JPH05346783A - Scale detecting device - Google Patents

Abstract

PURPOSE:To provide the scale detecting device which can detects a given acoustic signal even when the acoustic signal has an intermediate scale between respective fixed scales. CONSTITUTION:The input acoustic signal is digitally filtered to two mutually adjacent fixed scales which are close to the actual scale of the input acoustic signal; and the scale which has a large envelope value between the two scales is denoted as Max and the scale which has a small envelope value is denoted as Second. The low scale between those Max and Second is denoted as Low and the high scale is denoted as High (step Q1); and the address of a data table ROM for intermediate-sound detection is set to Low (step Q2). Then the ratio of the envelope of those Low and High is calculated and the offset of the intermediate-sound data is obtained from the value (step Q3). The detected scale is set to Low and bend data is set to a shift quantity from the Low based upon table data (step S4). The scale shifted from the detected scale by this shift quantity is detected as the scale of the actual input acoustic signal.

Description

Detailed Description of the Invention

[0001]

BACKGROUND OF THE INVENTION 1. Field of the Invention The present invention relates to a scale detection device capable of detecting a scale frequency intermediate between fixed scale frequencies based on the spectrum of an input acoustic signal.

[0002]

2. Description of the Related Art There is a method in which an input acoustic signal such as a musical instrument sound or a human voice sound is time-divisionally subjected to digital filtering processing by digital signal processing, and a scale frequency is detected from the output result (for example, Japanese Patent Laid-Open No. Hei 4). -19696).

Basically, this scale detecting device is applied to a digital waveform signal expressing a given acoustic signal,
In order to detect the level of the frequency spectrum relating to the frequency corresponding to each fixed scale in advance, digital filtering processing of different characteristics is sequentially performed in time division, and the result of the digital filtering processing executed by this digital signal processing means is performed. Based on this, the scale frequency included in the given acoustic signal is detected.

Therefore, according to this scale detecting device,
All the signal processing can be digitally processed, the scale frequency of the input sound can be detected in a short time, and the circuit can be operated in a small scale and stably.

[0005]

However, in the conventional scale detecting device, the digital waveform signal representing the given acoustic signal is sequentially subjected to the digital filtering process of different characteristics in a time division manner, and the digital waveform signal is sequentially processed in advance. The level of the frequency spectrum related to the frequency corresponding to each fixed scale was detected, and it was detected whether the scale frequency of the input acoustic signal was a fixed scale. At an intermediate scale frequency, one of the adjacent fixed scales is detected, so that there is a problem that the scale frequency of the acoustic signal cannot be accurately detected. Therefore, the present invention is
It is an object of the present invention to provide a scale detection device capable of accurately detecting the frequency of a given scale even when the given acoustic signal has a middle scale frequency of each fixed scale.

[0006]

SUMMARY OF THE INVENTION The present invention relates to frequencies corresponding to respective scales by sequentially performing time-divisional digital filtering processing having different characteristics on a digital waveform signal expressing a given acoustic signal. Filter means for detecting the level of the frequency spectrum, selecting means for selecting one of the scale having the highest level of the frequency spectrum detected by the filter means and the scale adjacent to this scale, which has the larger level of the frequency spectrum, and the selection Calculating means for calculating the level ratio of the frequency spectrums of the two scales selected by the means, and the level ratio from the calculating means, which is provided for each scale, is converted into shift data corresponding to the difference between the two scale frequencies. Selected by the selecting means based on the converting means and the shift amount from the converting means Detecting means for detecting an intermediate scale frequency between two scale frequency, it is characterized by comprising a.

[0007]

According to the present invention, the digital waveform signal expressing the given acoustic signal is filtered by the filter means.
The digital filtering process having different characteristics is sequentially executed in a time-division manner, and the level of the frequency spectrum regarding the frequency corresponding to each scale is detected. The tone scale having the highest level of the frequency spectrum detected by the filter means and the tone scale adjacent to this scale having the highest frequency spectrum level are selected by the selecting means, and one of the two scales selected by the selecting means is selected. The level ratio of the frequency spectrum is calculated by the calculating means. The conversion means provided for each scale converts the level ratio from the computing means into shift data corresponding to the difference between the two scale frequencies, and based on the shift amount from the conversion means, the detection means An intermediate scale frequency between the two scale frequencies selected by the selecting means is detected.

Therefore, digital filtering processing detects fixed scales adjacent to each other that are close to the actual scale of a given acoustic signal, calculates the frequency level ratio of the adjacent scales, and calculates the actual scale of the acoustic signal. The frequency of can be detected. As a result, it is possible to detect an intermediate scale frequency between the preset scale frequencies.

[0009]

EXAMPLE An example of the present invention will be described below.

