and the desired frequency characteristics can be reduced with a small circuit scale. It can be realized with high accuracy.
- the basic filters used in the filter design may be configured as eighty-one duels, and the digital filters may be implemented by connecting them as hardware.
- one or more basic filters are arbitrarily combined to calculate filter coefficients in a cascade connection, and furthermore, unnecessary filter coefficients are largely deleted by rounding.
- the number of taps can be significantly reduced as compared to the conventional FIR filter.
- the digital filter can be designed without windowing, there is no occurrence of a truncation error in the frequency characteristics. Therefore, a desired frequency characteristic of the digital filter can be realized with high accuracy.
- FIG. 37 is a flowchart showing the procedure of a digital filter measuring method according to the second embodiment.
- FIG. 38 is a frequency characteristic diagram for explaining the concept of the digital filter design method according to the second embodiment.
- a numerical sequence generates a first filter coefficient having a symmetric type (step S 1).
- the method for generating the first filter coefficient is not particularly limited. If the numerical sequence of filter coefficients is symmetric, a conventional design method using an approximate expression or window function may be used. Also, a plurality of amplitude values representing desired frequency characteristics are input, the input numerical sequence is subjected to inverse Fourier transform, and the obtained numerical sequence is windowed, thereby obtaining a first filter. The luta coefficient may be obtained. Further, the design method described in the first embodiment may be used. Preferably, the first filter coefficient is generated using the design method described in the first embodiment (excluding the rounding processing).
- the frequency characteristic indicated by reference symbol A in FIG. 38 shows an example of the frequency-gain characteristic of the original filter realized by the first filter coefficient generated in step S1.
- the symmetrical second filter coefficient that realizes the frequency-gain characteristic (B in Fig. 38) that has a contact at the position where the maximum value is obtained at A) in Fig. (Step S2). If the frequency-gain characteristic has such a characteristic, the second fill coefficient may be generated by any method. For example, the second fill coefficient can be obtained by the following calculation.
- the filter having the second filter coefficient is referred to as an “adjustment filter”.
- Step S3 By cascading the original filter and the adjustment filter, the first filter coefficient and the second filter coefficient are multiplied and added to generate a new filter coefficient.
- the operation content of the cascade connection is as described in the first embodiment.
- step S For the generated third filter coefficient, unnecessary filter coefficients are greatly reduced by rounding to reduce the number of bits, and the filter coefficients are simplified by integer conversion (step S). Four ) .
- the coefficient values directly 2 x magnification or N
- the process of reducing the number of bits in the filter coefficient and the process of converting the coefficient value to an integer can be performed simultaneously by one rounding operation. You may do it. If the y-bit coefficient value is less than 1 x 2 x , it is assumed to be zero, and if the coefficient value is 1 x 2 x or more, the coefficient value is multiplied by 2 x + x (X + X ⁇ y). Then, by performing a process of rounding off the decimal point, a (x + X) -bit digitized filter coefficient may be obtained.
- windowing as in the related art is not necessarily required to reduce the number of filter coefficients.
- Windowing — Filter design can be done in no time, so there is no truncation error in frequency characteristics. Therefore, it is possible to greatly improve the cutoff characteristics, and to obtain excellent filter characteristics with a linear phase characteristic.
- step S3 the third filter coefficient generated in step S3 is newly added to the first filter.
- the process returns to step S2 assuming that the filter coefficient is used.
- the second filter coefficient is again calculated. Ask (generate a new adjustment filter).
- a new third filter coefficient obtained when a new adjustment filter is further cascaded is obtained. Calculate filter coefficients. After repeating such an operation as many times as the number of adjustment filters to be connected in cascade, the rounding process in step S4 is performed on the third filter coefficient generated in step S3 in the final stage. .
- Fig. 39 shows the frequency-gain characteristics of the original filter (band-pass filter) and the frequency-gain characteristics obtained when one or three adjustment filters are cascaded to this original filter.
- 41 is the frequency-gain characteristic of the original filter.
- this Fig. 39 shows the frequency characteristics when the value of the parameter ⁇ for obtaining the second filter coefficient from the first filter coefficient is set to 1.5.0 shown in Fig. 39
- ⁇ ⁇ the frequency characteristic Overshoot ringing at the top occurs.
