JPS6149516A - Digital filter device for music signal - Google Patents

Abstract

PURPOSE:To control frequency characteristics effectively without generating any return noise by switching the delay length of a signal delay circuit of each stage selectively within a range of one - plural sampling cycles while one sampling cycle is determined as a minimum unit. CONSTITUTION:A music signal generated with a key of the keyboard part of an electronic musical instrument is timbre-selected and inputted to a digital filter part having the digital filter circuit DFC to perform timbre control. The circuit DFC consists of cascaded digital filter units L1-LN, whose signal delay lengths are changed selectively within the range of one - plural sampling cycles while one sampling cycle is regarded as a minimum unit; and a control signal DLY for selecting a delay length is supplied to the units L1-LN through a delay length control circuit. Thus, frequency characteristics are controlled effectively without generating any return noise.

Description

DETAILED DESCRIPTION OF THE INVENTION [Field of Industrial Application] The present invention relates to a digital filter device for musical tone signals used in an electronic musical instrument and a musical tone signal processing device.

[Conventional technology]

Generally, a digital filter consists of a plurality of stages of filter units, and each stage is set with a unit delay corresponding to one sampling time of the input musical tone signal. The filter characteristic of such a unit delay type digital filter is .fs as shown in FIG. 4(a), and exhibits a symmetrical characteristic around the position of half the sampling frequency f. The number of peaks and valleys in this filter characteristic is determined according to the number of stages of the filter unit, and in this case, the distribution of peaks and valleys is determined from the viewpoint of the accuracy of filter calculation and the number of data bits. It is easier to set the peaks and valleys to spread more or less uniformly, but if you try to concentrate the distribution of peaks and valleys only in a predetermined band, it is necessary to increase the filter calculation accuracy and increase the number of data points. Furthermore, special measures were required to set the coefficients. In general, filter operation is performed using one of the symmetrical characteristics (the characteristic in the band from OH to -H2), and the desired characteristic in this band (0 to 'H2') is used. The filter coefficients, the number of stages, and the connection type between each unit are set so as to obtain the following.

In this way, fS of a unit delay type digital filter is suitable for setting the characteristics so that the peaks and troughs of the filter characteristics are distributed almost uniformly over the whole area centered on -. In order to achieve such complex characteristics (complicated peak-trough characteristics in a biased frequency band), troublesome processing was required. In other words, a certain band (
For example, when trying to achieve fine filter characteristics in the range 0 to 8H2), it is necessary to provide as many stages as possible to achieve the characteristics, and to ensure that some characteristics occur even in unnecessary bands (for example, 5 or more). It was. For example, when trying to realize the formant characteristics of a human voice using a unit delay lattice digital filter, f8
It has been found that as many as 21 filter stages are required when is approximately 50KH2.

In order to eliminate the drawbacks of the unit delay type digital filter as described above, Japanese Patent Application Laid-Open No. 177026/1983 (herein, praise of the unit delay)? Kosara? It has been proposed to set an interstage delay greater than or equal to l sampling time. For example, when the total delay time per stage is 2 sampling times, the filter characteristics of a digital filter that introduces such an interstage delay are as shown in Figure 4 (b), and the effective filter time delay is The characteristic of the case shown in FIG. 4(a) is compressed by half in the frequency direction. Therefore, it is possible to control frequency characteristics more precisely than in the case of unit delay.
Fine frequency characteristic control can be performed without increasing the number of filter stages. For example, if a 2-sampling link time delay is set for each stage, human voice formant characteristics can be realized with about half the number of stages compared to a unit delay type.

[Problem that the invention seeks to solve]

Incidentally, in the interstage delay type digital filter as described above, the effective band is compressed to the low frequency side, so that aliasing noise is likely to occur in a relatively low band. for example,
If you set an interstage delay of 2 sampling times as shown in Figure 4 (1)), there will be no problem as long as it does not have frequency components in the upper band, but depending on the tone, for example, As shown, it has a frequency component that is extended in the aliasing noise band. If a musical tone signal with frequency components extending into the aliasing noise band is input to an interstage delay type digital filter with a fixed interstage delay time, aliasing noise will occur in the frequency components above -island, for example. The problem arises that it is included. Also,
If frequency components above fS are cut in order to remove aliasing noise, a problem arises in that the original tone color is lost.

Furthermore, even if an attempt is made to achieve a timbre with frequency characteristics as shown in Figure 4 (C), aliasing noise will occur in a frequency band of 1 C or higher in an interstage delay type digital filter. It was difficult to realize this.

This invention has been made in view of the above-mentioned points, and it is possible to efficiently improve the frequency characteristics of musical tone signals (audio signals) having various frequency components without generating aliasing noise or damaging the frequency components. The present invention aims to provide a digital filter device that can be controlled. Another object of the present invention is to provide a digital filter device that can easily obtain various variations of filter characteristics that could not be obtained with a single digital filter with a simple configuration.