<Basic Principle> First, the basic principle of digital filter processing used in the embodiment will be described. In the embodiment, the digital filter processing is performed by a DSP (Digital Sign).
al Processor: digital signal processor). That is, a program and data necessary for operating as a digital filter are written in the DSP to configure a bandpass filter H _t (z) as shown in FIG. 1 and an envelope extraction circuit to configure a digital filter processing and Envelope extraction processing is performed.

In FIG. 1, an input signal X (n) is input after converting a value obtained by sampling an analog acoustic signal at a predetermined sampling timing into a digital signal (original signal may be left unchanged if originally supplied as a digital signal). The input signal X (n) is subjected to the filtering processing of the n band-pass filters H _t (z) by the time division processing of the DSP. In this filtering process, the transfer functions of the n bandpass filters H _t (z) are changed depending on each scale of a plurality of octaves.

FIG. 2 shows the magnitude of frequency characteristics when a Chebyshev type band pass filter H _t (z) is used.

The transfer function in this case is as follows, where t is a suffix (subscript) designating each scale.

[0014]

[Equation 1] Here, if this bandpass filter H _t (z) is configured with i = 1, the DSP processing is

[0015]

[Equation 2] Will be executed.

When i ≧ 2, the same operation as the above equation is repeatedly executed.

The coefficient of each band pass filter H _t (z) can be obtained by numerical calculation.

As a concrete example, if a bandpass filter of A ₄ = 440 Hz is constructed under the following conditions (see FIG. 2), the digital filtering processing of the transfer function having the following coefficient values will be executed. ..

That is, = 1 dB = (sampling frequency f _s ) = 10 KHz = 12 dB or more = 415 Hz = 430 Hz = 450 Hz = 466 Hz, and the coefficients of the two-stage digital filter with i = 1 and 2 are as follows. become.

[0020] For H440Hz (0) = 0.08192384 i = 1, a 1 (1) = - 1.91442200776 a 2 (1) = 0.9933673 b 1 (1) = - 1.91105345727 b 2 (1) = a 1. i = 2 On the other hand, a ₁ (2) = − 1.9210712 a ₂ (2) = 0.993606 b ₁ (2) = − 1.93525314797 b ₂ (2) = 1. Thus, the calculation of the bandpass filter H _t (z) is performed at each scale. Are performed in a time-divisional manner for the resulting signals Y _t (n), t
= 1 to N is obtained.

For this result signal Y _t (n),
P performs the envelope extraction process in a time division manner. This envelope processing is performed by obtaining the peak level (absolute value) for each predetermined time interval (for example, for each frequency corresponding to each scale) for the waveform of each result signal Y _t (n). Alternatively, a specific digital filter such as described below is used as | Y _t (n) |
(Y _t (n) absolute value signal).

As described above, the envelope signal E _t (n) for each scale is obtained by the time division processing of the DSP.
From t = 1 to N, a CPU (microcomputer, etc.) such as a tone generator to which this DSP is applied executes level judgment for this output, and the original input signal X (n) It is possible to detect one or more scale signals included in the waveform.

[0023] Thus, the basic principle is a band-pass filter H _t (z) having a peak for each chromatic is performed by time division filtering process, band-pass filter H _t (z) is above The present invention is not limited to the Chebyshev bandpass filter, and various functions can be realized by various types of digital filters. The bandpass filter can also be realized by connecting a lowpass filter and a highpass filter in cascade.

In the example of the Chebyshev bandpass filter, A ₄ = 440 Hz, eight multiplications are required. Therefore, a description will be given below of one improvement principle of the filtering process that reduces the number of multiplications when performing the filter operation and facilitates real-time filtering.

<Improvement Principle> FIG. 3 shows an improvement principle for causing a DSP to perform a digital filter operation with a reduced number of multiplications. This bandpass filter is a highpass filter H ₁ (z) and a lowpass filter H _2t. (z) and a low-pass filter H _E (z).

[0026] In the band-pass filter is input by the digital representation acoustic signal X (n) is first, is input to a high-pass filter H ₁ (z), the high-pass filter H ₁ (z) is the details of which will be described later , 0 at frequency 0, frequency f
_It is a high-pass digital filter that maximizes at _s / 2.

The output Y (n) of this high-pass filter H ₁ (z)
Is a low-pass filter H _{2t that} operates in time division for each scale t
The low-pass filter H _2t (z) input to (z) has the characteristics of a resonance-type low-pass digital filter having a peak at the scale frequency, the details of which will be described later.

Therefore, the high pass filter H ₁ (z)
The magnitude of the frequency characteristics of the digital filter obtained by cascading low-pass filter H _2t (z) with
It has become a pseudo bandpass filter.

