JPS6412122B2 - - Google Patents

Description

DETAILED DESCRIPTION OF THE INVENTION The present invention relates to a digital filter device for an electronic musical instrument.

The timbre circuits in electronic musical instruments require delicate characteristics, so traditionally analog circuits were often used. However, analog timbre circuits (especially analog filters) tend to be large in scale, and are especially useful for tones that require a fixed formant (for example, human voices, wind instrument sounds such as oboes and bassoons, and other musical instrument sounds such as pianos and strings). (also has fixed formant characteristics), it was necessary to prepare many analog filter circuits in parallel, making the system large-scale. Furthermore, since it is not possible to directly input a digital musical tone signal to an analog tone color circuit, it is troublesome to apply a digital musical tone generating circuit.

This invention was made in view of the above points,
By configuring the timbre circuit of an electronic musical instrument using a digital filter, desired formant characteristics can be easily achieved with a small-scale and low-cost configuration, and a digital musical tone generation circuit can be easily applied. The purpose is to A further object of the present invention is to improve the conventional digital filter and provide a digital filter device suitable for the timbre circuit of an electronic musical instrument. Specifically, the object is to provide a digital filter device suitable for fine frequency characteristic control. Conventionally, the delay set at each stage inside a digital filter has been a unit delay (that is, a delay corresponding to one sampling time of a digital waveform signal). On the other hand,
The present invention is characterized in that a delay of one sampling time or more is added to each stage inside the digital filter where a unit delay is applied, thereby making it possible to easily realize fine frequency characteristics. .

Embodiments of the present invention will be described in detail below with reference to the accompanying drawings.

In FIG. 1, the keyboard section 10 includes, for example, an upper keyboard,
It includes a lower keyboard and a petal keyboard. The musical tone signal generating section 11 generates musical tone signals corresponding to keys pressed on the keyboard section 10, and can generate musical tone signals in a plurality of series depending on the type of keyboard, tone color, etc. The timbre selection device 12 includes a large number of switches for selecting timbres and various effects for each keyboard. A predetermined output of the output of the timbre selection device 12 is given to the musical tone signal generating section 11, and controls the musical tone signal generating operation in the generating section 11.
The musical tone signal generating section 11 outputs a plurality of series of musical tone signals corresponding to the type of keyboard, timbre, etc., in parallel and in digital format for each series. Of course, each series of musical tone signals is given a predetermined tone by the musical tone signal generating section 11 in accordance with the tone selection made by the tone selection device 12, but depending on the series, tone imparting may not be completed. Tone color control is performed by the digital filter section 14 at the subsequent stage. For example, a tone that always has the same spectral distribution regardless of the pitch (a so-called moving formant tone) is generated by the musical tone signal generator 11, and a fixed formant tone is generated by the digital filter section 14. Even for moving formant tones, such as the low-frequency characteristics of brass systems or the complex characteristics of string systems, it is preferable to perform spectral correction by further applying fixed formant filter control. The digital filter unit 14 is also used for these tones.

The digital musical tone signals for each series outputted from the musical tone signal generating section 11 are provided to a musical tone signal distribution and addition control circuit 13. This control circuit 13 is supplied with a predetermined output from among the outputs of the timbre selection device 12. The control circuit 13 includes the timbre selection device 1
According to the timbre selection information given from 2, the musical tone signals of each series are sorted into those that can be added and those that should be passed through the digital filter section 14, and those musical tone signals that can be added are added. Those to be outputted to line 15 and passed through digital filter section 14 are time-divisionally multiplied between each series of digital musical tone signals and output to a common signal line 16. Note that the control circuit 13 may serialize each series of digital musical tone signals and output the serial digital musical tone signal to one signal line 16.

