JPS61272795A - Musican sound signal processor - Google Patents

Abstract

(57) [Summary] This bulletin contains application data before electronic filing, so abstract data is not recorded.

Description

[Detailed Description of the Invention] [Industrial Application Field] The present invention relates to a device for processing audio frequency signals such as musical tone signals or voice signals, for example, a musical tone signal processing device in an electronic musical instrument, etc. The present invention relates to an improvement in a part related to an accumulator that accumulates audio frequency signals of a plurality of given sample points, and more specifically, to making it possible to easily realize a digital filter function.

[Conventional technology]

In electronic musical instruments, in order to accumulate musical tone signals from multiple sample points, an accumulator 10 having the configuration shown in FIG. has been done.

This accumulator 10 includes an adder 11 that inputs musical tone signals of a plurality of sample points given in a time-division manner to its A input, and a delay circuit (i.e., a clock pulse φ a temporary storage circuit (temporary storage circuit) 12 synchronized with
3, and a latch circuit 14 for latching the accumulation result output from the adder 11 at the end of a predetermined accumulation time interval. FIG. 3 is a diagram showing an example of the time relationship of each signal in FIG. 2, in which the number of time-division time slots of the musical tone signal given to the A input of the adder 11 is N in one cumulative time interval; allocated to each of these time slots.
.

The assigned N musical tone signal sample values are accumulated in the accumulation time interval. The clear signal πCLR becomes "O" in the first time division time slot 1 in each cumulative time interval, and becomes a1 in other time slots 2 to N.
．． This is the signal that becomes Jl.

The load signal LD is a signal that becomes '1' in the latter half of the last time-division time slot l-N in each cumulative time interval.

Clear signal ACCLR is applied to the control input of gate 13, and load signal LD is applied to the latch control input of latch circuit 14. The addition output of the adder 11 is delayed by one time slot in the delay circuit 12 and is applied to the B input of the adder 11 via the gate 13. Gate 13 receives signal ACC
Due to LR, time slots 2 to N are also released. Therefore, the addition result in a given time slot is applied to the B input of adder 11 in the next time slot and added to the sample value of that time slot applied to the A input. In this way, the sample values of time slots 1 to N are added (accumulated) one after another, and finally, in time slot I-N, the cumulative results of the sample values of all time slots 1 to N in one accumulation time interval are added. output from the device 11. Since the load signal LD becomes °'1'' in the latter half of this time slot l-N, all of the time slots 1 to 1
The cumulative result of N musical tone signal sample values is latched in the latch circuit 14 in response to the load signal LD. On the other hand, this accumulation result is delayed by one time slot in the delay circuit 12, and
It is applied to the gate 13 in the next time slot, that is, the first time division time slot 1 in the next cumulative time interval. However, when starting a new accumulation, the previous accumulation result is unnecessary, so the clear signal ACCLR is set to 0"
This closes the gate 16, thereby clearing the previous accumulation result and providing only the sample value data of time slot 1 to the adder 11.

[Problem that the invention seeks to solve]

Conventional accumulators configured as described above are used only for the purpose of accumulating multiple sample values, and when adding a signal processing function for other purposes, such as a filter function, the purpose function itself is A completely new predetermined functional circuit, such as a filter circuit, must be added. Therefore, there were problems in that the circuit scale increased and the cost also increased. Also, if the signal processing function you are trying to add is not that important, in other words, it is better to have it, but you can do without it, for example, if the desired sound quality is generally achieved. When a user wants to further improve the sound quality by further correcting the sound quality, it is uneconomical to add a relatively expensive digital filter or the like for simple sound quality correction.

This invention was made in view of the above points, and it is possible to add a filter function by simply adding a simple circuit to an existing accumulator for accumulating musical tone signals from multiple sample points. The present invention aims to provide a musical tone signal processing device that achieves the following.

[Means for solving problems]

The musical tone signal processing device according to the present invention feeds back the result obtained by multiplying the accumulation result in the previous accumulation time interval by a predetermined coefficient to the input side of the accumulator for each accumulation time interval of the accumulator. It is characterized by being equipped with a feedback means.