In FIG. 4, f ₁ , f ₂ , ... F _N correspond to each scale frequency, and N can be set to about 40 to 50 (3 octaves to 4 octaves). In addition,
To detect the scale in a wider octave range than this, it can be achieved by adopting a high-speed DSP or parallel processing by a plurality of DSPs.

Output W of this low-pass filter H _2t (z)
_t (n) and t = 1 to N are given to the low-pass filter H _E (z) that operates in time division for each scale, and this low-pass filter H _E
E characteristics of (z) is also described later, the low-pass filter H _E (z)
Each output E _t (n) of is an envelope signal for each scale, as in FIG. The subsequent processing is the same as in the case of the basic principle.

Next, the configuration and characteristics of each digital filter shown in FIG. 3 will be described in detail.

High Pass Filter H ₁ (z) FIG. 5 shows an example of the configuration of the high pass filter H ₁ (z).

This high-pass filter H ₁ (z) has a second-order F
It is an IR (Finite Impulse Response) digital filter whose transfer function is

[0034]

[Equation 3] Is.

In FIG. 5, 5-1, 5-2 are delay elements, 5-3, 5-4, 5-5 are multipliers, 5-6, 5-7.
Indicates an adder. When this high-pass filter H ₁ (z) is realized by a DSP,

[0036]

[Equation 4] Will be executed. In this case, the multiplication of the coefficient and the signal can be realized by simple shift processing.

The frequency characteristic of this high pass filter H ₁ (z) is

[0038]

[Equation 5] Therefore, as shown in the characteristics in Fig. 6, Ω = 0 (0Hz)
At the minimum and at Ω = π (f _s / 2 Hz) the maximum.

Low Pass Filter H _2t (z) FIG. 7 shows a configuration example of the low pass filter H _2t (z).

This low-pass filter H _2t (z) is a second-order IIR (Infinite Impulse Response) digital filter whose transfer function is

[0041]

[Equation 6] Is.

As will be described later, in the low-pass filter H _2t (z), θ and CY change depending on the suffix t indicating the scale, and r becomes a parameter indicating the strength of resonance (degree of peak). ..

In FIG. 7, 7-1 and 7-2 are delay elements, 7-3, 7-4 and 7-5 are multipliers, and 7-6 and 7-7.
Indicates an adder. This low-pass filter H _2t (z)
When implemented by calculation in the DSP includes performing _{W t (n) = CY ·} Y (n) + 2rcosθW t (n-1) -r 2 W t (n-2) ...... equation (2) Become.

The frequency characteristic of this low pass filter H _2t (z) is

[0045]

[Equation 7] Given in. Where the poles of this transfer function are

[0046]

[Equation 8] , There is a double zero at Z = 0. FIG. 8 shows the arrangement of poles and zeros of this transfer function, and the pole vector and zero point vector when θ is 0 <θ <π / 2. As can be seen from FIG. 8, as Ω moves along the unit circle from Ω = 0 to Ω = π, the length of vector v ₂ decreases first and then increases. Minimum vector v ₂
The length of

[0047]

[Outer 1] Is near.

Here, the magnitude of the frequency response at the frequency Ω is the ratio of the lengths of the zero point vector v ₁ and the vector v ₂ , and the phase of the frequency response is the vector v from the angle formed by the real axis and the vector v _1. It is known that the value is obtained by subtracting the angle formed by ₂ , and FIG. 9 shows only the amplitude characteristic.

That is, as can be seen from FIG. 9, the magnitude of the frequency response (amplitude characteristic) is proportional to the reciprocal of the magnitude of the pole vector v ₂ , and becomes maximum at Ω close to θ. And r
The sharpness of this peak is determined according to the magnitude of, and a filter having a steep peak (resonance characteristic) can be realized by bringing r closer to 1.

As is clear from the above description, if the value of θ is determined for each scale (θ = 2πf _t / f _s ), FIG.
As shown in, it is possible to realize a low-pass filter H _2t (z) with resonance having a peak at the scale frequency f _t .

In the experiment, r is set to a magnitude that does not affect the level of the next scale, and CY is set to a magnitude that an output W ₁ (n) of an equivalent level is obtained in each scale. Or it can be calculated mathematically.

For example, the scale frequency (f _t ) of f and Δf
Scale frequency f + Δf next to the distance (ie f _{t + 1} )
When the ratio of the magnitude of the frequency response to and is m: 1,

[0053]

[Equation 9] By solving the quartic equation for r, the one satisfying 0 <r <1 is selected, and each coefficient −2rcos θ, r ² can be obtained. Now, as a result of numerical calculation, for example,
If f _s = 5 KHz, f = 440 Hz, and m = 4, then −2r cos θ = −1.9773, r ² = 0.9851, CY = 36.
7 The same applies to other scales.

Low Pass Filter H _E (z) FIG. 11 shows an example of the configuration of the low pass filter H _E (z).