- a 1, overshoot and ringing do not occur at the top of the frequency characteristic, and the characteristic becomes flat.
- FIG. 40 is a diagram for explaining the principle of a change in frequency characteristics obtained when the adjusting filters of the present embodiment are cascaded.
- FIG. 40 is for explaining the basic principle. However, it does not match the waveform of the frequency characteristic shown in FIG. FIG. 40 shows the principle when a 1 is set.
- Figure 40 (a) shows the change in frequency-gain characteristics when the first adjustment filter is cascaded to the original filter.
- A is the frequency of the original filter minus the gain characteristic
- B is the frequency of the first adjustment filter having the second filter coefficient generated from the first filter coefficient of the original filter.
- C shows the frequency-gain characteristics obtained when the original filter and the first adjustment filter are cascaded.
- the new frequency-gain characteristic C is the frequency-gain characteristic A of the original filter and the frequency-gain characteristic B of the adjustment filter. Is multiplied by.
- the second adjustment filter is further connected in cascade, the third filter coefficient corresponding to the frequency-gain characteristic C thus generated is newly used as the first filter coefficient, and the second adjustment filter is used. Find a new second filter coefficient for the filter.
- FIG. 40 (b) shows a change in the frequency-gain characteristic when the second adjustment filter is further cascaded.
- a ' is the frequency-gain characteristic when the first adjustment filter is cascaded, and the frequency-gain characteristic obtained in the procedure shown in Fig. 40 (a). Same as C. Same Gender number
- B is the second adjustment filter having a new second filter coefficient generated from the new first filter coefficient corresponding to the 'frequency-gain characteristic A'. It is.
- C ' is a new frequency-gain characteristic obtained when the second adjustment filter is further cascaded, and is a form obtained by multiplying the two frequency-gain characteristics A' and B '. ing
- the filter coefficient corresponding to the new frequency-gain characteristic C generated in the procedure of FIG. Using as the first filter coefficient, a new second filter coefficient for the three adjustment filters is obtained. And there are few back lines.
- a new frequency-gain characteristic is obtained according to the same procedure as described above.
- the pass band width of the filter can be increased and the stopband slope can be made steeper.
- Fig. 41 shows the frequency characteristics obtained when three stages of adjustment filters (1.5) are connected in cascade to the original filter, and a further adjustment filter (1) is connected in the last stage.
- a design example of a band-pass filter has been described.
- a mouth-pass filter, a high-pass filter, and the like can be designed in a similar procedure.
- Fig. 42 shows the frequency-gain characteristics of the original low-pass filter and the frequency-gain characteristics obtained when 1 to 5 adjustment filters are connected in cascade to this original one-pass filter.
- 51 is the frequency-gain characteristic of the original low-pass filter
- 52 to 56 are the frequency-gain characteristics obtained when one to five adjustment filters are connected in cascade.
- the adjustment filter is cascaded to widen the pass band of the filter, and to reduce the stop band.
- the inclination can be steep. Also, by increasing the number of cascaded adjustment filters, it is possible to obtain a filter characteristic having a wider passband and a steeper slope.
- the device described above can be realized by any of the following methods: DSP, DSP, and software.
- DSP digital signal processor
- DSP digital signal processor
- software for example, when using a software Xa
- the filter design apparatus of the present embodiment is actually configured by a CPU or MPU, RAM, R ⁇ M, etc. of a computer, and is stored in a BD, RAM, R ⁇ M, or ⁇ ⁇ ⁇ ⁇ disk. Can be realized by running a program
- Determining the first filter coefficient can be configured in the same manner as in the first embodiment. That is, filters relating to various basic filters L man, L an, H ms ⁇ , H sn B msn, and B sn The coefficient is stored in the storage device as an overnight message. Then, the user sets the basic filter L m a ⁇ ,
- the CPU multiplies all the coefficients other than the median value of the numerical sequence by one, and only the median value is multiplied by - ⁇ . It is possible to do this by adding 1 + h).
- the third filter coefficient by cascade connection can be obtained from the first filter coefficient and the second filter coefficient by the CPU performing the above-described calculation shown in FIG. .
- the rounding of the filter coefficients can be automatically performed by CPU.