The basic configuration of the present invention is shown in FIG. 1. Parts substantially equivalent to the conventional accumulator 10 shown in FIG. Circuit 120, latch circuit 14
0 is also similar to the above-mentioned 11.12°14. The output of the delay circuit 120 is applied to the A input of the selector 15. This selector 15 selects the A input when the clear signal ACCLR applied to the selection control manual SA is "1", and applies the selected output to the B input of the adder 110.

Therefore, the portion passing through the A input of the selector 15 corresponds to the gate 13 described above. The output of the delay circuit 120 is also applied to the multiplier 16, where it is multiplied by a predetermined coefficient g. The output of the multiplier 16 is given to the B input of the selector 15.

Selector 15 is B when clear signal ACCL is @o#
Select the input and add the selected output to the B input of adder 110. The signal output from the delay circuit 120 when the clear signal ACCLR is 60'' is the cumulative result of the sample values of all time slots 1 to N in the previous cumulative time interval, and is multiplied by the coefficient g. It will be fed back to the B input of the adder 110. At this time, the musical tone signal sample value data of the first time division time slot 1 in the current cumulative time interval is given to the eight inputs of the adder 110. This and the previous accumulation result multiplied by the coefficient g are added.Thus, the portion passing through the multiplier 16 and the B input of the selector 15 corresponds to the feedback means 17 described above.

[Effect]

In the present invention, in the conventional accumulator 10 (FIG. 2), the gate 13 is closed in the first time division time slot 1 to clear the previous accumulation result, and the adder 11 does not actually perform an addition operation. Considering that the sample value data of time slot 1 is simply passed through, that is, the calculation timing of time slot 1 is wasted, this empty time slot is used for another purpose, that is, a filter. It is characterized in that it is used for calculations and realizes a digital filter function with an extremely simple and economical circuit configuration.

That is, if the basic configuration of FIG. 1 is shown by an equivalent circuit, it becomes as shown in FIG. 4, where the first additive amateur D1 performs a filter function in the musical tone signal sample D and the multiplication element G of each time slot 1 to N. (In particular, it is designed to function as an engineering IR filter: infinite impulse response filter). In this equivalent circuit, the first addition element AD1 performs accumulation, the result is input to the second addition element AD2, and then the delay element is used to output the second addition element AD2, that is, the sum which is the accumulation result. A time delay corresponding to one sampling time (not one time slot width but one accumulation interval width) of musical tone signal sample value data is set, and this is multiplied by a coefficient g by a multiplication element G.
and feeds back its output to the second addition element AD2, thus combining the current accumulation result given from the first addition element AD1 and the delayed (that is, temporarily stored) previous accumulation result. The product multiplied by a coefficient g is added. The accumulator function part in this equivalent circuit is an equivalent circuit of the accumulator function realized by a circuit consisting of an adder 110, a delay circuit 120, and a part passing through the A input of the selector 15 in time division time slots 2 to N. The filter function part is
This is an equivalent circuit of the function realized by the circuit consisting of the adder 110, the delay circuit 120, the multiplier 16, the part passing through the B input of the selector 15, etc. in the time division time slot 1. The reason why the delay time of each delay element corresponds to one accumulation time interval is that this filter function is executed once per accumulation time interval.

As is clear from the equivalent circuit shown in FIG. 4, the filter function part constitutes a first-order IIR filter whose delay unit is one sampling time.

Therefore, this part can realize the function of an IIR filter. The characteristics of this filter can be controlled by variably controlling the value of the coefficient g or by appropriately changing the delay unit time (that is, changing the order of the filter).