This is the low pass filter H described above.
_{It is} a second-order IIR digital filter of the same form as _2t (z) and its transfer function is

[0056]

[Equation 10] Is.

This is obtained by _setting r = 0.9 and θ = 0 in the transfer function of the low pass filter H _2t (z).

In FIG. 11, 11-1 is an absolute value circuit for converting the input signal (output signal of the low-pass filter H _2t (z)) W _t (n) into an absolute value, and its output | W _t (n ) | Is digitally filtered. 11-2, 11-3 are delay elements, 11-4, 11-5, 11-6 are multipliers, 1
Reference numerals 1-7 and 11-8 denote adders. When this low-pass filter H _2t (z) is realized by a DSP, E _t (n) = CE | W _t (n) | + 1.8E _t (n-1) -0.81E _t (n-2) ... The formula (3) is executed.

This low-pass filter H _2t (z) is a low-pass filter with resonance whose frequency characteristic has a peak at θ = 0 as described above, and has a characteristic (amplitude characteristic) as shown in FIG. ). Where the coefficient CE
Is a factor that makes the level of each scale uniform, and can be obtained as appropriate by experiments.

FIG. 13 schematically shows the embep signal E _t (n) obtained by the configuration of FIG.

Thus, the low pass filter H _2t (z)
Is low-pass filtered after all the negative peak values (broken lines in FIG. 11) have been converted to positive peak values by the absolute value circuit 11-1, so that this waveform signal | W _t (n) |
The filter circuit takes an operation to obtain the DC component of the.

<Overall Structure of Embodiment> Next, a specific structure of an embodiment of the scale detecting device of the present invention will be described.

FIG. 14 is an overall block diagram of the musical tone generating apparatus 1 to which the scale detecting apparatus according to the present invention is applied.

The tone generator 1 is a CPU (Central Proc
essing unit) 2, ROM (Read Only Memory) 3, RA
An M (Random Access Memory) 4, a scale detection device 5, a keyboard 6, a display 7, a printer 8, a tone generation circuit 9, an audio system 10, a speaker 11 and the like are provided, and these units are connected by a bus 12. ..
The ROM 3 stores various programs such as a musical tone generator and various data. In particular, the ROM 3 stores an intermediate sound detection data table used in an intermediate sound detection process described later, and this intermediate sound detection data table will be described later.

The CPU 2 controls each part of the musical tone generating apparatus 1 according to the program in the ROM 3 to perform the processing as the musical tone generating apparatus 1 and the intermediate tone detecting processing. CPU2 is
In particular, as will be described later, a selecting means for selecting the scale having the highest level of the detected frequency spectrum of the scale detecting device 5 and the scale adjacent to the scale having the larger frequency spectrum level, the selected two scales. Calculating means for calculating the level ratio of the frequency spectrum, conversion means for converting the level ratio from the calculating means into shift data corresponding to the difference between the two scale frequencies, and the shift amount from this converting means. Based on the above, the detecting means for detecting an intermediate scale frequency of the two scale frequencies selected by the selecting means is executed.

The RAM 4 stores data for operating the scale detecting device 5 described later as a digital filter and an envelope extracting device, and also functions as a work memory of the CPU 2.

The scale detecting device 5 includes a microphone 41,
Low-pass filter 42, A / D converter 43, DSP4
4, a filter coefficient ROM 45, a work RAM 46, etc. are provided, and the line input L is used as an acoustic signal input terminal.
It has INE IN.

The scale detecting device 5 is provided with the microphone 4
1 or line input The sound signal input from LINE IN (this may be a musical instrument sound, a human voice sound, or a reproduced sound from a tape recorder, radio, TV, CD player, etc.) is appropriately filtered by the low-pass filter 42. Then, at an appropriate sampling frequency f _s , A / D
Converted to digital signal X (n) from converter 43
44.

The filter coefficient ROM 45 stores various coefficients necessary for causing the DSP 44 to function as a digital filter, and is read out as needed and output to the DSP 44.

The work RAM 46 is a work memory when the DSP 44 operates as a digital filter,
Data for filtering calculation and A / D converter 4
Digital input signal X (n) and DSP input from 3
The waveform signal and the like processed by 44 are stored.

As will be described later, the DSP 44 stores the coefficients stored in the filter coefficient ROM 45 and the work RAM 4
6 is used to perform arithmetic processing, digital filtering processing, and envelope extraction processing.

The processing result of the DSP 44 is sent to the CPU 2, and the CPU 2 of the scale detection device 5 is based on the detection result of the scale detection device 5 and the storage state of various modules incorporated in the tone generation circuit 9 described later. The detected musical scale and the state of the musical tone generated by the musical tone generating device 9 are judged, and the process of allocating the musical scale to the generating module of the musical tone generating circuit 9 and the deleting process are performed.