- the first filter coefficient calculation, the second filter coefficient calculation, and the third filter coefficient are calculated.
- the desired operation, the first It is also possible to perform an operation for rounding the filter ⁇ number of 3.
- the calculation is actually performed by a CPU, R ⁇ M, RAM, or the like of a personal computer or the like on which the spreadsheet software is installed.
- the obtained filter coefficient may be automatically subjected to FFT conversion, and the result may be displayed on a display screen as a frequency-gain characteristic diagram. In this way, the frequency characteristics of the designed filter can be visually confirmed, and the filter can be designed more easily.
- the original filter and the adjustment filter may be configured as hardware, respectively, and the digital filter may be mounted by connecting them as hardware.
- FIG. 43 and FIG. 44 are flowcharts showing procedures of a method for designing a digital filter according to the third embodiment.
- FIGS. 45 to 48 are frequency characteristic diagrams for explaining the concept of a digital filter design method according to the third embodiment.
- FIG. 43 is a flowchart showing an overall processing flow of the digital filter design method according to the third embodiment.
- a numerical filter of filter coefficients generates a symmetric basic filter (step S1).
- This basic filter has a frequency-gain characteristic having a pass bandwidth of the sampling frequency f s (J3 is an integer of 1 or more) of the signal to be filtered.
- Figure 45 shows the frequency-gain characteristics of the basic filter.
- the 4 5 illustrates the frequency one gain characteristic of a basic filter having a bandwidth of half 1 2 8 equal portions of sampling frequency f s.
- the basic filter group is set so that adjacent filter groups overlap in the area of amplitude 1Z2.
- a plurality of frequency shift filters in which the frequency-gain characteristics of the filter are shifted by a predetermined frequency are generated (step S12). This frequency shift can be performed by the following calculation.
- Hj r H j 0 * 2 cos (27 T rj / ( ⁇ / 2))
- FIG. 46 shows the frequency-gain characteristics of the plurality of frequency shift filters generated in step S12 (the dotted line indicates the frequency-gain characteristics of the basic filter).
- bandwidth is sampling frequency of the basic filter: when is that half of the f s 1 2 8 divided, as one example, basic filter frequency The total is 128, including the shift filter.
- the frequency range determined by the number of filters generated here is the design area for the final product, the digital filter.
- step S13 For example, when adding the (a + 1) th frequency shift filter counting from the basic filter to the (a + 1) th frequency shift filter, the obtained filter coefficient is
- Fig. 47 is a diagram showing an example of the frequency-gain characteristics of the azimuth filter generated in step S13 of Fig. 47. In Fig. 47, the scale of the frequency axis is shown in the figure. 4 5 The compression ratio is much larger than that of Fig. 46.
- the figure shows the frequency characteristics of a digital filter generated by extracting a plurality of filters corresponding to 338 and adding those filter coefficients with corresponding coefficient codes.
- the fills adjacent to each other have a half width of te.
- step S 14 For the filter coefficients generated in step S13, unnecessary filter coefficients are significantly reduced by rounding to reduce the number of bits, and the filter coefficients are simplified by integer conversion (step S 14 4) Note that, similarly to the first embodiment, it is not necessary to separately perform the process of reducing the number of bits of the filter coefficient and the process of converting the coefficient value into an integer, and directly multiply the coefficient value by 2 ⁇ . Or ⁇ times the resulting value By rounding (rounding down, rounding up, or rounding down), the process of reducing the number of filter coefficient bits and the process of converting the coefficient value to an integer may be performed simultaneously by one rounding operation.
- the coefficient value of the y bit is smaller than 1/2 x , the coefficient value is set to zero, and if the coefficient value is 12 x or more, the coefficient value is multiplied by 2 x + x (X + X ⁇ y). It is also possible to obtain a (x + X) -bit integerized filter coefficient by performing a process of rounding off the decimal part.
- windowing as in the related art is not necessarily required to reduce the number of filter coefficients. Since the filter can be designed without windowing, there is no truncation error in the frequency characteristics. Therefore, it is possible to greatly improve the cutoff characteristic, and obtain an excellent filter characteristic with a linear phase characteristic.