〔Example〕

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

In the embodiment shown in FIG. 5, an accumulator 100 forms a part of an arithmetic circuit element of an FIR filter (finite impulse response filter), and the FIR filter multi-main arithmetic circuit 18 provided in the preceding stage and this accumulator 100 An N-order FIR filter is configured by the combination of . This FIR filter main operation circuit 18 inputs a digital musical tone signal MW, and generates a plurality of sample values X1 to X, l (here, 1 to
N corresponds to the order of the delay stage) in a time-division manner according to the clock pulse φ, and each time-division sample value X1 to XM is multiplied by a predetermined filter coefficient h□ to h, The multiplication result h1.X.

~hM-xN are output in a time-division manner in correspondence with time-division time slots 1 to N. Accumulator 100
Now, as mentioned above, the sample values of each time slot 1 to N, that is, the sample values h1·X multiplied by the filter coefficients
, ~hM-XM are accumulated. As a result, the accumulator 100 obtains one digital musical tone signal sample value Y, which can be expressed by the following equation: Y=Σ h··X·−@(1) weight=1. Here, the sample point number of the musical sound waveform is indicated using the suffix n,
Rewriting the above (υ equation) for the sample value Y at an arbitrary sample point number n, Yn=, Σ hi@Xn
-ia ae (2)! It can be expressed as =1. As is clear from this, the FIR filter main arithmetic circuit 18 and the accumulator 100 constitute an N-order FIR filter.

The shift and inversion circuit 16a provided between the delay circuit 120 and the B input of the selector 15 corresponds to the multiplier 16 in FIG.
, is to be multiplied by . Therefore, the shift and inversion circuit 16a converts the output data of the delay circuit 120 to a negative value by shifting it one bit lower and inverting the value, which is equivalent to multiplying by a coefficient "-d". I'm trying to get results. This simplifies the configuration of the multiplication means.

In this way, the shift and inversion circuit 16a and the selector 15
A feedback means 17 consisting of a portion passing through the B input of
By adding , the characteristics of the IIR filter become as shown in FIG. f is a sampling frequency, where the sampling frequency is a frequency in which one cumulative time interval is one sampling period as described above. fl is the Nyquist frequency, which is the diagonal frequency of fS. In this case, a mountain-shaped filter characteristic having a peak at the Nyquist frequency and wave number fM is obtained.

Here, for example, the main arithmetic circuit 18 and the accumulator 100
Assuming that the characteristics of the FIR filter constituted by the above characteristics are as shown by the solid line 19 in FIG. ) are superimposed, and the finally obtained filter characteristics are as shown by the dotted line 21. The FIR filter characteristic shown by the solid line 19 is a low-pass filter having a cutoff frequency near the Nyquist frequency fN, and is for removing aliasing noise due to sampling. However, in the characteristic of solid line 19, the shoulder portion is not very steep, so there is room for improvement. Therefore, when the characteristics of the filter function added by the feedback means 17, which has a peak at the Nyquist frequency fM, are superimposed, the shoulder portion becomes steeper (that is, the Q increases) as shown by the dotted line 21, resulting in suitable characteristics. In this way, the characteristics that were insufficient with the original FIR filter alone can be corrected to suitable ones.Moreover, by simply adding the feedback means 17 without adding a special completed IIR filter circuit. Such correction can be achieved.

Note that the multiplication means of the feedback means 17 is not limited to the shift and inversion means 16a, but may of course be a multiplier that multiplies by an arbitrary coefficient g.

In addition, the accumulation result given to the B input of the selector 15 via the multiplication means is not limited to the one in the immediately preceding accumulation time interval (that is, the one sampling period before), but the number of delay stages can be further increased along that route, and the result can be adjusted accordingly. The data may be from an accumulated time interval that is the number of sampling cycles before.

Further, in the embodiment of FIG. 5, the accumulator 100 is
It forms part of the calculation element of the IR filter, and this FI
Although the filter function according to the present invention is utilized to correct the characteristics of the R filter, it goes without saying that the present invention can be applied to other cases as well. For example, FIG. 8 shows an example, in which the tone generator 22 outputs N
Sample values of one different musical tone signal are generated in a time-divisional manner in time slots 1 to N, and in the accumulator 100, the sample values of these N different musical tone signals are summed and combined into one sample point. Obtain the amplitude signal. In this case as well, the IIR filter function can be realized by adding the feedback means 17 in relation to the accumulator 100, as described above. Therefore, simple tone quality correction of the musical tone signal obtained by the tone generator 22 can be realized without adding a special completed digital filter circuit.

