JPH04318597A - Chord decision device and automatic accompaniment device using the same - Google Patents

Abstract

PURPOSE:To accurately and speedily decide a chord from an input acoustic signal and to automatically play an accompaniment. CONSTITUTION:The input acoustic signal is filtered digitally by a digital signal processing means 44 to detect envelope values of frequency spectra corresponding to respective intervals and an interval detecting means 44 detects an interval sound included in the acoustic signal from the detection result. An integrating means 2 integrates an envelope value of the same sound name to an interval sound whose envelope value exceeds a specific reference value among the detected interval sounds and a chord data generating means 2 compares chord constitution as the integration result with normal chord constitution stored in a chord data storage means 3. A chord data generating means 2 generates corresponding chord data when the chord constitution of the inputted interval sound is the normal chord constitution and also generates an accompaniment sound with the generated chord data.

Description

[Detailed description of the invention]

0001

[Industrial Application Field] The present invention relates to a chord detection device and an automatic accompaniment device using the same, and more specifically, it extracts pitches from musical instrument sounds, human voice sounds, etc., and detects musical scales from the extracted pitches. The present invention relates to a chord discrimination device that discriminates chord data based on the scale data, and an automatic accompaniment device using the same.

0002

[Background Art] Traditionally, musical instrument sounds, human voice sounds, etc. are input, pitches are extracted, scales are determined, and the results are printed out in the form of musical scores, or a series of determination results are encoded and recorded. After that, a technology has been proposed in which the sound is output as a separate instrument sound and played automatically (Japanese Patent Laid-Open No. 57-692).
No., JP-A-58-97179, etc.).

[0003]However, the reality is that such conventional techniques can basically only handle the input of single tones, and have not considered the input of multiple tones (including chords).

[0004] Therefore, it has been proposed to detect a chord name from an input chord and display the chord name in accordance with the chord name signal (Utility Model Application No. 60-26091). However, what is disclosed in this publication is that analog bandpass filter circuits are provided for the number of musical scales, peak hold is performed on each output, and a level detection circuit is used to select constituent notes that make up a chord from those with large peaks. This means that

[0005] According to the technology using such an analog filter, the determination result may fluctuate due to the influence of external temperature,
There is a problem that it is not stable, and there is also the disadvantage that the circuit configuration becomes large-scale and becomes large-scale.

[0006] Therefore, the present applicant first attempted to detect the scale of the input acoustic signal in a short time, whether it was a single note or a compound note, and in a small-scale circuit. We are proposing a digital scale detection device that operates stably and an electronic musical instrument using it (Patent application No.
No. 3789).

[0007] This scale detection device basically includes a storage means for sequentially storing digital waveform signals representing a given acoustic signal, and a storage means for sequentially storing digital waveform signals representing a given acoustic signal. Digital signal processing means that sequentially performs digital filtering with different characteristics in a time-division manner in order to detect the level of the frequency spectrum regarding the corresponding frequency, and the above-described method based on the results of the digital filtering performed by this digital signal processing means. Detecting means for detecting one or more scale notes included in an acoustic signal, and using a scale note detection device, an electronic musical instrument using a scale note detecting device, in addition to the above-mentioned means, detects one or more scale notes detected by the detection means. The apparatus includes musical tone signal generating means for generating musical tone signals corresponding to a plurality of scale tones with predetermined tones.

[0008] Therefore, according to the scale detection device previously filed by the present applicant, all signal processing can be performed digitally, regardless of whether the input acoustic signal is a single note or a compound note. , detect acoustic scales in a short time,
Moreover, it is possible to operate stably with a small-scale circuit.

[0009]

[Problem to be Solved by the Invention] However, in the scale detection device and electronic musical instrument using the same, which the present applicant previously applied, it is possible to accurately detect and detect scale tones from input acoustic signals. Since the scale notes were to be pronounced accurately, the detected scale notes were only pronounced regardless of whether or not they had a scale structure according to normal chord data. There was a risk that the accompaniment notes could not be used to produce appropriate accompaniment notes.

[0010] Therefore, the inventions of the present application perform digital filtering processing on an input acoustic signal in a time-division manner to extract a scale, and determine the type of chord from the scale structure, so that the detected acoustic signal is a chord. The purpose is to determine whether the chord data is appropriate, detect appropriate chord data, and enable accompaniment using it.

[0011]

[Means for solving the problem] The invention according to claim 1 includes:
A digital signal processing means that detects the envelope value of a frequency spectrum regarding a frequency corresponding to each scale by sequentially performing digital filtering processing with different characteristics on a digital waveform signal representing an acoustic signal in a time-sharing manner; scale detection means for detecting one or more scale tones included in the acoustic signal based on a processing result of digital filtering by the means; chord data storage means for storing chord data corresponding to a chord configuration; The present invention is characterized by comprising chord data generating means for comparing the chord structure of the scale tones detected by the means with the chord structure of the chord data storage means to generate corresponding chord data.

[0012] According to the second aspect of the invention, the envelope value of the frequency spectrum regarding the frequency corresponding to each scale is obtained by sequentially performing digital filtering processing with different characteristics on a digital waveform signal expressing an acoustic signal in a time-sharing manner. digital signal processing means for detecting; scale detecting means for detecting one or more scale tones included in the acoustic signal; and chord data corresponding to a chord structure based on the processing result of digital filtering by the digital signal processing means; a chord data storage means for storing, a chord data generation means for generating corresponding chord data by comparing the chord structure of the scale tones detected by the scale note detection means with the chord structure of the chord data storage means; chord generating means having a predetermined accompaniment pattern and generating accompaniment tones by converting changes in the accompaniment pattern into changes in scale notes based on chord data generated from input of the chord data generating means; It is characterized by the fact that it is equipped with

[0013] The chord data storage means may store, for example, a chord structure consisting of a plurality of scales, and chord data consisting of a root note and a chord type corresponding to this chord structure. , may be stored in the form of a data table.