The musical tone signal on line 15 is applied to a mixing circuit 17, and the musical tone signal on line 16 is applied to mixing circuit 17 via digital filter section 14. The mixing circuit 17 is for mixing (digital addition) the musical tone signal filtered by the digital filter section 14 and the musical tone signal of the line 15 which is not filtered. Since the signals are time-division multiplexed, these musical tone signals are demultiplexed for each series and then the above-mentioned mixing is performed. Note that the "allocation" and "addition" operations in the control circuit 13 and the "demultiplexing" operations in the mixing circuit 17 can be easily performed using known digital technology, so detailed explanation thereof will be omitted. . The digital musical tone signal outputted from the mixing circuit 17 is converted into an analog signal by a digital/analog converter 18 and is provided to a sound system 19.

A predetermined output of the output of the timbre selection device 12 is given to a digital filter section 14, and filter characteristics for each series are set in accordance with the timbre selection. Therefore, the filter section 14 includes a filter coefficient internal ROM (ROM stands for read-only memory, the same applies hereinafter), and a predetermined filter coefficient is read from this internal ROM according to the tone selection information. It is adapted to be used in the filter section 14.

The control circuit 13 is designed to output a synchronization pulse SYNC in accordance with the reference timing for time-division transmission of musical tone signals to the line 16. This synchronization pulse SYNC is generated by the digital filter section 1.
4 and is used to control the filter operation in synchronization with the digital tone signal provided via line 16. This synchronization pulse
One period of SYNC corresponds to one sampling time of the digital musical tone signal on line 16.

An example of the digital filter section 14 is shown in FIG. The digital filter section 14 includes a digital filter main circuit consisting of N stages (N is an arbitrary natural number) of digital filter units _L1 to _LN connected in series, and a filter coefficient supply circuit 20 that supplies filter coefficients to this main circuit. and a timing signal generation circuit 21 that supplies an arithmetic control timing signal to this main circuit. The digital musical tone signal applied from _the control _{circuit 13} in FIG. Tier (Nth tier) unit
Output from _LN . Timing signal generation circuit 2
1 generates various timing signals for controlling time-division filter calculation operations in each filter unit _L1 to _LN based on a synchronization pulse SYNC, and supplies these signals to each unit _L1 to _LN . . The filter coefficient supply circuit 20 includes the aforementioned filter coefficient internal ROM 22, and reads predetermined filter coefficients from the ROM 22 in accordance with the timbre selection information provided from the timbre selection device 12 (FIG. 1). A set of filter coefficients corresponding to one timbre consists of N filter coefficients corresponding to each unit _L1 to _LN . Such a set of filter coefficients is read out corresponding to each coefficient depending on the timbre selected in each coefficient. The filter coefficient supply circuit 20 receives the synchronization pulse SYNC, and in synchronization with the time-division calculation timing of the digital tone signal of each coefficient in each filter unit _L1 to _LN , corresponds to the tone selected in each series. The aforementioned filter coefficients are supplied to the individual units _L1 to _LN , respectively.

Any type of digital filter may be used as the digital filter units _L1 to _LN . Basic types of digital filters include the Latisse filter, finite impulse response filter (hereinafter referred to as FIR filter), and infinite impulse response filter (hereinafter referred to as IIR filter), among others, the Latisse filter is suitable for speech synthesis. It is known to be a filter. Moreover, compared to other types, this Lattice filter requires fewer multipliers and has the advantage of being able to downsize the hardware. It also requires fewer bits for the filter coefficients and can achieve the desired filter characteristics. It has the advantage that the method for setting coefficients has been established. Therefore, as a preferable example, a digital filter unit is
First, a case will be described in which a latex type filter is used for _L1 to _LN .

The basic model of the Lattice filter is as shown in FIG. In the figure, numerals 23 to 29 are adders or subtracters, 30 to 36 are multipliers, and 37 to 42 are delay circuits. In the figure, one filter unit is shown, and a suitable number of these units are connected in cascade to form a filter circuit. K _i ,
-K _i , 1-K _i , 1+K _i are filter coefficients to be multiplied by each multiplier, and the subscript i indicates the coefficient of the i-th filter unit.