[Means for solving problems]

In the signal delay circuit of each stage in the digital filter circuit, the delay length is not fixed, but the delay length is selectively set in the range of 1 to a predetermined plurality of sampling link periods, with one sampling period as the minimum unit. It is characterized by being able to switch.

[Effect]

Since the delay length in the signal delay circuit for each stage can be switched, it is possible to use unit delay type digital filters or interstage delay type digital filters with arbitrary delay lengths without changing the hardware configuration of the digital filter circuit. , you can freely switch to any type of digital filter. Therefore, all of the above-mentioned problems can be solved by switching the delay length depending on the desired filter characteristics to be achieved or the frequency characteristics of the input musical tone signal.

〔Example〕

Hereinafter, an embodiment of the present invention will be described in detail with reference to the accompanying drawings.

1 to 3 show an embodiment in which a digital filter device according to the present invention is applied to an electronic musical instrument, and the features of the present invention are best shown in FIG. 1.

First, the overall structure of the electronic musical instrument will be briefly explained with reference to FIG. 2. The keyboard section 1o includes, for example, an upper keyboard, a lower keyboard, and a pedal 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,
The musical tone signal of each series is given a predetermined tone by the musical tone signal generating section 11 in accordance with the tone selection in the tone selection device 12, but depending on the series, the tone imparting may not be completed, and these are transferred to the digital signal at the subsequent stage. The filter section 14 performs timbre control. 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-type tones, for example, positive low-frequency characteristics or complex characteristics of IJ songs, it is preferable to perform spectral correction by further applying fixed formant-type 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 given to a musical tone signal distribution and addition control circuit 13. This control circuit 16 is provided. The control circuit 13 controls the timbre selection device 12 to select a predetermined output from among the outputs of the timbre selection device 12 according to timbre selection information given from the timbre selection device 12, which is capable of adding musical tone signals among each series, and a digital filter. The musical tone signals that can be added are added (mixed) and output to the line 15, and then the digital filter section 1
4, the digital musical tone signals for each series are time-division multiplexed between each series and sent to a common signal line 16.
Output to. Note that the control circuit 1-3 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 of the line 15 that has not been filter-controlled with the musical tone signal that has been filter-controlled by the digital filter section 14. Because it is time-division multiplexed, are these musical tone signals separated for each series? After this demultiplexing, the above mixing is performed.

Note that the "allocation" and "addition" operations in the control circuit 16 and the "demultiplexing" operations in the mixing circuit 17
Since the operation can be easily implemented using known digital technology, 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 (timbre selection information TC) among the outputs of the timbre selection device 12 is given to the digital filter section 14, and the filter characteristics for each series are set respectively according to 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 predetermined filter coefficients are read from this internal ROM in accordance with the tone selection information TC. The filter section 14 is adapted to be used.

The control circuit 13 generates a synchronization pulse 5 in response to the reference timing for time-division transmission of musical tone signals to the line "16".
It is designed to output YNC.

This synchronization pulse 5YNC is transmitted to the digital filter section 14.
and is used to control the filter calculation operation in synchronization with the digital musical tone signal provided via line 16. One period of this synchronization pulse 5YNC 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 unit 14 includes N stages (N is an arbitrary natural number) of cascade-connected unitary filter units L1 to L.
The circuit includes a filter coefficient supply circuit 20 that supplies filter coefficients to this circuit DFC, and a timing signal generation circuit 21 that supplies a timing signal for calculating ζζ calculation to this circuit DFC.

The digital musical tone signal applied from the control circuit 13 in FIG. 2 via the line 16 is input to the first stage filter unit L1, passes through each unit L1 to LN sequentially, and is subjected to filter control, and then is filtered to the final stage (N It is output from the unit LN of the second stage). The timing signal generation circuit 21 generates various timing signals for controlling time-division filter calculation operations in each of the filter units L1 to LN based on the synchronization pulse 5YNC, and transmits these signals to each unit L-L to LN. supply to. The filter coefficient supply circuit 20 includes the above-mentioned filter coefficient internal ROM 22, and stores predetermined filter coefficients in the ROM 22 according to the timbre selection information TC given from the timbre selection device 12 (FIG. 2).
Read from. (2) A set of filter coefficients corresponding to a tone is composed of N filter coefficients corresponding to each unit L1 to LN.

Such a set of filter coefficients is read out corresponding to each series according to the timbre selected in each series.

The filter coefficient supply circuit 20 receives the synchronization pulse 5yNC, and synchronizes with the time-division calculation timing of the digital musical tone signal of each series in each filter unit L to LN, and calculates the above-mentioned signal corresponding to the tone selected in each series. Filter coefficients are supplied to the individual units Ll to LN, respectively.