[0014] The digital signal processing means, for example,
As described in claim 4, bandpass filtering processing with a frequency corresponding to each of the scales as a center frequency is sequentially executed in a time-sharing manner, and as described in claim 5, each of said scales is By sequentially performing bandpass filtering processing with a frequency corresponding to the musical scale as the center frequency in a time-sharing manner, and sequentially performing signal processing operations to extract the envelope from the waveform signal obtained as a result of this bandpass filtering processing, each of the above-mentioned The envelope value of a frequency spectrum related to a frequency corresponding to a musical scale may be detected, and as described in claim 6, high-pass filtering processing with a predetermined characteristic is performed, and the A method in which low-pass filtering processing to which resonance having a peak in frequency is added is performed sequentially in a time-sharing manner, and claim 7
As described in , high-pass filtering processing with predetermined characteristics is performed, and low-pass filtering processing with peaked resonance is sequentially performed at frequencies corresponding to the above-mentioned scales in a time-sharing manner. The envelope value of the frequency spectrum regarding the frequency corresponding to each scale may be detected by sequentially performing signal processing calculations to extract the envelope from the waveform signal obtained as a result of the filtering process.

Further, as described in claim 8, the scale detecting means detects an envelope having the same note name for a scale whose envelope value exceeds a predetermined reference value among the detected scale notes. It may include an integrating means for integrating the values a predetermined number of times, and may select a plurality of scale tones in descending order of the envelope values integrated by the integrating means and supply them to the chord data generating means.

Furthermore, the scale detecting means may assign the detected scale note to one of the 12 scales regardless of the octave difference between the detected scale notes.

[0017]

According to the first aspect of the invention, an acoustic signal generated from a musical instrument such as a piano or an electronic musical instrument is digitally converted and input to the digital signal processing means. The digital signal processing means sequentially time-divisionally performs digital filtering processing with different characteristics on the digital waveform signal representing the acoustic signal, detects the envelope value of the frequency spectrum regarding the frequency corresponding to each scale, and generates the scale. Output to detection means. The scale detection means is
Based on the results of digital filtering performed by the digital signal processing means, one or more scale tones included in the acoustic signal are detected and output to the chord data generation means. On the other hand, chord data corresponding to the chord structure is stored in the chord data storage means, and the chord data generation means
The scale tones inputted from the scale detection means are compared with the chord structure of the chord data storage means to generate corresponding chord data.

Therefore, it is possible to quickly and appropriately determine whether the input acoustic signal has an appropriate chord structure, and when the input acoustic signal has an appropriate chord structure, the corresponding chord data can be generated. can do.

According to the second aspect of the invention, an acoustic signal generated from a musical instrument such as a piano or an electronic musical instrument is digitally converted and input to the digital signal processing means. The digital signal processing means sequentially time-divisionally performs digital filtering processing with different characteristics on the digital waveform signal representing the acoustic signal, detects the envelope value of the frequency spectrum regarding the frequency corresponding to each scale, and generates the scale. Output to detection means. The scale detection means detects one or more scale tones included in the acoustic signal based on the result of digital filtering performed by the digital signal processing means, and outputs the detected scale tones to the chord data generation means. On the other hand, chord data corresponding to the chord structure is stored in the chord data storage means, and the chord data generation means compares the scale tones inputted from the scale detection means with the chord structure of the chord data storage means, and Generate chord data. The generated chord data is output to an accompaniment sound output means, and the accompaniment sound output means converts a previously prepared accompaniment pattern into a scale change pattern based on the input chord data and outputs an accompaniment sound.

Therefore, it is possible to quickly and appropriately determine whether the input acoustic signal has an appropriate chord structure, and when the input acoustic signal has an appropriate chord structure, the corresponding chord data can be generated. Automatic accompaniment can be performed. As a result, appropriate accompaniment sounds can be output.

According to the third aspect of the invention, the chord data storage means stores a chord structure consisting of a plurality of scales and chord data consisting of a root note and a chord type corresponding to this chord structure in a data table. I remember it in format.

Therefore, it is possible to determine whether each scale note data inputted from the scale detecting means has an appropriate chord structure, and when it has an appropriate chord structure, to generate the corresponding chord data. This can be done appropriately and quickly, and an appropriate accompaniment sound can be output.

According to the invention as set forth in claim 4, the digital signal processing means sequentially executes bandpass filtering processing using a frequency corresponding to each of the scales as a center frequency in a time-sharing manner, According to the above method, a bandpass filtering process with a center frequency corresponding to each scale is sequentially executed in a time-sharing manner, and a signal processing operation is sequentially performed to extract an envelope from a waveform signal obtained as a result of this bandpass filtering process. By doing this, the envelope value of the frequency spectrum related to the frequency corresponding to each of the scales is detected. Also,
In the invention as set forth in claim 6, the digital signal processing means performs high-pass filtering processing with predetermined characteristics, and sequentially executes low-pass filtering processing in which resonance having a peak at a frequency corresponding to each scale is added in a time-sharing manner. The invention according to claim 7 performs high-pass filtering processing with predetermined characteristics, and sequentially performs low-pass filtering processing to which resonance with a peak is added at frequencies corresponding to each of the scales, and further comprises: By sequentially performing signal processing operations to extract envelopes from waveform signals obtained as a result of these digital filtering processes, envelope values of frequency spectra related to frequencies corresponding to each of the above-mentioned scales are detected.

Therefore, according to the inventions recited in claims 4 to 7, all signal processing can be performed in the digital domain, and the process of extracting scale notes and, furthermore, discriminating chords, can be performed at high speed and efficiently. can be done.

According to the eighth aspect of the invention, for scale tones detected by the scale detecting means whose envelope values are greater than or equal to a predetermined value, the envelope values of the same note names are integrated a predetermined number of times, and the integrated values are A plurality of scale tones are selected in descending order of their envelope values and used to generate chord data.