Delay circuits 37 to 42 in each filter unit
is for setting a predetermined delay time for filter calculation, and conventionally, this delay time has been one sampling time (unit delay) of a digital waveform signal. That is, in the conventional digital filter, 1
The signal before the sampling time is fed back to the filter unit in the previous stage (however, the filter unit in the final stage receives a signal whose output signal is delayed by one sampling time in delay circuits 38, 40, and 42). ).
In contrast, the present invention is characterized in that the delay time in each filter unit is set not to one sampling time but to an appropriate integer of two or more sampling times.

In the latex filter shown in FIG. 3, c
The type shown in Figure 1 requires the least number of multipliers and has a simple configuration. Therefore, if the filter units _L1 to _LN shown in FIG. 2 are constituted by the latex type filters shown in FIG. 3c, the result will basically be as shown in FIG. 4. The first stage filter unit shown in the same figure
In L ₁ , code 43 is a subtractor, 44 and 45
is an adder, 46 is a multiplier, and 47 is a delay circuit. FS-IN is a musical tone signal input terminal, FS-OUT is a musical tone signal output terminal, BS-IN is a feedback signal input terminal, and BS-OUT is a feedback signal output terminal. Final stage filter unit L _N
The other units _L2 to LN _-1 except for this have the same configuration as unit _L1 , and the musical tone signal output terminal FS-OUT of each unit _L1 to _LN-1 is connected to the next stage unit _L2.
Connected to the musical tone signal input terminal FS−IN of L to _N ,
The feedback signal output terminal BS-OUT of each unit _L2 to _LN is connected to the previous unit _L1 to _LN-1.
Connected to the feedback signal input terminal BS-IN. In the final stage filter unit _LN , its own output signal is fed back.

The negative input (-) of the subtracter 43 of filter unit _L1 is connected to line 16 (Figures 1 and 2).
A digital musical tone signal is inputted via the input terminal FS-IN. The plus input (+) of this subtracter 43 is supplied with a feedback signal from the next stage unit L ₂ via a delay circuit 47 . A multiplier 46 multiplies the output signal of the subtracter 43 by a filter coefficient K ₁ . filter coefficient
_K1 is supplied from the filter coefficient supply circuit 20 (Fig. 2), and is supplied from the other units _L2 to _LN.
Also, filter coefficients K ₂ to K _N corresponding to each are given. The output of the multiplier 46 is applied to an adder 44, where it is added to the input musical tone signal applied from the input terminal FS-IN. The output of this adder 44 is applied to the next stage filter unit via the output terminal FS-OUT. Further, the output of the multiplier 46 is given to an adder 45 and added to the signal given via a delay circuit 47. The output of this adder 45 is fed back to the preceding filter unit via the output terminal BS-CUT (however, in the first stage unit _L1 , since there is no preceding stage, it is virtually not given anywhere).

As mentioned above, the delay circuit 4 of each unit _L1 to _LN
7, a delay corresponding to an appropriate integer sampling time of two or more sampling times is provided. The symbol 2SD shown in the block of the delay circuit 47 in FIG. 4 indicates that a delay of two sampling times is performed, as an example.

The filter characteristics of a digital filter circuit consisting of a plurality of digital filter units _L1 to _LN as shown in Fig. 4 are determined by the number of stages and each stage (each unit).
It has been known that the filter coefficients K ₁ to K _N of L ₁ to L N ) are determined by the filter coefficients K 1 to K _N . However, it has not been elucidated that the signal delay time at each stage is involved in determining filter characteristics. Therefore, this point will be explained below.

First, a case will be described in which the delay in each filter unit is a unit delay, that is, one sampling time, as in the conventional case. Letting the impulse response in the digital filter be h(n),
The frequency response characteristic (transfer function) H(e ^j 〓) can be expressed using a Fourier series as follows.