The signal delay length at each stage (each unit) L+ to LN of the digital filter circuit DFC can be selectively changed in the range from 1 to a predetermined plurality of sampling periods, with one sampling period as the minimum unit. It has become. A control signal DLY for selecting this delay length is generated from the delay length control circuit 23, and each filter unit Ti1 to L
given to N. The delay length control circuit 23 is configured to generate a control signal DLY for selecting a delay length according to the timbre selected in each series according to the timbre selection information TC, and has, for example, a ROM. Note that this control circuit 26 also
Similar to the circuit 20, receiving the synchronization pulse 5YNC,
A control signal DLY for each series is generated in synchronization with the time division timing of each series of digital musical tone signals.

Any filter calculation format may be used in the digital filter circuit DFC. Basic types of digital filters include lattice filters, finite impulse response filters (hereinafter referred to as FIR filters), and infinite impulse response filters (hereinafter referred to as fIR filters), among others, lattice filters are suitable for speech synthesis. It is known to be a filter. Furthermore, 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 allows for desired filter characteristics. It has the advantage that the method for setting coefficients has been established. Therefore, as a preferable example, FIG. 1 shows details of a digital filter circuit DFC constructed using the calculation format of a lattice filter.

In the first stage filter unit L1, reference numeral 24 is a subtracter, 25 and 26 are adders, 27 is a multiplier, and 28 is a delay circuit. FS-IN is musical tone signal input terminal, F'5
-OUT is a musical tone signal output terminal, BS-IN is a feed bank signal input terminal, and BS-O'UT is a feedback signal output terminal.

Other units L except for the final stage filter Uniso 1-LN
2 to LN-1 have the same configuration as the unit L1, and the musical tone signal output terminals FS-O of each 22) Lt to LN,
UT is connected to the musical tone signal input terminals F and S-IN of the units L2 to LN in the next stage, and the feedback signal output terminal B5-0UT of each unit L2 to LN receives the feedback signal of the units L1 to LN in the previous stage. Connected to input terminal BS-IN. In the final stage filter unit LN, its own output signal is fed-banked.

The minus input (-
), a digital musical tone signal on line 16 (FIGS. 2 and 3) is inputted through an input terminal FS-IN. The plus input (+) of this subtracter 24 is connected to the next stage unit L2.
A feedbanked signal is applied via the delay circuit 28. The multiplier 27 multiplies the filter coefficient of the output signal of the subtracter 24 by 1. Filter coefficient Kl
is supplied from the filter coefficient supply circuit 20 (FIG. 3), and the other units Lz to LN are also provided with filter coefficients 2 to KN corresponding to each of them. The output of the multiplier 27 is given to the adder 25, and the input terminal FS-I
It is added to the input musical tone signal given from N. The output of this adder 25 is given to the next stage filter unit via the output terminal FS-OUT. Further, the output of the multiplier 27 is given to an adder 26 and added to the signal given via a delay circuit 28. The output of this adder 26 is fed back to the preceding stage filter unit via output terminals BS and -0UT (however, since the first stage unit LI does not have a preceding stage, it is not actually given anywhere).

Delay circuit 28.282-28N for each unit L1-LN
The delay length can be selectively switched in the range of 1 to a predetermined plurality of sampling periods in units of /J% per sampling period. Although a detailed example is shown only for the first stage 22l-t delay circuit 28,
Delay circuits 282-28N of other units L2-LN also have the same configuration.

In this example, the delay circuit 28 outputs one of the digital musical tone signals.
A delay length corresponding to either the sampling period or two sampling periods can be selected.

That is, two delay elements 29 and 30 that set a delay for one sampling period are connected in series ("1D" in the block means a delay for one sampling period).
, the signal outputted from the delay element 60 and set with a delay of two sampling periods is applied to the A input of the selector 61, and the signal outputted from the delay element 29 and set with a delay of one sampling period is applied to the selector 61. Add to B input. The aforementioned delay length control signal DLY is applied to the selection control input of the selector 31 from the delay length control circuit 26 (FIG. 3), and when the value of the signal DLY is 'J', the A input is selected and ' Select B input after '0pa. When the A input of the selector 31 is selected, the delay length of the delay circuit 28 is 2.
This is the sampling period, and when the B input is selected, the delay length is one sampling period.

As already explained at the beginning, the signal delay length per stage is 1
Effective filter characteristics can be obtained in a sampling period (unit delay only), and effective filter characteristics can be obtained in a band where the signal delay length per stage is two sampling periods (including inter-stage delay). In addition, in the case of a two sampling period delay, J
A Quilter characteristic is obtained in which the filter characteristic in the case of sampling period delay is compressed by half in the frequency direction, and fine frequency characteristic control is possible without particularly increasing the filter coefficient Kl-KN and the number of stages N. Therefore, in this digital filter circuit DFC, by selectively controlling the delay length using the control signal DLY, either unit delay type or interstage delay type filter characteristics can be immediately realized.