Therefore, even if the pitch of the input acoustic signal fluctuates, it is possible to accurately detect the scale tones used for generating chord data.

According to the ninth aspect of the invention, the scale tone detected by the scale detecting means is set to 1 regardless of the octave difference.
Assigned to either of the two scales.

Therefore, for example, even if an acoustic signal of scale C3 is input, an acoustic signal of scale C4 is input, they are treated as the same scale C, so one of the scales making up the chord is a scale one octave higher. It becomes possible to correspond to inverted forms composed of .

[0029]

DESCRIPTION OF THE PREFERRED EMBODIMENTS An embodiment of the chord discrimination device and automatic accompaniment device using the same according to the present invention will be described below. <Basic Principle> First, the basic principle of digital filter processing used in the embodiment will be explained. In the embodiment, digital filter processing is performed using DSP (Digital Signal
This is done by using a digital signal processing processor (Al Processor). That is, D
Programs and data necessary to operate the SP as a digital filter are written to configure a bandpass filter Ht(z) as shown in Figure 1, and an envelope extraction circuit is also configured to perform digital filter processing and envelope extraction processing. I do.

In FIG. 1, the input signal X(n) is a value obtained by sampling an analog acoustic signal at a predetermined sampling timing, converting it into a digital signal (if it is originally supplied as a digital signal, it can be left as is) and inputting it. and for this input signal X(n), D
By time-sharing processing of SP, n bandpass filters H
Perform filtering processing on t(z). During this filtering process, n bandpass filters Ht(z
) is changed depending on each scale of multiple octaves.

FIG. 2 shows the magnitude of the frequency characteristic when a Chebysiev type filter is used as the bandpass filter Ht(z). The transfer function in this case is
The suffix (subscript) specifying t for each scale is as follows.

[Equation 1] Here, with i=1, this bandpass filter Ht(z
), the DSP processing is as follows:

[Math 2] will be executed.

Note that if i≧2, the same calculation as in the above equation will be repeatedly executed. Furthermore, the coefficients of each bandpass filter Ht(z) can be determined by numerical calculation. As a specific example, if a bandpass filter of A4=440 Hz is constructed under the following conditions (see FIG. 2 for ■ to ■), digital filtering processing of a transfer function having the following coefficient values will be executed.

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

[0034]H440Hz(0)=0.0819238
For 4i=1, a1(1)=-1.91442200776a2(1)
= 0.9933673 b1 (1) = -1.91105345727 b2 (1)
= 1. For i=2, a1(2)=-1.9210712 a2(2)=0.993606 b1(2)=-1.93525314797b2(2)
= 1. In this way, the calculation of the bandpass filter Ht(z) is performed for each scale in a time-sharing manner, and as a result, the signal Yt(
n), t=1 to N are found.

For this result signal Yt(n), next:
The DSP performs envelope extraction processing in a time-sharing manner. This envelope processing is performed by determining the peak level (absolute value) of the waveform of each resultant signal Yt(n) at each predetermined time interval (for example, at each frequency corresponding to each scale). Or a specific digital filter as described below |Y
t(n)|(absolute value signal of Yt(n)).

As described above, by the time division processing of the DSP, the envelope signals Et(n),
t=1 to N is determined, and the CPU (microcomputer, etc.) of the automatic accompaniment device to which this DSP is applied performs level judgment on this output, thereby determining the original input signal X(n). The scale signal included in the waveform is 1
It becomes possible to detect multiple numbers.

As described above, the basic principle is to time-divisionally perform filtering processing using a bandpass filter Ht(z) that has a peak for each scale, but the bandpass filter Ht(z) is The present invention is not limited to a band-pass filter of any type, and various functions can be realized using digital filters of various types. A bandpass filter can also be realized by cascading a lowpass filter and a highpass filter.

In the above example of the Chebyshev type bandpass filter and A4=440Hz, eight multiplications are required. Therefore, one improved principle of filtering processing that reduces the number of multiplications when performing filter calculations and facilitates filtering in real time will be described below.

<Improved Principle> FIG. 3 shows an improved principle that allows the DSP to perform digital filter operations with a reduced number of multiplications. ) and a low-pass filter HE(z). In this bandpass filter, the digitally expressed input acoustic signal X(n) is first passed through the highpass filter H1(z
), and the high-pass filter H1(z) is 0 at frequency 0 and the frequency fs/
This is a high-pass digital filter with a maximum value of 2.

Output Y of this high-pass filter H1(z)
(n) is input to the low-pass filter H2t(z) which operates in time division for each scale t, and the low-pass filter H2t(z)
Although the details will be described later, t(z) has characteristics of a resonance type low-pass digital filter having a peak at the scale frequency.

Therefore, the above high-pass filter H1 (
The frequency characteristic of the digital filter obtained by cascading the low-pass filter H2t(z) and the low-pass filter H2t(z) is as shown in FIG. 4, and is a pseudo band-pass filter.

In FIG. 4, f1, f2, ... fN are
It is possible to set N to about 40 to 50 (3 to 4 octaves) corresponding to each scale frequency. Incidentally, when detecting a scale in a wider octave range than this, it can be achieved by employing a high-speed DSP or parallel processing using multiple DSPs.

The output Wt(n), t=1 to N of this low-pass filter H2t(z) is given to a low-pass filter HE(z) which operates in a time-division manner for each scale, and the output of this low-pass filter HE(z) is Although the characteristics will be described later, each output Et(n) of this low-pass filter HE(z) becomes an envelope signal for each scale, similar to FIG. 1. 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 explained in detail. High Pass Filter H1(z) FIG. 5 shows an example of the configuration of the high pass filter H1(z). This high-pass filter H1(z) is a second-order F
IR (Finite Impulse Responses)
e) A digital filter whose transfer function is:

[Math 3] It is.