H(e ^j 〓)= _∞ 〓 ⁿ⁼⁰ h(n)e ^-j 〓 ⁿ …(1) Here, n is a number indicating the order of continuous sampling timing, and the time when the impulse is given is Let n=0, and thereafter change sequentially to 1, 2, 3, 4, etc. every sampling time. When the delay at each stage in a Lattice filter as shown in FIG. 3 is a unit delay (one sampling time), the impulse response h(n) can be illustrated as shown in FIG. 5a, for example. This impulse response h(n)
Specifically, is determined by the filter coefficient K _i of each stage and the number of stages, but the details will not be particularly explained. As is clear from Fig. 5a, since it is a unit delay, pulses of levels a ₀ , a ₁ , a ₂ , etc. are output every sampling time in response to an impulse input at sampling timing n = 0. be done.

An example of the filter characteristic realized by the frequency response H(e ^j 〓) determined according to the impulse response h(n) in such a unit delay type digital filter is shown in FIG. 6a. _s is the sampling frequency of the digital waveform signal, and in a conventional unit delay type digital filter, the filter has filter characteristics that are symmetrical about the position of _{frequency s} /2, which is half of this sampling frequency s. It is known that properties can be obtained. Furthermore, in a digital filter, the number of peaks and valleys in the filter characteristics is _determined according to the number of stages of the filter unit. It is easier to set it so that it spreads almost uniformly over the entire band from OHZ to _s , and if you try to concentrate the distribution of peaks and valleys only in a predetermined band, you will need to increase the filter calculation accuracy and increase the number of data bits. In addition, special measures were required to set the coefficients. Generally, filter operation is performed using one of the symmetrical characteristics (the characteristic in the band from 0Hz to _s /2Hz), and the desired characteristic can be obtained in this band (O to _s /2Hz). The filter coefficients, the number of stages, and the connection type between each unit are set so that the

Next, as in this invention, each filter unit
A case where the delay in L ₁ to L _N is two or more sampling times will be explained. Note that a delay exceeding a unit delay is referred to as an interstage delay. Let h'(n) be the impulse response when there is an inter-stage delay of one sampling time in addition to the unit delay, that is, there is a delay of two sampling times in total.
The following relationship holds true for the impulse response h(n) in a unit delay type filter having the same filter coefficients, number of stages, and connection type between units.

If this impulse response h'(n) is illustrated, the fifth
It will look like Figure b. In other words, since there is a delay of 2 sampling times per stage, n
In response to an impulse input at sampling timing =o, pulses of levels a ₀ , a ₁ , a _{2 .} . . are output every two sampling times.

The frequency response of this digital filter L ₁ to _{L N} is defined as H′(e ^j 〓), and this is called the impulse response.
When expressed as a Freeier series using h'(n), it becomes as follows.

H′(e ^j 〓)= _∞ 〓 ⁿ⁼⁰ h′(n)e ^-j 〓 ⁿ …(3) Equations (2) to (3) above can be rewritten as follows, and H′ (e ^j 〓)= _∞ 〓 ⁿ⁼⁰ h(n/2)e ^-j 〓 ⁿ …(4) If n/2=n′, H′(e ^j 〓)= _∞ 〓 ^no h(n′ )e ^-j 〓 ²ⁿ ′ =H(e ^2j 〓) …(5). In other words, the frequency response H(e ^j 〓) of a digital filter with a unit delay and the frequency response H′(e ^j 〓) of a digital filter with an interstage delay are as follows.
The only difference is the coefficient of the variable (frequency) ω, and the rest are functions of the same characteristics. Therefore, an example of the filter characteristics realized by the digital filter of FIG. 4 which performs a two-sampling time delay is as shown in FIG. 6b. That is, in the frequency response H'(e ^j 〓) shown in FIG. 6b, the same level as the frequency response H(e ^j 〓) shown in FIG. 6a is obtained at half the frequency. As a result, the band in which effective filter characteristics can be obtained is O~ _s /4Hz, which is half of that in FIG. 6a. However, in FIG. 6b, the distribution of peaks and valleys in the filter characteristics is finer than in FIG. 6a, indicating that fine frequency characteristic control is possible.