Note that the delay lengths that can be selected in the delay circuits 28, 282 to 28N are not limited to 1 sampling period and 2 sampling periods as shown in the embodiments, but also other delay lengths (for example, 1 sampling period and 3 sampling periods, or 2 sampling periods).
sampling period, 3 sampling periods, etc.). Also, the number of selectable delay lengths is not limited to 2, but 3.
or more (for example, 1.2.3 sampling periods, or 1
2.3.4 sampling period, etc.).

As is clear, as the delay length increases, the filter characteristics are more 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 a delay length of 3 sampling periods is selected, the delay length is selected and controlled according to the timbre selection information TC (that is, according to the type of timbre to be realized by the filter). However, it is not limited to this. For example, as shown by the broken line on the input side of the delay length control circuit 23 in FIG. or take without indicating the operation details of the controls that the user can freely operate.
It is also possible to input one or more as input parameters and generate the delay length control signal DLY according to the contents.

The filter calculation format in the digital filter circuit DFC is not limited to the lattice filter, but any format such as the above-mentioned FIR filter, IIR filter, or high-order cyclic filter (a type of IR filter) may be used. In that case, the signal delay length of the delay circuit of each stage may be variably controlled by means similar to the above embodiment.

Furthermore, in general, there is a calculation time delay in disortal multiplication, so a time delay occurs in the musical tone signal at the multiplier 27 in each unit L, -LN in FIG. It is possible to delay the signal input timing to the other inputs of the adders 25 and 26 as appropriate in the design in order to accommodate such a calculation time delay, and in actual circuits, such design Appropriate delay measures will be provided upon request.

In addition, in consideration of such calculation time delay, the subtracter 2
In order to match the calculation timing in 4, in the actual circuit, the delay time at the delay circuit 28 is not exactly one or two sampling periods, but is set to offset the calculation time delay.

However, when viewed as a whole, it goes without saying that the time delay set between each unit (each stage) is one to a predetermined plurality of sampling link periods with the minimum unit being one sampling period. Further, it goes without saying that such design considerations are also taken when other filter calculation formats are adopted.

〔Effect of the invention〕

As described above, according to the present invention, the signal delay length of each stage of the digital filter can be selectively and variably controlled within the range of 1 to a plurality of predetermined sampling periods. It is possible to easily obtain various variations of filter characteristics that could not be achieved with conventional filters, and the configuration is simple. In addition, since the signal delay length can be changed at any time according to the frequency components or frequency bands of the input musical tone signal, it is possible to change the signal delay length at any time depending on the frequency component or frequency band of the input musical tone signal. It is possible to efficiently control frequency characteristics without impairing the frequency characteristics, and the configuration is simple.

[Brief explanation of drawings]

FIG. 1 is a circuit diagram showing one embodiment of the present invention.
2 is a block diagram showing the overall structure of an example of an electronic musical instrument that employs the digital filter device according to the present invention in the digital filter section. FIG. A block diagram showing an example of the internal configuration of the digital filter section in FIG. 2; FIG. 4(a) is a diagram showing an example of filter characteristics realized by a digital filter in which the signal delay length of each stage is one sampling period; FIG. 4(b) is a diagram showing an example of filter characteristics realized by a digital filter in which the signal delay length for each stage is two sampling periods.
C) is a diagram illustrating the distribution characteristics of frequency components in a musical tone signal. DFC digital filter circuit, L, -LN, filter unit for each stage, 24, pull 8! , 25.26・Adder, 27・Multiplier, 28,282°28N・・Delay circuit,
29.30 delay element, 31 selector, 12...timbre selection device, 14...digital filter section, 20.
Filter coefficient supply circuit, 26...delay length control circuit.

Claims (1)

[Claims] 1. A digital filter circuit that includes a plurality of stages of signal delay circuits and arithmetic circuits connected according to a predetermined filter calculation format, and filters a digital musical tone signal input at a predetermined sampling period; In the digital filter device for musical tone signals, the signal delay circuit of each stage in the digital filter circuit has a delay time of 1 to 1 within a predetermined plurality of sampling periods.
delay length control means capable of selectively switching the delay length using a sampling period as the minimum unit; A digital filter device for musical tone signals, characterized in that: 2. The delay length control means generates the control signal for selecting a delay length corresponding to the selected timbre in response to timbre selection information for selecting filter characteristics of the digital filter circuit. A digital filter device for musical tone signals according to claim 1. 3. The digital filter device for musical tone signals according to claim 1, wherein the predetermined filter calculation format in the digital filter circuit is a calculation format of a lattice type digital filter.