The keyboard 6 is provided with a function switch, a keyboard, etc. When the keyboard 6, etc. is operated by a switch, a keyboard, etc., the CPU 2 detects this operation and generates a sound module for the musical sound generating circuit 9. Assign the generated tone to.

The display 7 and the printer 8 are CP
It operates under the control of U2 and is detected by the scale detection device 5.
To display multiple scales and print on paper. For example, the CPU 2 may display the scale included in the sound being input in real time on the display 7, or display the score as the score on the display 7 after the editing work or the like in the non-real time. Or print it on.

The scale detecting device 5, as will be described later,
As a whole, it functions as a filter means for detecting the level of the frequency spectrum related to the frequency corresponding to each scale by sequentially performing digital filtering processing with different characteristics on a digital waveform signal expressing a given acoustic signal in a time division manner. ing.

As the tone generator 9, various types of tone generators can be applied. For example, PCM system, F
A sound source generation circuit such as an M method, an iPD method, or a sine wave synthesis method is applied. The musical tone generating device 9 has a plurality of musical tone generating channels, for example, four channels, and drives the audio system 10 to generate the musical scale of the musical scale number assigned to the sound generating module of the CPU 2, thereby causing the speaker 11 to operate. Sound output via.

As the audio system 10, a normal audio system is used, and not only the signal from the tone generating circuit 9 but also the audio system 10 is used in the audio system 10.
Signals from the microphone 41 and line input LINE IN are also given, and output as an acoustic output as required. Further, the musical tone generating circuit 9 can generate a musical tone signal of a tone color according to the tone color designation of the keyboard 6, and in this case also, the musical tone generating operation is performed by allocating the scale tone to be output by the CPU 2 to the tone generating module. Further, the CPU 2 sequentially stores the changes in the scale sound detected by the scale detection device 5 in the RAM 4 as sequencer information, and sequentially reads this sequencer information in response to a play start instruction of the keyboard 6 and the like from the tone generation circuit 9. It is also possible to generate a corresponding tone signal.

<Configuration of DSP> FIG. 15 shows a DSP functioning as a digital filter and an envelope extraction circuit.
44 is a circuit configuration diagram of a DSP 44, which includes an interface 441, an operation ROM 442, an address counter 443, a decoder 444, a multiplier 445, an adder / subtractor 446, a register group 447, and a flag register 44.
It has 8 etc.

The interface 441 is connected to the CPU 2 and the A / D converter 43 shown in FIG. 14 via a bus, receives an audio input signal and a command from the CPU 2 via the interface 441, and outputs the processing result. A scale signal or the like is output.

The operation ROM 442 stores a program as a digital filter and an envelope extraction circuit used in the musical tone generator 1, and the program memory 2 sequentially outputs the program contents to the decoder 444 by the address designation of the address counter 443. And output to each section.

The decoder 444 is an operation ROM
Decode the program contents read from 442,
The control signal is output to each unit of the DSP 44.

The filter coefficient ROM 45 and the work RAM 46 of the scale detection device 5 are connected to the bus of the DSP 44, coefficient data and waveform signals are appropriately supplied to the DSP 44 according to the program of the operation ROM 442, and the DSP 44 performs arithmetic processing. The generated waveform signal is output to and written in the work RAM 46.

The multiplier 445 multiplies the input data and multiplies the operation result by the adder / subtractor 446 and the register group 4
Output to 47 etc.

The adder / subtractor 446 performs addition processing or subtraction processing on the input data and outputs the operation result to the register group 44.
7 to the multiplier 445, the work RAM 46, etc., and the code data of the operation result is output to the flag register 4
Output to 48.

The flag data of the flag register 448 is
The program content output from the operation ROM 442 is output to the address counter 443, and the flag data of the flag register 448 to the address counter 443 determines the program content output from the operation ROM 442. That is, the judgment process is performed by the flag data of the flag register 448.

Next, the operation will be described.

The processing of the musical tone generator 1 is mainly a scale detection processing for detecting a scale note of a fixed scale from an input sound signal, an intermediate tone detection processing for detecting an intermediate scale, and the detected scale range. According to the level, it is possible to classify into scale tone generation processing in which a predetermined tone color is generated. Hereinafter, the scale detection process, the intermediate tone detection process, and the scale tone generation process will be described.

<Scale Detection Processing> First, the scale detection processing for the fixed scale in the scale detection device 5 will be described.

The scale detection process is performed under the control of the CPU 2, and the scale detection process is performed by the digital filter process described in the above-mentioned improvement principle.

The CPU 2, first, as shown in FIG.
At the start of the scale detection process, an initial process is performed (step S1). This initial process is mainly a process of clearing the work RAM 46.