As described above, according to the digital filter that introduces the interstage delay, it is possible to obtain a filter characteristic in which the filter characteristic at a unit delay is compressed in the frequency direction at a ratio corresponding to the number of delayed sampling times. Therefore, finer frequency characteristic control is possible than in the case of unit delay. If such fine frequency characteristic control is to be achieved using a unit delay type digital filter, an unnecessarily large number of stages must be provided. As mentioned above, this is
This is because it is easier to distribute the peaks and troughs of the filter characteristics in a digital filter almost uniformly over the entire area centered on _s /2, and it is better to achieve fine frequency characteristics in a certain specific band (for example, OHZ to _s /4). In this case, it was necessary to provide a large enough number of stages to realize the characteristic, and to ensure that some characteristic occurred even in unnecessary bands (for example, _s /4 or higher). In this regard, according to the present invention, fine frequency characteristic control is possible with a small number of filter stages, and both manufacturing cost and circuit configuration scale can be saved.

Note that the number of delay sampling times in each stage is not limited to two, but may be three or more. As the number of delay sampling times increases, the filter characteristics are compressed in the frequency direction, making it possible to control the frequency characteristics more precisely. However, the effective frequency band becomes narrower accordingly. For example, if the delay time of the delay circuit 47 in FIG. 4 is three sampling times, the band in which the input musical tone signal can be effectively controlled is 0Hz to _s /6. Therefore, it is assumed that the number of sampling times for the inter-stage delay is determined by the sampling frequency _s and the frequency band of the input musical tone signal. For example, sampling frequency _s
When the frequency band of the input musical tone signal is 50 KHz and the frequency band of the input musical tone signal is about 0 Hz to 10 kHz, it is preferable to set the delay time at each stage (unit) to 2 sampling times as shown in FIG.

Figure 7 shows the basic configuration of the FIR filter, and Figure 8 shows the basic configuration of the FIR filter.
The figure shows the basic configuration of an IIR filter. Such FIR filters, IIR filters, or a combination thereof may be used as the filter units _L1 to _LN . Furthermore, each of the units _L1 to _LN can be constructed using a high-order cyclic digital filter (a type of IIR filter) as shown in FIGS. 9 and 10. In FIGS. 7 to 10, the blocks labeled "delay", such as the blocks labeled with reference numbers 48, 49, 50, and 51, indicate delay circuits, and are labeled with reference numbers 52, 53, 54, and 55. Triangular blocks, such as the blocks shown above, indicate multipliers, and blocks marked with a + sign, such as the blocks marked with reference numbers 56, 57, 58, and 59, indicate adders. Also, multipliers 52, 53, 5
K ₁ , K ₂ , K ₃ ,...K _o , − input to 4, 55...
K′ ₁ , −K′ ₂ ...−K′ _o , K ₀₁ , K ₁₁ , ... are filter coefficients. Even when using digital filters of the type shown in FIGS. 7 to 10,
The present invention can be implemented by setting the delay time of each delay circuit 48, 49, 50, 51, . . . to two sampling times or more.

Since digital multiplication generally involves a calculation time delay, a time delay occurs in the musical tone signal at the multiplier 46 in each of the units _L1 to _LN in FIG. It is possible to delay the signal input timing to the other inputs of the adders 44 and 45 as appropriate in order to accommodate such calculation time delay, and in actual circuits, such design Appropriate delay measures will be provided upon request. In addition, in order to adjust the calculation timing in the subtracter 43 in consideration of such calculation time delay, in the actual circuit, the delay time at the delay circuit 47 is not exactly 2 sampling times, but is equal to the calculation time delay described above. shall be offset. However, when viewed as a whole, it goes without saying that the time delay set between each unit (each stage) is an integer sampling time of two sampling times or more. Of course, such design consideration is also taken when other filter types (for example, those shown in FIGS. 7 to 10) are adopted.

As explained above, according to the present invention, by adding an interstage delay of one sampling time or more to the location where a unit delay is applied,
A filter characteristic obtained by compressing the filter characteristic based on only a unit delay in the frequency direction can be obtained. This allows fine frequency characteristic control, and the same filter characteristics can be realized using fewer filter stages than a filter with only unit delay, which simplifies the circuit configuration.