The present invention is applicable not only to musical sound signals such as scale tones or percussion instrument sounds, but also to processing of audio signals and other audio frequency signals.

Next, a specific example in which the present invention is applied to a musical tone signal generating device such as an electronic musical instrument will be described with reference to FIGS. 9 and 10. In this specific example, the accumulator related to the present invention functions as a part of the arithmetic circuit element of the FIR filter as in the embodiment shown in FIG. 5, and the FIR filter characteristics are changed by the IIR filter function added by the present invention. The correction is made as described above (as indicated by the dotted line 21 in FIG. 7).

First, to briefly explain the overall configuration based on FIG. 9, the musical tone signal generating means 26 uses a pitch synchronization method to generate each pitch
A musical tone signal corresponding to a pitch name (tone name) is digitally generated, and the musical tone to be generated is specified by a keyboard (not shown) or other appropriate means. The musical tone signal generating means 26 is capable of outputting a mixture of one or more digital musical tone signals of different pitches (note names).
Therefore, when such a digital musical tone signal is viewed as a whole, the sampling frequency corresponds to the least common multiple of a plurality of sampling frequencies pitch synchronized with each note name, and is quite high speed (for example, about 800 kHz). Such a pitch-synchronized musical tone signal generating means 26 is disclosed in, for example, Japanese Patent Application Laid-Open No. 57-171395 or Japanese Patent Application No. 59-26.
A device such as that shown in No. 67 can be used. The digital musical tone signal outputted from the musical tone signal generating means 23 is sent to the sound system 25 via the D/A converter 24.
and is pronounced.

On the other hand, the digital musical tone signal output from the musical tone signal generating means 23 is also supplied to a series of digital effect applying devices 26. The digital effect imparting device 26 applies vibrato, chorus, unsample, etc. to the digital musical tone signal.
This is a digital circuit that selectively applies effects such as reverberation, and the sampling frequency of the digital musical sound signal to be input is relatively low (for example, about 50 kHz).
It is aimed at those of. As such a digital effect imparting device 26, it is possible to use, for example, the device shown in Japanese Patent Application Laid-Open No. 58-50595, or any other appropriate device. The digital musical tone signal output from the digital effect imparting device 26 reaches a sound system 38 via a D/A converter 37.

The sampling frequency of the digital musical tone signal output from the musical tone signal generating means 26 is set to a high speed (for example, 800 kHz).
A resampling device 27 is provided between the musical tone signal generating means 23 and the digital effect applying device 26, and the resampling device 27 performs sampling according to the low speed sampling frequency. The reconstructed digital musical tone signal is input to the digital effect imparting device 26.

A digital filter 28 is provided between the musical tone signal generating means 26 and the resampling device 27. This digital filter 28 has a characteristic that can almost eliminate aliasing noise related to a low-speed sampling frequency (for example, 50 kHz), and since aliasing noise occurs in a frequency band of a digital musical tone signal that follows a high-speed sampling frequency, this aliasing noise can be eliminated. In order to remove noise, it is preferable to set the filter characteristics of the digital filter 28 to a low-pass filter whose cutoff frequency is equal to the low-speed sampling frequency (for example, 25 kHz).

Below, the high-speed sampling frequency is 800 kHz.

A more specific example will be described with a low-speed sampling frequency of 50 kHz.

The sample value of the digital musical tone signal with a high-speed sampling frequency given from the musical tone signal generating means 26 is indicated by xn. The suffix n is a sample point number within one cycle of the musical tone signal, and is assumed to be one of 0 to 63, for example.