- the method of generating the basic filter in step S11 is not particularly limited. As long as the numerical sequence of the filter coefficients is symmetric, various generation methods can be applied. For example, a conventional design method using an approximate expression or a window function may be used. Further, a design method of performing inverse Fourier transform on a plurality of amplitude values representing desired frequency characteristics may be used. Further, the design method (excluding the rounding process) described in the first embodiment may be used.
- FIG. 44 is a block diagram showing an example of a basic file generation process.
- the basic filter as in the first embodiment having a symmetric basic numerical sequence as a filter coefficient, a plurality of “0” s are inserted between the numerical sequences. Adjust the filter bandwidth by
- the frequency of a single-pass filter having a passband having a bandwidth obtained by equally dividing half of the sampling frequency f s into 128 is used.
- One gain characteristic is obtained.
- the frequency characteristic of a continuous wave with 128 passbands in the band lower than the center frequency is obtained, so a basic filter as shown in Fig. 45 is constructed from this continuous wave. It is necessary to cut out the frequency characteristics of a single wave. This extraction is performed in steps S22 and S23 described below.
- a window filter WF as shown in FIG. 48 is generated (step S22).
- This window filter WF has a common passband only with the passband of a single wave to be extracted as a basic filter as shown in FIG. Then, a basic filter as shown in FIG. 45 is extracted by cascading such a window filter WF and a basic port—pass filter L 4 a 4 (127). 3).
- the cascade connection between the window filter WF and the basic single-pass filter L4a4 (127) can be performed by calculating the filter coefficients as described in FIG.
- the generation method of the window filter WF is not particularly limited, and various generation methods can be applied. As an example, a plurality of amplitude values representing the frequency-characteristics of the window filter WF are input, and the input numerical sequence is inversely Fourier transformed.
- the filter WF if the passband includes all of the passbands of only the basic filter, it is not required to be a higher degree, so the input data of the numerical sequence 3 ⁇ 4 (the filter coefficient of the window filter WF) Need not be so large.
- the numerical value of each sample point may be directly input, or the waveform of the desired frequency characteristic on a two-dimensional input coordinate for representing the frequency-gain characteristic may be input. It is good to replace the drawn and drawn waveform with the corresponding numerical sequence. If the latter input method is used, it is possible to input the data while ensuring the desired frequency gain as an image. To obtain the desired frequency characteristics.
- a two-dimensional plane representing the frequency gain characteristic is displayed on the display screen of the combination display, and a waveform of a desired frequency characteristic is drawn on the two-dimensional plane using a GUI (Graphical User Interface) or the like.
- GUI Graphic User Interface
- a pointing device such as a digitizer or a plotter may be used.
- the method described here is merely an example, and a numerical sequence may be input by another method.
- the desired frequency-gain characteristic is input as a numerical sequence here, it may be input as a function representing the waveform of the frequency-gain characteristic.
- the filter coefficients of a plurality of frequency shift filters are further obtained by calculating the frequency shift. Then, one or more arbitrary filters are extracted from the basic filter and the plurality of frequency shift filters, and the filter coefficients are added by corresponding coefficient numbers, whereby a new filter coefficient is obtained. Ask. By arbitrarily changing the filter to be extracted, a digital filter having an arbitrary frequency characteristic can be generated.
- Fig. 47 shows an example of the generation of a mouth filter having a trap V in part, but in addition to this, a mouth filter having a passband in an arbitrary frequency band can be used. It is possible to generate band filters, band filters, and band filters, and it is also easy to generate comb filters and other digital filters with special frequency characteristics. If the number of divisions (number of [] 3) is increased when the basic filter is generated, the slope of the stop band of the basic filter and the individual frequency shift filters will increase, and the filter design error will increase. Since the resolution for the filter is also high, it is possible to generate a digital filter that precisely matches the desired frequency characteristics.
- the device for realizing the file design method according to the third embodiment described above can also be realized by any of the hardware configuration, DS #, and software.