[Brief explanation of the drawing]

FIG. 1 is a block diagram of the overall configuration of an electronic musical instrument showing an embodiment of the present invention; FIG. 2 is a block diagram showing an example of the internal configuration of the digital filter section of FIG. 1;
FIG. 3a is a block diagram showing the basic format of the Lattice digital filter, FIG. 3b and c are block diagrams showing equivalent conversion examples of the basic format of the Lattice digital filter, and FIG. FIG. 5a is a block diagram showing an example of an impulse response in a Latisse-type digital filter with only unit delay, and FIG. A diagram showing an example of an impulse response in a filter, that is, a Lattice-type digital filter that introduces an interstage delay; FIG. 5b is a diagram showing an example of the frequency response characteristic corresponding to the impulse response shown in FIG. is a block diagram showing the basic type of an infinite impulse response filter that can also be applied to the filter unit shown in FIG.
FIGS. 9 and 10 are block diagrams respectively showing examples of high-order recursive filters that can be applied to the filter unit of FIG. 2. 11... musical tone signal generation section, 12... timbre selection device,
13... musical tone signal distribution and addition control circuit, 14...
Digital filter section, L ₁ to L _N ... Filter unit, 37 to 42, 47 to 51... Delay circuit, 43
...Subtractor, 44, 45, 56-59, 23-29
... Adder, 30-36, 46, 52-55... Multiplier.

Claims (1)

[Scope of Claims] 1. A musical tone signal supply means for supplying a digital musical tone signal at a predetermined sampling period; and a musical tone signal supplying means for inputting the digital musical tone signal supplied from the musical tone signal supplying means,
a digital filter circuit that performs the following predetermined filter calculation, applies a delay of at least two sampling times or more between each calculation stage in synchronization with the sampling period, and generates an output digital musical tone signal synchronized with the sampling period; A digital filter device for an electronic musical instrument, comprising: filter coefficient supply means for supplying n-th order filter coefficients to each of the arithmetic stages according to a timbre to be imparted to the digital musical instrument. 2. The digital filter circuit is provided in association with a calculation means for performing a predetermined filter calculation in response to the digital musical tone signal input from the musical tone signal supply means, and is provided in association with the calculation means, and A delay of at least two sampling times or more is applied to the digital musical tone signal obtained by the calculation in the calculation means or the digital musical tone signal obtained by the calculation in the calculation means, and the delayed signal is applied to the digital musical tone signal obtained by the calculation in the calculation means. 2. A digital filter device for an electronic musical instrument according to claim 1, further comprising a delay means for use in the digital filter device. 3. The digital filter device for an electronic musical instrument according to claim 2, wherein the predetermined filter calculation type in the digital filter circuit is a calculation type of a latex type digital filter. 4. The arithmetic means includes a plurality of cascade-connected arithmetic units, and the delay means applies a predetermined delay of two sampling times or more to a signal fed back from each arithmetic unit to each preceding arithmetic unit. a subtracter that includes a plurality of delay circuits to be set, each of the arithmetic units receives a musical tone signal from a preceding arithmetic unit, and subtracts this input musical tone signal from the feedback signal that has passed through the delay circuit; a multiplier that multiplies the output signal of this subtracter by a filter coefficient; a first adder that adds the output signal of this multiplier and the input musical tone signal and provides the added output to the next stage arithmetic unit; , adding the output signal of the multiplier and the feedback signal via the delay circuit;
4. A digital filter device for an electronic musical instrument according to claim 3, further comprising a second adder that feeds the addition output back to a preceding arithmetic unit. 5. The digital filter device for an electronic musical instrument according to claim 2, wherein the predetermined filter calculation format in the digital filter circuit is a finite impulse response filter calculation format. 6. The digital filter device for an electronic musical instrument according to claim 2, wherein the predetermined filter calculation format in the digital filter circuit is an infinite impulse response filter calculation format.