When the initial processing is completed, it is checked whether or not the digital conversion of the acoustic signal into the digital signal X (n) by the A / D converter 43 is completed (step S
2) When the digital conversion is completed by the A / D converter 43, the digital signal X input from the A / D converter 43 is input.
Addresses (n) are sequentially set and stored in the work RAM 46 (step S3). In this case, by using a specific area of the work RAM 46 as a ring buffer (a buffer configured by virtually connecting the end and the start), an unlimited input signal (digital signal X
(n)).

A digital signal X (n) is input to the work RAM 46.
When is stored, the CPU 2 causes the DSP 44 to operate as a high pass filter H ₁ (z) of FIR (step S
4). The operation processing as the FIR high pass filter H ₁ (z) is performed by the operation ROM 442 of the DSP 44.
Address setting of the program of the address counter 443
And the coefficient setting in the filter coefficient ROM 45 for the high pass filter H ₁ (z).

The calculation as the high-pass filter H ₁ (z) is based on the above equation (1). In addition to the input digital signal X (n) this time, the input digital signal input X (n -1), X (n-2) is read,
It is executed by using the multiplier 445 and the adder / subtractor 446 in the DSP 44.

When the calculation process as the high pass filter H ₁ (z) is completed, the set value t for performing the filter process for each scale is set to the initial set value t = 1 (step S5), and the low pass filter H _2t ( The filtering operation as z) is performed (step S6). The calculation as the low-pass filter H _2t (z) is based on the above equation (2), and each coefficient CY, 2rcos θ, r ² is converted into a filter coefficient R.
Multiplier 4 in DSP44 while reading from OM45
45, using the adder / subtractor 446. This calculation result W _t (n) is also sequentially stored by using another specific area of the work RAM 46 as a ring buffer. In this case as well, by using the work RAM 46 as a ring buffer, it is possible to read out the previous and the previous two previous calculation results W _t (n-1) and W _t (n-2) one after another and use them in the calculation. ..

When the calculation processing as the high-pass filter H ₁ (z) and the low-pass filter H _2t (z) is completed, the envelope extraction processing for each scale is next performed. This envelope extraction process is performed by the DSP 44 for each scale.
The R low-pass filter H _E (z) is calculated and processed (step S7).

[0096] The operation of the low-pass filter H _E (z), due to the above formula (3), each coefficient CE, 1.8, -
While reading 0.81 from the filter coefficient ROM 45,
This is performed using the multiplier 445 and the adder / subtractor 446 in the DSP 44. Of these operations, absolute value calculation | W _t (n) | is also executed using the adder / subtractor 446.

This calculation result E _t (n) is also stored in the work RAM 46.
By further using another specific area 46 of as a ring buffer and sequentially storing, the previous and last two calculation results E _t (n-1) and E _t (n-2) are read one after another. It can be used for calculation.

Upon completion of the above filter processing and envelope extraction processing, it is checked whether or not these processings have been performed for all scales (t = N) (step S).
8) If the processing has not been completed for all scales, the set value t is incremented (step S
9) and proceeds to step S6. When the process proceeds to step S6, the filtering process of steps S6 and S7 is executed again.

In step S8, if the filtering process and the like have been completed for all the scales, the envelope E _t (n) (t = 1 to N) for each scale is notified to the CPU 2 (step S10). It is checked whether or not the scale detection processing mode is ended (step S11). When the scale detection processing mode has not ended, the process proceeds to step S2, waits for the A / D conversion of the next acoustic signal, and the same processing is performed. That is, the DSP 44 executes digital filtering of three systems in time-division in order for each sampling and repeatedly for each scale, and in real time, according to the envelope of each scale.
It is possible to detect the level of the frequency spectrum with respect to the frequency corresponding to each fixed scale.

In step S11, the CPU 2 causes the DSP 4 to terminate the scale detection processing mode by operating the keyboard 5 or the like.
When 4 is notified, the series of processing operations is ended.

<Scale tone generation processing of CPU2> CPU2
Is the envelope signal E _t (n) (t = 1 to 1) for each scale at every sampling period from the DSP 44 as described above.
N), that is, the level of the frequency spectrum related to the frequency corresponding to each scale is given, and can be used for various purposes, but in this embodiment, the case where the input acoustic signal is a single tone will be described. ..

The scale sound generation processing corresponding to the mode setting will be described below with reference to FIG.

As shown in FIG. 17, the CPU 2 first checks whether or not there is a result notification from the DSP 44 in order to perform the scale sound generation processing (step P1). The result notification check process is performed by checking whether or not a scale detection end signal indicating that the scale detection process of the scale detection device 5 has been completed is input from the scale detection device 5.