In FIG. 5, 5-1 and 5-2 are delay elements, 5-3, 5-4 and 5-5 are multipliers, and 5-6 and 5-7 are adders. This high-pass filter H1(z)
When realized by calculation on DSP,

[Formula 4] will be executed. In this case, multiplication of the coefficient and the signal can be achieved by simple shift processing.

The frequency characteristics of this high-pass filter H1(z) are as follows:

[Equation 5], and as shown in Figure 6, Ω=0 (0Hz
) and maximum at Ω=π (fs/2Hz).

Low Pass Filter H2t(z) FIG. 7 shows an example of the configuration of the low pass filter H2t(z). This low-pass filter H2t(z) is a second-order II
R (Infinite Impulse Response
se) a digital filter whose transfer function is

[Math 6] It is.

As will be described later, in the low-pass filter H2t(z), θ and CY change depending on the suffix t indicating the musical scale, and r is 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 are adders. This low-pass filter H2t(z
) is realized by calculation on DSP,
Wt(n)=CY・Y(n)+2rcosθWt(n
-1)-r2Wt(n-2)...Equation (2) will be executed.

The frequency characteristics of this low-pass filter H2t(z) are

[Math 7] is given by Here, the poles of this transfer function are

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 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 v2 first decreases and then increases. minimum vector v2
The length of

[Outside 1] It is near.

Here, the magnitude of the frequency response at the frequency Ω is the ratio of the lengths of the zero point vector v1 and the vector v2, and the phase of the frequency response is the ratio between the length of the zero point vector v1 and the vector v1.
It is known that the value is obtained by subtracting the angle formed by vector v2 from the angle formed by
become that way.

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 v2, and is maximum at Ω close to θ. And r
The sharpness of this peak is determined according to the magnitude of r, and as r approaches 1, a filter with a steep peak (resonance characteristic) can be realized.

As is clear from the above explanation, if the value of θ is determined for each scale (θ=2πft/fs),
0, it is possible to realize a low-pass filter H2t(z) with resonance that has a peak at the scale frequency ft.

In addition, r is set to a size that does not affect the level of the adjacent scale, and CY is set to a size that allows the same level of output W1(n) to be obtained for each scale.
Or it can be calculated mathematically.

For example, the scale frequency (ft) of f and Δf
The scale frequency f+Δf (i.e. ft+1
) is m:1,

[Formula 9] Solve the quartic equation for r, choose one that satisfies 0<r<1, and calculate each coefficient −2rcosθ,r2
can be found. Now, as a result of numerical calculation, for example, if fs=5KHz, f=440Hz, and m=4, -2rcosθ=-1.9773, r2=0.
9851, CY=36.7. The same applies to other scales.

Low Pass Filter HE(z) FIG. 11 shows an example of the configuration of the low pass filter HE(z). This is a second-order IIR digital filter of the same form as the low-pass filter H2t(z) explained earlier, and the transfer function is:

[Formula 10]. This is the previous low-pass filter H2t(z)
In the transfer function, r=0.9 and θ=0.

In FIG. 11, 11-1 is an input signal (
Output signal of low-pass filter H2t(z)) Wt(n
) is an absolute value circuit that converts into an absolute value, and its output |Wt
(n)| is digitally filtered. 11-2
, 11-3 is a delay element, 11-4, 11-5, 11-6
indicates a multiplier, and 11-7 and 11-8 indicate adders. When realizing this low-pass filter H2t(z) by calculation on a DSP, Et(n)=CE|Wt(n)|+1.8Et(n
-1) -0.81Et(n-2)...Equation (3) will be executed.

This low-pass filter H2t(z) is
As explained above, the frequency characteristic is a low-pass filter with resonance that has a peak at θ=0, as shown in Figure 1.
The characteristics (amplitude characteristics) shown in 2 are taken. Here, the coefficient CE is a factor that makes the level of each scale uniform, and can be determined as appropriate through experiments or the like.

FIG. 13 schematically shows the envelope signal Et(n) obtained by the configuration of FIG. 11. In this way, in the low-pass filter H2t(z), all negative peak values (broken line in FIG. 11) are converted into positive peak values by the absolute value circuit 11-1, and then the low-pass filter is applied. The filter circuit begins to operate to obtain the DC component of this waveform signal |Wt(n)|.

<Overall Configuration of Embodiment> Next, a specific configuration of an embodiment of the chord discrimination device and automatic accompaniment device using the same according to the inventions of the present application will be described. FIG. 14 is an overall block diagram of a musical tone generating device 1 to which the chord discriminating device and the automatic accompaniment device using the chord discriminating device of the present invention are applied.

[0061] The musical tone generating device 1 has a CPU (Central
l Processing Unit) 2, ROM (R
ead Only Memory) 3, RAM (Ran
dom Access Memory) 4, scale note detection device 5, keyboard 6, display 7, printer 8
, a musical tone generating circuit 9, an audio system 10, a speaker 11, etc., and these parts are connected by a bus 12.

[0062] The ROM 3 stores programs and the like for the chord discriminating device of each invention of the present application and an automatic accompaniment device using the same, and also as a chord discriminating device and an automatic accompaniment device using the same as necessary. Stores data necessary for operation. In particular, the ROM 3 stores chord data consisting of root notes and chord types corresponding to a chord configuration consisting of a plurality of scales, and for example, a chord configuration and chord data corresponding to this chord configuration are stored in a data table. is stored in the format. Therefore, the ROM 3 constitutes a chord data storage means for storing chord data corresponding to a chord configuration.

The CPU 2 controls each part of the musical tone generating device 1 according to the program in the ROM 3, and performs processing as the chord discriminating device of the present invention and an automatic accompaniment device using the same. Further, the CPU 2 includes various registers, and these registers store various data necessary for processing as a chord discriminating device and an automatic accompaniment device using the same.