- DS # hardware configuration
- software software
- the file design apparatus of this embodiment is actually composed of a computer CPU or MPU, RAM, ROM, etc., and has a program stored in RAM, R ⁇ M, or an octal disk. Can be realized by operating
- an operation for obtaining a basic filter 3 ⁇ 4r an operation for obtaining a frequency shift filter, an operation for obtaining a basic filter and a plurality of frequency shift filters. It is also possible to perform a calculation of adding the filter coefficients of those arbitrarily selected from. In this case, the calculation is y CPU of a personal computer or the like on which the fan is installed
- the filter coefficients of the basic filter and the filter coefficients of the plurality of frequency shift filters are calculated in advance, and stored in the pu1 unit.
- Fig. 49 is a block diagram showing a configuration example of a digital filter design device on the spot.
- reference numeral 61 denotes a filter coefficient table, which is a filter coefficient group including a filter coefficient of the above-described basic filter and a filter coefficient of a plurality of frequency shift filters (all frequency bands constituting a filter design area). It takes into account the table values of the region's coefficient group.
- the numbers on the horizontal axis indicate the filter numbers. That is, the 0th column shows the filter coefficient of the basic filter, and the 1st and subsequent columns show the frequency shift filter and the filter coefficient of fe.
- 62 is the controller, and 2 is the controller. Take control
- the operation unit 13 is composed of input devices such as a keyboard and a mouse, for example. 64 is turned on the display unit to display a selection screen for selecting one or more files. . Is to display the column number of the filter coefficient table 61 and select one of them, or to display the waveform of the frequency characteristic as shown in Fig. 46 and select one of them. Good
- Reference numeral 65 denotes an arithmetic unit, which is a filter selected from the basic filter and a plurality of frequency shift filters by the operating unit 13 by the operation unit 13 (the filter 12 is a filter coefficient table 11). Corresponding) The filter coefficient of the digital filter is obtained by adding the coefficient numbers.
- the operation unit 65 rounds the y-bit data to X bits by truncating the lower-order bits of the filter coefficient data obtained in this manner. Is also multiplied by 2 x to round the decimal point.
- the filter coefficients of this filter and a plurality of frequency shift filters in advance and converting them to a table By obtaining the filter coefficients of this filter and a plurality of frequency shift filters in advance and converting them to a table, the filter coefficients of the filter selected by the user operating the operation unit 63 can be simply obtained.
- a desired digital filter can be designed with only a very simple operation of adding.
- a digital filter is mounted in an electronic device or in a semiconductor IC.As shown in Fig. 50 to Fig. 52, the final filter is determined by the filter design described above. What is necessary is just to configure an FIR filter having the numerical value sequence as a filter coefficient. In this case, the number of found filter coefficients is greatly reduced by rounding, and is converted to a simple integer.
- a multiplier is basically unnecessary, and a bit shift circuit can be used, and a desired frequency characteristic can be realized with a small circuit scale and with high accuracy.
- the basic filter and the frequency shift filter may be configured as eighty-one and four, respectively, and the digital filter may be mounted by connecting them as eighteen-degree air.
- the third embodiment configured as described above, it is extremely simple to select one or more desired filters from the basic filter and a plurality of frequency shift filters generated from the filter and add the filter coefficients.
- it is possible to precisely design a digital filter having an arbitrary shape with frequency-gain characteristics.
- unnecessary filtering by rounding The number can be greatly reduced, and the filter coefficients can be simplified.
- a digital filter that realizes a desired frequency characteristic with high accuracy can be configured with an extremely small circuit scale.
- a low-pass filter is used as the basic filter and the frequency is shifted to the high frequency side
- the present invention is not limited to this.
- An eight-pass filter may be used as the basic filter and the frequency may be shifted to the low frequency side, or a band-pass filter may be used as the basic filter and the frequency shifted to the low and high frequencies.
- the arithmetic unit 63 selects the filter coefficient of one or more files selected by the operation unit 13 (the controller 12 reads the filter coefficient from the filter coefficient table 11). ), Add an arbitrary weight to each of the selected one or more filter coefficients when calculating the new filter coefficients. In this way, it is possible to extremely easily design a digital isolator having a frequency-gain characteristic of a breath shape in which only a specific frequency band is emphasized or attenuated. Furthermore, a graphic equalizer or the like utilizing this characteristic can be easily designed.