When the scale detection end signal is input, the envelope value E _t (n) of each scale detected by the scale detecting device 5 is written in the RAM 4 (step P2). When the writing of the detected envelope value E _t (n) of each scale to the RAM 4 is completed, the largest envelope value E _t (n) of these scales is taken out and stored in the RAM 4 as Max.
Write to (step P3) and judge whether Max exceeds a predetermined threshold value (step P4).

When Max does not exceed the threshold value in step P4, it is checked whether or not there is a musical tone being generated, that is, whether there is a musical tone signal being generated by the musical tone generating circuit 9 (step P5). When there is no tone of No., the procedure returns to step P1 to prepare for input of the envelope value E _t (n) of the next scale from the DSP 44.

In step P5, when there is a musical tone being generated, the CPU 2 judges that the sound input of the scale tone adopted as Max from the microphone 41 or the line input LINE IN has stopped by the previous sampling, and the CPU 2 generates the musical tone. The circuit 9 is instructed to start muting (step P6), and the process returns to step P1 to prepare for the next input from the DSP 44.

In step P4, when Max exceeds the predetermined threshold value, the envelope value of the upper and lower two scales adjacent to the maximum scale of the envelope value E _t (n) of each scale detected by the scale detecting device 5 Se with a larger E _t (n)
The cond is stored in the RAM 4 (step P7), and an intermediate sound detection process is performed (step P8).

That is, the actual scale of the input acoustic signal is
The maximum scale of the envelope detected by the scale detection device (Ma
x), it is considered that the scale is shifted to the side with the larger envelope of the upper and lower scales adjacent to this scale.

By this intermediate tone detection process, as will be described later, when the scale of the input sound is a scale between fixed scales that can be detected by the scale detection process, and between the fixed scales and the fixed scales. Detects the actual scale of.

When the actual scale is detected by the intermediate tone detection process, it is next checked whether or not the scale tone detected by the above scale detection process is currently sounding (step P9). When the sound is being generated, it is not necessary to newly generate a fixed-tone note as a sounding target. Therefore, a bend value is set in the bend range detected by the intermediate tone detection processing, and the process returns to step P1 to perform the same processing. (Step P1
0). By setting the bend value in this bend range, a scale note of the actual scale shifted by the bend value with respect to the fixed scale is produced.

At step P9, if the scale sound detected by the scale detection process is not currently being sounded, it is checked whether another scale sound is being sounded (step P9).
11). In the present embodiment, when a different tone is being sounded, a single tone is handled, and the number of sounds that can be pronounced is limited to one. After muting the scale sound (step P12),
Instructing the pronunciation of the scale note extracted in the scale detection process this time (step P13). When the pronunciation instruction is given, a bend value is set in the bend range detected by the intermediate tone detection processing, and a scale note of an actual scale shifted by the bend value with respect to the fixed scale is sounded (step P10). That is, the musical tone generating circuit 9 produces a scale tone shifted from the designated scale by the bend value when the tone generation instruction is issued and the bend value is set.

At step P11, when another scale tone is not being sounded, the scale tone extracted in the scale detection process this time is set as the sounding target and the pronunciation is instructed (step P1).
3), a bend value is set to the bend range detected by the intermediate tone detection processing, and a scale tone shifted from the designated scale by the bend value is generated (step P10).

<Intermediate Tone Detection Process of CPU2> In the intermediate tone detection process, the CPU 2 adjoins the maximum scale of the envelope value E _t (n) of each scale detected by the scale detection device 5 in step P7 of FIG. If the one with the larger envelope value E _t (n) of the upper and lower scales is stored in the RAM 4 as the Second, the one with the lower scale between Max and Second is Low.
, Set the higher scale to High (step Q
1).

Next, the address of the intermediate scale detection data table of the ROM 3 is set to Low (step Q2), and Low
And the high envelope are calculated, and the offset of the intermediate scale data (actual scale data) is obtained from this calculation result (step Q3).

The scale detected by the scale detection process is set to Low.
Then, the shift amount from Low based on the intermediate scale detection data table of the ROM 3 stored therein is stored as bend bend data (step Q4), and the intermediate tone detection processing ends.

In the intermediate scale detection data table of the ROM 3, for example, the transfer function of the digital filter is shown in FIG.
As shown in FIG. 9, assuming that they are 221 Hz and 234 · 14 Hz, the ratio of these two transfer functions, that is, the transfer function of the peak frequency 221 Hz is the peak frequency 234.14 Hz.
20 is obtained by dividing by the transfer function of
21 is an enlarged view of the vicinity of 200 Hz to 250 Hz. As shown in FIG. 21, the transfer function ratios of the digital filters for detecting adjacent tones are such that the maximum point and the minimum point coincide with the peaks of the two transfer functions, and between these two points, it decreases monotonically. ing. Therefore, when the ratio of the transfer function between the extreme values is known, the actual frequency is known by the inverse transformation. Therefore, a conversion table for converting the frequency from the transfer function ratio is stored in the ROM 3 as an intermediate scale detection data table, and the shift amount is obtained from the intermediate scale detection data table in step Q4 of FIG. For example, in the example of the above frequency, a conversion table to intermediate scale based on 221 Hz is stored in the ROM 3, and the intermediate scale detection data table is based on all the fixed scales detected by the scale detection processing. There is a conversion table for each intermediate scale. In the intermediate scale detection data table, it is actually sufficient to divide the semitones into cents.