The sample value of the digital musical tone signal output from the digital filter 28 is indicated by yn. Digital filter 2
8 is a 64th-order FI with the following transfer function as an example.
It consists of an R filter. - yn=Σ hi*)(n-1@*a (3) i=Q The sample value of the digital musical tone signal output from the resampling device 27 is indicated by -.Resampling device 27
So, the high-speed sampling rate of 800kHz is changed to 50kHz.
Since the sampling rate is converted to a low sampling rate of z, the filter output signal yn corresponding to the digital musical tone signal xn sampled at high speed is resampled every 16 sample points. Therefore, it can be expressed as zm=y16n. FIG. 10 shows a detailed example of the digital filter 28 and sampling device 27. The digital filter 28 is configured such that one multiplier 29' is used in a time-division manner as a multiplication means for multiplying the filter coefficients hi of each stage (i=Q to 63).

Three 16-stage delay circuits 30, 31, and 32 are connected in cascade, and the delay operation is controlled by a sampling clock pulse φ synchronized with a high-speed sampling frequency of 800 kHz. The digital musical tone signal xn supplied in 16-bit parallel is input to the first stage of the first delay circuit 30, and the sampling clock pulse φ
1, the data is sequentially delayed in synchronization with the high-speed sampling period.

The undelayed digital musical tone signal The output of the delay circuit 31 delayed by a sampling period is inputted, and the output of the delay circuit 32 delayed by 48 sampling periods is inputted to the "0" input. A selection signal SEL is applied to the selection control input of the selector 33.

This selection signal SEL changes in four states from "0" to "3" during one period of high-speed sampling, and from "0" to "3".
"3" Sequentially select input digital musical tone signal sample values. The state of this selection signal SEL is a clock pulse φ having a frequency of 3.2 MHz, which is four times the high-speed sampling frequency. changes according to

In this way, the selector 36 receives the clock pulse φ.

Sample values xn are selected intermittently at every 16 sample points according to the period of , and are input to the multiplier 29 .

The other input of the multiplier 29 is given the filter coefficient read out from the coefficient ROM 34. Coefficient reading circuit 6
5 is a clock pulse φ. , and specifies the rank i of the coefficient hi to be read out for each cycle of the clock pulse φ0. The coefficient ROM 34 reads out the coefficient hi of the rank i designated by the coefficient reading circuit 35.

Thus, in the multiplier 29, each term hi'' of the equation (3)
xn-1 is the clock pulse φ. It is sequentially filled with water every cycle.

The accumulator 100' uses the value of each term hi-xn-1 given from the multiplier 29 as a clock pulse φ. Accordingly, the accumulator and the sum yn of the series shown in equation (3) above are determined one after another. Since i=Q~63, the clock pulse φ. By performing accumulation for 64 cycles, the sum yn in equation (3) can be found.

The construction of the accumulator 100' and the associated feedback means 17 is similar to that shown in FIG. 1 or FIG. 5. However, in this example, the first
The accumulator 100' does not have a latch circuit 140 as in the accumulator 100 of FIG. 5, but the latch circuit 36 of the resampling device 27 plays this role.

Therefore, as described above, the IIR filter function is added by the feedback means 17, and the original FIR filter characteristic of the digital filter 28 is changed from this IIR filter function.
For example, the seventh factor will be corrected depending on the IR filter characteristics.

Note that the clock pulse φ in FIG. 10 corresponds to the clock pulse φ of the fifth factor in FIG. The clear signal ACCLR is this clock pulse φ.

becomes O# every 64 cycles.

Output beam sampling device 2 of accumulator 100/
7 is input to the latch circuit 36.

A load signal LD is applied to the latch control input of this latch circuit 36, similar to the latch circuit 140 of FIGS. 1 and 5. In this example, the frequency of the load signal LD and clear signal ACCLR is a low-speed sampling frequency of 50kHz.
It is 2.