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

The scale sound detection device 5 includes a microphone 41
, low-pass filter 42, A/D converter 43, DSP4
4. Equipped with a filter coefficient ROM 45, work RAM 46, etc., and a line input as an input terminal for audio signals.
Equipped with LINE IN.

[0066] This scale tone detection device 5 receives an acoustic signal inputted from the microphone 41 or the line input (LINE IN) (this may be a musical instrument sound, a human voice sound, or a reproduced sound from a tape recorder, radio, television, CD player, etc.). ) is suitably filtered by the low-pass filter 42, and then set at an appropriate sampling frequency f.
At step s, the A/D converter 43 converts the signal into a digital signal X(n) and inputs it to the DSP 44.

The filter coefficient ROM 45 stores various coefficients necessary for the DSP 44 to function as a digital filter, and is read out and output to the DSP 44 as necessary. The work RAM 46 is a work memory when the DSP 44 operates as a digital filter, and stores data for filtering calculations and the A/D converter 43.
Digital input signal X(n) input from and DSP
The waveform signals and the like that have been processed in step 44 are stored.

The DSP 44 performs arithmetic processing using the coefficients stored in the filter coefficient ROM 45 and the work RAM 46 to perform digital filtering processing and envelope extraction processing.

The processing result of the DSP 44 is sent to the CPU 2, and the CPU 2 determines the envelope value of the same note name for the scale note whose envelope value exceeds a predetermined value among the scale notes detected by the scale note detection device 5. , and extract the chord structure. Further, the CPU 2 converts this extracted chord structure into R
Generates corresponding chord data by comparing it with the chord structure stored in OM3, converts a pre-prepared accompaniment pattern into a scale change pattern based on the generated chord data, and uses this scale tone to generate musical sounds. They are sequentially assigned to the sound generation modules of circuit 9. Therefore, the CPU 2 functions as an integrating means, a chord data generating means, and an accompaniment sound generating means.

The keyboard 6 is provided with function switches, keys, etc., and when the switches, melody keys, etc. are operated on the keyboard 6, the CPU 2
detects this operation and assigns the generated musical tone to the generated tone module of the musical tone generating circuit 9.

Display 7 and printer 8 are CP
It operates under the control of U2, and displays one or more musical scales detected by the pitch data detection device 5, and also prints them on paper. For example, the CPU may display the scale included in the sound being input on the display 7 in real time, or it may display the scale included in the sound being input on the display 7 in non-real time as a musical score after undergoing editing work, or it may display it on the display 7 as a musical score after editing, or it may print it on the printer 8 It may be printed on.

This scale note detection device 5 as a whole detects the frequency spectrum of the frequency corresponding to each scale by sequentially performing digital filtering processing with different characteristics on a digital waveform signal representing an acoustic signal in a time-sharing manner. It functions as a digital signal processing means for detecting an envelope value, and a scale detecting means for detecting one or more scale tones included in the acoustic signal based on the processing result of this digital filtering.

As the musical tone generator 9, various types of tone generator circuits can be applied, such as PCM type, F
Sound source generation circuits such as the M method, the iPD method, and the sine wave synthesis method are applied. This musical sound generating device 9 has a plurality of musical sound generating channels, for example, 4 channels, and by driving the audio system 10, the musical sound generating device 9 transmits the scale tones of the scale numbers assigned to the sound generation module of the CPU 2 to the speaker 11. The sound is output via the .

A normal audio system is used as the audio system 10, and the audio system 10 receives not only signals from the musical tone generating circuit 9 but also
Signals from the microphone 41 and line input LINE IN are also given, and output as acoustic output as required. Further, the musical tone generation circuit 9 can generate a musical tone signal having a tone according to the tone specified by the keyboard 6, and in this case, the CPU 2
assigns the scale tones to be output to the sound generation module and performs musical sound generation operations. Furthermore, the CPU 2 controls the scale detection device 5.
It is also possible to sequentially store changes in the scale notes detected by the controller as sequencer information in the RAM 4, and read out the sequencer information sequentially in response to a play start instruction from the keyboard 6, etc., to generate a corresponding musical tone signal from the musical tone generating circuit 9. It is possible.

<Configuration of DSP> FIG. 15 is a circuit diagram of the DSP 44. The DSP 44 functions as a digital signal processing means and a scale detection means by performing digital filtering processing and envelope extraction processing. The DSP 44 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, a flag register 448, and the like.

The interface 441 is connected to the CPU 2 and the A/D converter 43 shown in FIG. 14 via a bus, and receives audio input signals and commands from the CPU 2 through the interface 441, and also receives processing results. Pitch data etc. are output.

The operation ROM 442 contains programs as digital signal processing means and scale detection means used in the chord discrimination device and automatic accompaniment device using the same according to the inventions of the present application, specifically, a digital filtering processing program and an envelope extraction program. Operation ROM 442 stores processing programs, etc.
The program outputs the program contents sequentially to the decoder 444 according to the address designation of the address counter 443, and also to each section.

The decoder 444 performs operation RO
The program contents read from M442 are decoded and output as control signals to each part of DSP44. DS
The filter coefficient ROM 45 and work RAM 46 of the scale detection device 5 are connected to the bus P44, and coefficient data, waveform signals, etc. are supplied to the DSP 44 as appropriate according to the program of the operation ROM 442, and waveform signals processed by the DSP 44 are also supplied to the bus P44. is work RAM46
is output and written to.

Multiplier 445 multiplies input data and outputs the result to adder/subtractor 446, register group 447, and the like. The adder/subtractor 446 performs addition or subtraction processing on the input data, and sends the operation result to the multiplier 445 or work RA via the register group 447.
At the same time, the code data of the operation result is output to the flag register 448.

The flag data in the flag register 448 is output to the address counter 443, and the program content to be output from the operation ROM 442 is determined by the flag data in the flag register 448 to the address counter 443. That is, judgment processing is performed based on flag data in the flag register 448.