The bend data stored in step Q4 of FIG. 18 is used as the scale detected by the scale detection process in step P10 of FIG. 17, that is, as the shift amount from Low.
The bend value is set to the bend range, and the tone generation circuit 9
A tone of a scale shifted from the scale (Low) instructed to generate by this bend value is generated.

Therefore, the fixed scales adjacent to each other, which are close to the actual scale of the given acoustic signal, are detected by the digital filtering process, the frequency level ratio of the adjacent scales is calculated, and the actual scale of the acoustic signal is calculated. Can be detected. As a result, it is possible to detect an intermediate scale between pre-fixed scales, and it is also possible to generate an intermediate scale tone between the detected pre-fixed scales.

In the above embodiment, a single tone has been described, but the present invention is not limited to this, and an intermediate tone can be similarly detected for a compound tone and a compound tone of the detected intermediate tone can be generated.

[0120]

According to the present invention, fixed scales adjacent to each other that are close to the actual scale of a given acoustic signal are detected by digital filtering processing, and the frequency level ratio of the adjacent scales is calculated, Intermediate scale frequencies between pre-fixed scale frequencies can also be detected. As a result, the actual scale of the acoustic signal can be accurately detected.

[Brief description of drawings]

FIG. 1 is a block diagram showing the basic principle of an embodiment of a scale detection device according to the present invention.

2 is a frequency characteristic diagram of the bandpass filter H _t (z) of FIG.

FIG. 3 is a configuration diagram based on the principle improved from FIG.

4 is a frequency characteristic diagram when the high-pass filter H ₁ (z) and the low-pass filter H _2t (z) of FIG. 3 are cascade-connected.

5 is a configuration diagram of a high pass filter H ₁ (z) of FIG.

6 is a frequency characteristic diagram of the high-pass filter H ₁ (z) of FIG.

7 is a configuration diagram of the low pass filter H _2t (z) of FIG.

8 is a diagram showing poles and zeros and pole vectors and zero vectors of the digital filter shown in FIG.

9 is a diagram showing frequency characteristics corresponding to FIG.

10 is a frequency characteristic diagram of the low pass filter H _2t (z) of FIG. 7.

11 is a configuration diagram of the low-pass filter H _E (z) of FIG.

12 is a frequency characteristic diagram of the low pass filter H _E (z) of FIG.

13 is an explanatory diagram illustrating that envelope extraction is performed by the configuration of FIG. 11.

FIG. 14 is an overall circuit configuration diagram of a musical tone generating device to which an embodiment of a scale detecting device according to the present invention is applied.

15 is an internal circuit configuration diagram of the DSP 44 of FIG.

16 is a flowchart of digital filter processing by the DSP 44 of FIG.

FIG. 17 is a flowchart showing a scale sound generation process.

FIG. 18 is a flowchart showing the intermediate sound detection processing of FIG.

FIG. 19 is a diagram showing an example of a transfer function of a digital filter.

20 is a diagram showing a ratio of transfer functions of FIG.

21 is a partially enlarged view of FIG.

[Explanation of symbols]

 1 tone generator 2 CPU 3 ROM 4 RAM 5 scale detector 6 keyboard 7 display 8 printer 9 tone generator circuit 10 audio system 11 speaker 12 bus 22 tone color buffer 23 polypitch extraction data buffer 24 maximum level buffer 25 specific range buffer 44 DSP 45 Filter coefficient ROM 46 Work RAM

Claims (1)

[Claims] 1. A filter for detecting a level of a frequency spectrum relating to a frequency corresponding to each scale by sequentially performing time-division digital filtering processing having different characteristics on a digital waveform signal expressing a given acoustic signal. Means, selection means for selecting one of the scales having the highest level of the frequency spectrum detected by the filter means and the scale adjacent to this scale, which has a larger level of the frequency spectrum, and two scales selected by the selection means. Calculating means for calculating the level ratio of the frequency spectrum, converting means provided for each scale, and converting the level ratio from the calculating means into shift data corresponding to the difference between the two scale frequencies; Between the two scale frequencies selected by the selection means based on the shift amount of Scale detecting apparatus characterized by comprising detecting means for detecting the intermediate scale frequency, a.