The latch circuit 36 has a low-speed sampling frequency of 50kH.
With the function of resampling the output musical tone signal of the digital filter 28 according to z, the accumulator 1001
Accumulated value (filter output value of 1 sample point)
It also has the function of latching. As can be seen, in the digital filter 28, the clock pulse φ. 64 of
The filter operation for one sample point is performed using a period of 16 periods of high-speed sampling, ie, one period of low-speed sampling. Therefore, a filter output is not obtained for each sample point of high-speed sampling, but a filter output is obtained intermittently for every 16 sample points. However, this is not an inconvenience. This is because the resampling in the latch circuit 36 can be performed intermittently every 16 sample points of high-speed sampling, and at least the filter output of the necessary sample points can be obtained when performing this resampling. .

In FIG. 9, the digital musical tone signal output from the musical tone signal generating means 23 is input to the D/A converter 24, but instead of this, as shown by the dotted line in FIG. The low-speed converted digital tone signal output from the sampling device 27 may be input to the D/A converter 24. In this way, an inexpensive low-speed operating device can be used as the D/A converter 24.

[Effects of the invention-]

As described above, according to the present invention, it is possible to realize an IIR type digital filter function by simply adding a simple feedback means to an accumulator for accumulating audible signals such as musical tones or audio signals at a plurality of sample points. Therefore, characteristic correction or sound quality correction can be performed easily and economically.

[Brief explanation of the drawing]

FIG. 1 is a block diagram showing an example of the basic configuration of the present invention, FIG. 2 is a block diagram showing the configuration of a conventional accumulator, and FIG. 3 is an example of the generation timing of various signals in FIGS. 1 and 2. The timing chart shown in Fig. 4 is the timing chart shown in Fig. 1.
5 is a block diagram showing an embodiment of the present invention; FIG. 6 is a diagram showing an example of the characteristics of the IIR filter realized by the feedback means of the embodiment; FIG. 7 is a diagram showing the characteristics of only the FIR filter and the characteristics corrected by the IIR filter characteristics in the same embodiment, FIG. 8 is a block diagram showing another embodiment of the present invention, and FIG. 9 is a diagram showing the characteristics corrected by the IIR filter characteristics. FIG. 10 is a block diagram showing a specific example of an electronic musical instrument to which the invention is applied. FIG. 10 is a block diagram showing a specific example of the digital filter sampling device shown in FIG. 100.100'...Accumulator, 110...
Adder, 120...Delay circuit (temporary memory circuit) Q15
... Selector, 16... Multiplier, 16a... Shift and inversion circuit, 17... Feedback means, 18
...FIR filter main calculation circuit, 22...Tone generator, 28...Digital filter.

Claims (1)

[Claims] 1. An accumulator that receives an audio frequency signal such as a musical tone or voice signal at a plurality of sample points given in a time-sharing manner and accumulates the input signal in a predetermined time interval; A musical tone signal processing device comprising feedback means for feeding back, for each accumulation time interval, the result obtained by multiplying the accumulation result in the previous accumulation time interval by an arbitrary coefficient to the input side of the accumulator. 2. The accumulator has a first input for inputting the input signal for each time-division time slot and a second input for inputting the accumulation result fed back for each time-division time slot; and an adder for adding the signals of the second input, temporary storage means for temporarily storing the output signal of the adder and feeding it back to the second input, and a first time division time slot of each accumulation time interval. and means for inhibiting feedback of the output of the temporary storage means to the second input, and the feedback means includes a multiplication means for multiplying the output signal of the temporary storage means by the coefficient, and a multiplication means for multiplying the output signal of the temporary storage means by the coefficient; 2. The musical tone signal processing apparatus according to claim 1, further comprising means for inputting the obtained signal to the second input of the adder in the first time division time slot of each cumulative time interval. 3. The accumulator is a part of an arithmetic circuit element of a digital filter, and the audio frequency signals at the plurality of sample points are provided from other arithmetic circuit elements of the digital filter. The musical tone signal processing device according to item 1. 4. The accumulator is for summing sample point amplitude signals of a plurality of musical tones or audio signals generated in a time-divisional manner in a plurality of channels to obtain a sample point amplitude signal of a musical tone or audio signal that is a composite of them. A musical tone signal processing device according to claim 1.