Next, the operation will be explained. <Scale Detection Process> First, the scale detection process in the scale detection device 5 will be explained. The scale detection process is performed under the control of the CPU 2, and is performed by the digital filter process described in the above-mentioned improvement principle.

[0082] First, as shown in FIG.
At the start of the scale note detection process, initial processing is performed (step S1). This initial process is mainly a process for clearing the work RAM 46.

When the initial processing is completed, it is checked whether the digital conversion of the acoustic signal to the digital signal X(n) by the A/D converter 43 is completed (step S2), and the digital conversion is performed by the A/D converter When the process is completed, the digital signal X input from the A/D converter 43
(n) are sequentially addressed 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 start ends), unlimited input signals (digital signals
(n)).

Digital signal X(n
) is stored, the CPU 2 causes the DSP 44 to operate as an FIR high-pass filter H1(z) (step S4). The operation processing as this FIR high-pass filter H1(z) is performed by the operation ROM 442 of the DSP 44.
The address setting of the program is performed using the address counter 443.
This is done by setting the coefficients of the filter coefficient ROM 45 for the high-pass filter H1(z).

The calculation for this high-pass filter H1(z) is based on the above equation (1), and in addition to the current input digital signal X(n), the previous input digital signal input X(n- 1), X(n-
2) is read out and executed using the multiplier 445 and adder/subtractor 446 in the DSP 44.

When the arithmetic processing as the high-pass filter H1(z) is completed, the setting value t for filtering each scale is set to the initial setting value t=1 (step S5), and the low-pass filter H2t(z) A filtering operation is performed as follows (step S6). The calculation for this low-pass filter H2t(z) is performed using the above formula (
2), each coefficient CY, 2r cos θ, r2
While reading out the filter coefficient ROM 45, the DSP
This is executed using a multiplier 445 and an adder/subtractor 446 in the 44. This calculation result Wt (n) is also the work RA
Another specific area of M46 is used as a ring buffer for sequential storage. In this case as well, by using the work RAM 46 as a ring buffer, the calculation results Wt (n-1), Wt
(n-2) can be read one after another and used for calculation.

[0087] When the arithmetic processing of the high-pass filter H1(z) and the low-pass filter H2t(z) is completed, next, envelope extraction processing for each scale is performed. This envelope extraction process is performed using the DSP for each scale.
44 as an IIR low-pass filter HE(z) (step S7).

The calculation for this low-pass filter HE(z) is based on the above equation (3), and each coefficient CE, 1 .
8, -0.81 from the filter coefficient ROM 45, the multiplier 445 and the adder/subtractor 44 in the DSP 44
Do this using 6. Among these operations, absolute value calculation | W
t(n)| is also executed using the adder/subtractor 446.

This calculation result Et(n) is also stored in the work RAM.
By using another specific area 46 of 46 as a ring buffer and sequentially storing it, the calculation results Et(n-1) and Et(n
-2) can be read out one after another and used for calculations.

[0090] When the above filtering process and envelope extraction process are completed, it is checked (t=N) whether these processes have been performed for all scales (step S8).
), and if the processing has not been completed for all scales, the set value t is incremented (step S9).
, the process moves to step S6. When the process moves to step S6, the filtering processes of step S6 and step S7 are executed again.

[0091] In step S8, if the filter processing etc. have been completed for all scales, the envelope value Et(n) (t=1 to N) for each scale is notified to the CPU 2 (step S10). , it is checked whether to end the scale detection processing mode (step S11). If the scale detection processing mode has not ended, the process moves to step S2, waits for A/D conversion of the next acoustic signal, and performs the same processing. In other words, the DSP 44 performs three systems of digital filtering in a time-division manner and repeatedly for each scale for each sampling, thereby calculating the frequency related to the frequency corresponding to each scale in real time according to the envelope of each scale. Spectral levels can be detected.

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

<Chord discrimination processing of CPU2> The CPU1
As mentioned above, the DSP 44 outputs the envelope value Et(n) (t=1 to N
), that is, the envelope value Et(n) of the frequency spectrum for the frequency corresponding to each scale is given, and can be used for various purposes. In each invention of the present application,
It identifies chords from each given scale and checks whether the identified chords follow a regular chord code. When a chord follows a chord code, the chord is converted into chord data and an accompaniment based on the chord is performed.

The chord discrimination process will be explained below. C
As shown in FIG. 17, the PU2 checks whether the detection of the scale note has been completed (step P1), and when the detection of the scale note has been completed, the PU2 receives the detected scale number Tn and its envelope value from the DSP 44. Et(n) is read (step P2). The completion of this scale note detection is checked by checking whether a scale note detection end signal from the DSP 44 is input.

The CPU 2 transmits the scale number Tn read from the DSP 44 and its envelope value Et(n) to the CPU 2.
The envelope value Et(n) of each scale number Tn stored in a register or the like in the DSP 44 and read from the DSP 44 is compared with a preset threshold value (Step P3). When the envelope value Et(n) does not reach the threshold, the envelope value Et(n) of the scale number Tn is
0" (step P4), and checks whether comparison processing has been performed for diatonic scale numbers Tn (step P5). If the processing has not been completed for all scale numbers Tn, the value n of scale number Tn is set to 1. (step P6), return to step P2, and similarly calculate the envelope value Et(
Compare n) with a threshold. In step P3, if the envelope value Et(n) is larger than a predetermined threshold, it is checked whether processing for all scale numbers Tn has been completed without changing the envelope value Et(n) (step P5). When the comparison process of the envelope values Et(n) is completed for all scale numbers Tn,
It is checked whether the detection process has been performed a preset predetermined number of times (step P7), and if the preset predetermined number of detection processes have not been performed, the process returns to step P1 and waits for the DSP 44 to finish detecting the scale. , the envelope value Et(n) of the detected scale note is checked. In this way, the scale tones detected by the DSP 44 are selected for those whose envelope value Et(n) is larger than a preset threshold, and by comparing the scale tones detected a predetermined number of times with this threshold, chords can be created. Select only the scale notes that have been input reliably as
Check that it has been input as a chord.