- each of the first to third embodiments is merely an example of a concrete embodiment for carrying out the present invention, and the technical scope of the present invention is limitedly interpreted. It must not be. That is, the present invention departs from its spirit or its main features; Variously It can be implemented in the form. Industrial applicability
- the present invention provides an FIR digital filter of a type that includes a delay line with taps composed of a plurality of delay units, multiplies the output signal of each tap by a filter coefficient, adds the multiplication results thereof, and outputs the result. Useful.
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Abstract
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GB0617380A GB2427093A (en) | 2004-02-17 | 2004-10-14 | Digital filter design method and device, digital filter design program, and digital filter |
US11/465,056 US20070053420A1 (en) | 2004-02-17 | 2006-08-16 | Method, apparatus, and program for designing digital filters |
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JP2010021860A (ja) * | 2008-07-11 | 2010-01-28 | Japan Science & Technology Agency | 帯域分離フィルタ及び帯域分離方法 |
JP2010041311A (ja) * | 2008-08-04 | 2010-02-18 | Japan Science & Technology Agency | フィルタ、フィルタの構成システム及び構成方法 |
JP2013520919A (ja) * | 2010-02-26 | 2013-06-06 | インダストリー−ユニバーシティー コオペレーション ファウンデーション ハンヤン ユニバーシティー | 周波数再構成が可能なデジタルフィルタ及びこれを用いたイコライザ |
US8949303B2 (en) | 2008-06-10 | 2015-02-03 | Japanese Science And Technology Agency | Filter |
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TW200501565A (en) * | 2003-05-15 | 2005-01-01 | Neuro Solution Corp | Digital filter and design method, design apparatus, and digital filter design program thereof |
KR100660841B1 (ko) * | 2004-10-22 | 2006-12-26 | 삼성전자주식회사 | 오버랩된 필터 뱅크들을 가지는 부분 탭 적응 등화기 및이를 이용한 등화 방법 |
IL178744A0 (en) * | 2006-10-19 | 2007-09-20 | Eci Telecom Ltd | Method for estimating bandwidth limiting effects in transmission communication systems |
KR101231080B1 (ko) * | 2008-07-30 | 2013-02-07 | 마이크로 모우션, 인코포레이티드 | 하나 이상의 디지털 필터들을 포함하는 프로세싱 시스템에서의 프로세서 동작의 최적화 |
CN102739195B (zh) * | 2012-06-06 | 2015-12-09 | 华为技术有限公司 | 一种fir滤波器的处理方法、装置和*** |
CN104954051A (zh) * | 2014-03-31 | 2015-09-30 | 富士通株式会社 | 脉冲成型滤波器的优化装置、发射机及方法 |
US9450601B1 (en) | 2015-04-02 | 2016-09-20 | Microsoft Technology Licensing, Llc | Continuous rounding of differing bit lengths |
JP7497659B2 (ja) * | 2020-09-23 | 2024-06-11 | ヤマハ株式会社 | Firフィルタを制御する方法および装置 |
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- 2004-10-14 JP JP2005517898A patent/JPWO2005078925A1/ja not_active Withdrawn
- 2004-10-14 CN CNA2004800427694A patent/CN1938947A/zh active Pending
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Cited By (4)
Publication number | Priority date | Publication date | Assignee | Title |
---|---|---|---|---|
US8949303B2 (en) | 2008-06-10 | 2015-02-03 | Japanese Science And Technology Agency | Filter |
JP2010021860A (ja) * | 2008-07-11 | 2010-01-28 | Japan Science & Technology Agency | 帯域分離フィルタ及び帯域分離方法 |
JP2010041311A (ja) * | 2008-08-04 | 2010-02-18 | Japan Science & Technology Agency | フィルタ、フィルタの構成システム及び構成方法 |
JP2013520919A (ja) * | 2010-02-26 | 2013-06-06 | インダストリー−ユニバーシティー コオペレーション ファウンデーション ハンヤン ユニバーシティー | 周波数再構成が可能なデジタルフィルタ及びこれを用いたイコライザ |
Also Published As
Publication number | Publication date |
---|---|
GB2427093A (en) | 2006-12-13 |
TW200529552A (en) | 2005-09-01 |
US20070053420A1 (en) | 2007-03-08 |
GB0617380D0 (en) | 2006-10-11 |
CN1938947A (zh) | 2007-03-28 |
JPWO2005078925A1 (ja) | 2008-01-10 |
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