In step P7, when the detection process is completed a predetermined number of times, regardless of the octave difference, for example, the scale C2
, C3, the envelope value Et(n) is accumulated for each same note name in the order of the above-mentioned predetermined number of detection processes, so that the scale is C.

[Extra 1] is obtained (step P8).

Four envelope values Et(n) of Ec to Eb resulting from the above integration are selected in descending order (step P9), and accompaniment processing, which will be described below, is performed on the selected scale numbers Tn (step P10). In this embodiment, the detection process is performed a predetermined number of times, but the detection process may be performed only once. Furthermore, although the detected scale does not take into account the octave difference, it may be detected for each scale number Tn.

<Accompaniment processing> As shown in FIG. 18, the CPU 2 determines the envelope value Et(n
) are selected in descending order of scale number Tn, the selected scale number Tn is compared with the chord table stored in the ROM 3 in advance, and the scale data of the selected scale number Tn is determined from the code table. It is checked whether the code matches the code registered in (Step T1). The chord table stored in the ROM 3 employs a series of regular chords that are normally used as accompaniment. In step T1, if the selected scale data does not match the chord table, it is determined that the input acoustic signal is not a regular chord, and the process proceeds to the next accompaniment process without accompaniment processing.

In step T1, when the selected scale data matches the chord table, it is determined that the input acoustic signal is a regular chord, and the chord data is read out and stored in a register or the like. .

Next, it is checked whether the accompaniment flag BF is set (BF=1 or not) (step T2
). This accompaniment flag BF is set (BF=1) when accompaniment is being performed. Assuming that accompaniment is not currently being performed, the accompaniment flag BF is not set, so first, the accompaniment flag BF is set (BF=1) (
Step T3), and a rhythm counter is started (Step T4).

This rhythm counter is a counter that counts addresses for reading accompaniment patterns prepared in advance, and as shown in FIG. 19, counting is performed by interrupt processing. That is, step T in FIG.
When the rhythm counter is started by the interrupt processing in step 4, the rhythm counter starts incrementing (step U1) and checks whether the count value is the maximum value (MAX) (step U2). If the count value is not the maximum value, the process returns and continues counting at a predetermined count timing. When the count value is the maximum value in step U2, the rhythm counter is reset, the count value is returned to 0 (step U3), and counting is continued in the same manner.

Returning to FIG. 18 again, when the rhythm counter is started, the chord type (for example, major chord, minor chord, seventh chord, etc.) and root of the chord are determined from the chord data read in step T1. The sound is stored in a register or the like (step T5), the count value of the rhythm counter is read, and rhythm pattern data is obtained from the ROM 3 from the count value of the rhythm counter (step T6). Once the chord type, its root note, and rhythm pattern are determined, the scale of the rhythm pattern is converted to a scale corresponding to the chord (step T7), and the scale-converted scale is sent to the sound generation channel of the musical sound generation circuit 9. Output (step T8). Musical sound generation circuit 9
produces the accompaniment sound set to the sound generation channel.

[0103] In this manner, scale tones are detected from the input acoustic signal by digital processing, and four tones with large envelope values Et(n) are selected from the detected scale tones. Check whether the selected scale note corresponds to a regular chord, and if it corresponds to a regular chord, the chord type,
The root note and rhythm pattern are determined, and the accompaniment note is produced. Therefore, it is possible to quickly and appropriately determine whether the input acoustic signal has an appropriate chord structure, and when the input acoustic signal has an appropriate chord structure, the corresponding chord data can be generated. It is possible to generate accompaniment sounds corresponding to chord data.

[0104]

According to the invention as claimed in claim 1, it is possible to quickly and appropriately determine whether an acoustic signal generated from a musical instrument such as a piano, an electronic musical instrument, etc. has an appropriate chord structure. In addition, when the chord structure is appropriate, corresponding chord data can be generated. Therefore, chord determination can be performed in a short time and efficiently, and more accurate chord determination can be performed.

According to the second aspect of the invention, it is possible to quickly and appropriately determine whether an acoustic signal generated from a musical instrument such as a piano or an electronic musical instrument has an appropriate chord structure. When the chord structure is appropriate, it is possible to generate corresponding chord data and perform accompaniment using the generated chords.

[0106] Therefore, it is possible to perform chord determination efficiently in a short time, and to perform automatic accompaniment based on the determined chord.

According to the third aspect of the invention, since the chord structure and the chord data corresponding to this chord structure are stored in the chord data storage means in the form of a data table,
It is possible to determine whether an input acoustic signal has an appropriate chord structure, and to generate corresponding chord data more appropriately and quickly when the input acoustic signal has an appropriate chord structure. , can also be accompanied by appropriate chords.

According to the invention described in claims 4 to 7, all signal processing can be performed in the digital domain,
Scale extraction processing can be performed even faster and more efficiently.

According to the invention as set forth in claim 8, the envelope values of the detected scales are integrated a predetermined number of times, and the scale tones constituting the chord are selected in descending order of the integrated values. Erroneous detection of chords due to pitch fluctuations can be prevented. Further, even if each scale is input with a time difference, chord data can be generated accurately.

Furthermore, according to the ninth aspect of the present invention, the scale note detected by the scale detecting means is assigned to one of the 12 scales regardless of the octave difference, so that one of the scales constituting the chord is assigned. It also becomes possible to detect inverted chords that are composed of a scale one octave higher.

[Brief explanation of drawings]

FIG. 1 is a configuration diagram showing the basic principle of an embodiment of a chord discrimination device and an automatic accompaniment device using the same according to the inventions of the present application.

FIG. 2 is a frequency characteristic diagram of the bandpass filter Ht(z) in FIG. 1.

FIG. 3 is a configuration diagram based on an improved principle of FIG. 1;

FIG. 4 is a frequency characteristic diagram when the high-pass filter H1(z) and low-pass filter H2t(z) in FIG. 3 are connected in cascade.

FIG. 5 is a configuration diagram of the high-pass filter H1(z) in FIG. 3.

6 is a frequency characteristic diagram of the high-pass filter H1(z) in FIG. 5. FIG.

FIG. 7 is a configuration diagram of the low-pass filter H2t(z) in FIG. 3;

8 is a diagram showing poles, zeros, pole vectors, and zero vectors of the digital filter in FIG. 7; FIG.

FIG. 9 is a diagram showing frequency characteristics corresponding to FIG. 8;

10 is a frequency characteristic diagram of the low-pass filter H2t(z) in FIG. 7. FIG.

FIG. 11 is a configuration diagram of the low-pass filter HE(z) in FIG. 3;

FIG. 12 is a frequency characteristic diagram of the low-pass filter HE(z) in FIG. 11.

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

FIG. 14 is an overall circuit configuration diagram of an embodiment of a chord discrimination device and an automatic accompaniment device using the same according to each invention of the present application.

15 is an internal circuit configuration diagram of the DSP 44 in FIG. 14. FIG.

16 is a flowchart of digital filter processing and envelope extraction processing by the DSP 44 of FIG. 15. FIG.

FIG. 17 is a flowchart showing a process of extracting candidate scale tones that are chords from among the scale tones detected by the DSP.

FIG. 18 is a flowchart showing chord discrimination processing.

FIG. 19 is a flowchart showing counting processing of a rhythm counter for detecting a chord rhythm pattern.

[Explanation of symbols]

1 Musical sound generator 2 CPU 3 ROM 4 RAM 5 Scale detection device 6 Keyboard 7 Display 8 Printer 9 Musical sound generation circuit 10 Audio system 11 Speaker 12 Bus 44 DSP 45 Filter coefficient ROM 46 Work RAM

[Outside 2]

Claims (9)

[Claims] 1. A digital signal processing means for detecting the envelope value of a frequency spectrum regarding a frequency corresponding to each musical scale by sequentially performing digital filtering processing with different characteristics on a digital waveform signal representing an acoustic signal in a time-sharing manner. a scale detection means for detecting one or more scale tones included in the acoustic signal based on the processing result of digital filtering by the digital signal processing means; and a chord data storage means for storing chord data corresponding to a chord structure. and chord data generation means for comparing the chord structure of the scale tones detected by the scale detection means with the chord structure of the chord data storage means to generate corresponding chord data. Chord discrimination device. 2. Digital signal processing means for detecting envelope values of frequency spectra regarding frequencies corresponding to each musical scale by sequentially performing digital filtering processing with different characteristics on a digital waveform signal representing an acoustic signal in a time-division manner. a scale detection means for detecting one or more scale tones included in the acoustic signal based on the processing result of digital filtering by the digital signal processing means; and a chord data storage means for storing chord data corresponding to a chord structure. a chord data generating means for generating corresponding chord data by comparing a chord structure based on the scale tones detected by the scale detecting means with a chord structure in the chord data storage means; and a predetermined accompaniment pattern. and an accompaniment sound generation means for generating an accompaniment sound by converting changes in the accompaniment pattern into changes in scale notes based on the chord data generated from the input of the chord data generation means. automatic accompaniment device. 3. The chord data storage means stores, in the form of a data table, a chord configuration consisting of a plurality of scales and chord data corresponding to the chord configuration and consisting of a root note and chord type. The chord discrimination device according to claim 1 or 2, or an automatic accompaniment device using the same. 4. The digital signal processing means is characterized in that the digital signal processing means sequentially executes a bandpass filtering process using a frequency corresponding to each scale as a center frequency in a time-division manner. 3. The chord discrimination device according to 3 or an automatic accompaniment device using the same. 5. The digital signal processing means sequentially time-divisionally executes a bandpass filtering process with the center frequency being a frequency corresponding to each of the scales, and generates an envelope from a waveform signal obtained as a result of the bandpass filtering process. The chord discrimination device according to claim 1, 2 or 3, wherein an envelope value of a frequency spectrum regarding a frequency corresponding to each of the scales is detected by sequentially performing a signal processing operation to extract the chord. An automatic accompaniment device using it. 6. The digital signal processing means performs high-pass filtering processing with predetermined characteristics, and sequentially executes low-pass filtering processing in which resonance having a peak at a frequency corresponding to each scale is added in a time-sharing manner. A chord discrimination device according to claim 1, claim 2, or claim 3, or an automatic accompaniment device using the same. 7. The digital signal processing means performs high-pass filtering processing with predetermined characteristics, and sequentially performs low-pass filtering processing to which resonance with a peak is added at frequencies corresponding to each of the scales in a time-sharing manner, 2. Further, by sequentially performing signal processing operations for extracting envelopes from waveform signals obtained as a result of these digital filtering processes, envelope values of frequency spectra regarding frequencies corresponding to each of the scales are detected. , the chord discrimination device according to claim 2 or claim 3, or an automatic accompaniment device using the same. 8. The scale detecting means further includes an integrating means for integrating the envelope value of the same note name a predetermined number of times for the detected scale notes whose envelope value exceeds a predetermined reference value. The chord discriminating device according to claim 1 or 2, or the chord discriminating device according to claim 2, characterized in that a plurality of scale tones are selected in descending order of the envelope values accumulated by the accumulating means and supplied to the chord data generating means. Automatic accompaniment device used. 9. The scale detecting means further assigns the detected scale note to one of the 12 scales regardless of the octave difference thereof.
The chord discrimination device described above or an automatic accompaniment device using the same.