JPS6017497A - Ensemble sound source device - Google Patents

Abstract

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

Description

DETAILED DESCRIPTION OF THE INVENTION Field of the Invention The present invention relates to a sound source device for an electronic musical instrument, and more particularly to an ensample sound source device capable of producing musical tones with a natural unsampled feel.

Conventional configurations and their problems In general, the timbre of multiple instruments playing the same melody, such as the first or second violin in an orchestra, is
This is called an unsampled effect or an unsampled sound source. Conventionally, to create such sounds, multiple delay time modulators using variable delay elements such as bucket brigade devices (BBDs) were installed in parallel and a single violin musical tone signal was applied to them. , by adding ultra-low frequency modulation signals with different phases to each modulator, each modulator produces violin sounds with different pitches, and by mixing these output signals, an unsampling effect is created. I was trying to make it happen. However, it was difficult to say that the effect was necessarily similar to that of an actual violin ensample.

Another conventional example would be to simply create a violin sound based on square waves or sawtooth waves output from one or more oscillators and mix them together, but this would require a large number of oscillators. It is uneconomical.

OBJECTS OF THE INVENTION The present invention solves these problems and provides an unsampled sound source device that can produce sounds very close to the natural unsampled feel with a simple configuration. A scale information generator that generates information that is consistent with each other; an address counter that divides the clock signal according to the output of the scale information generator and outputs an address code; and a musical tone signal. A memory that stores sample values of a waveform and from which the sample values are repeatedly read out according to the address code, and a digital-to-analog converter that converts the output of the memory into an analog signal, the memory having different fundamental frequencies. A sample that is a mixture of multiple musical tone signals using a series of sample values cut from a sample value of a signal waveform with a multiple period of several times or more the period determined by the above fundamental frequency. By storing the values, musical tones with a natural unsampled feel can be read out from the memory.

Description of examples FIG. 1 is a block diagram of one embodiment of the present invention.

In FIG. 1, reference numeral 1 denotes a scale information generator that generates information that determines the pitch of a sound, that is, a musical scale, using an operating device such as a keyboard. A digital pitch code DK representing the pitch is output. 2
is the clock frequency f from a clock generator (not shown)
The address code DA is obtained by dividing the clock signal having the address code DA.
This is a programmable address counter that outputs R.

Reference numeral 3 denotes a memory that stores waveform information of the sound source, and a digital code S representing a sample value of the waveform written in the address specified by the address code ADR is repeatedly read out at a period corresponding to the musical scale. Let this be a waveform sample s.

The waveform sample S is converted into an analog waveform signal by a digital-to-analog converter (not shown) and becomes a musical tone signal.

When the memory 3 stores 1024 samples of waveform sample values in 1 word of 12 bits, the memory 3 is 12x1024ζ
12.288Kbit is required. Further, address code ADRij can be expressed by a 10-bit code.

Conventionally, the number of samples of musical sound waveforms stored in the memory 3 is about 64 to 128, and one period corresponding to the fundamental frequency of the musical sound signal is assigned to these samples, and this is repeatedly read out. On the other hand, in this embodiment, a plurality of periods are cut out and stored as the period with respect to the fundamental frequency of the musical tone signal. The sample value is not a musical tone signal of a single musical instrument, but a musical tone signal of a so-called simultaneous performance state in which a plurality of musical instruments are played at the same time. An example of this waveform is shown in FIG. When playing in unison, multiple instruments play the same melody, but the pitches are not exactly the same, resulting in a beat-like pattern.

When two musical instruments play in unison, the musical sound waveform repeats at the frequency difference, that is, the cycle of the beat frequency. For example, if the fundamental frequencies of two musical instruments are fl and T2, and the period is T1°T2, and the period ratio T1/T2 is expressed as the ratio m/n of integers m and n, then a wave with period T1 is n waves, and a wave with period T2 is The wave is the m wave, and the phases of both waves match again.

Therefore, if a waveform for a time length nT (=mT2) is stored as a sample value, a two-tone beat-shaped musical tone signal is completely stored.

When actually cutting out the waveform, it is necessary to pay attention to the fine structure of the waveform so that the first part a and the last part of the cut are smoothly connected, as illustrated in FIG.

Furthermore, in general, when a plurality of tones are slightly different from each other, it is difficult to find a clear repetition period as described above, but in this case, it is possible to successfully connect them by using the following method.

The first method is to record the sounds of each instrument separately, and when playing and mixing them, finely increase or decrease the playback speed of those individual signals to closely match the beginning of the section to be cut out. In this method, the end of the section is defined as the waveform obtained, the individual signals are mixed, and the sample between the start and end sections is sampled and stored in the memory 3.

The second method is to cut out the desired section when playing in unison,
Modulate the beginning part so that the volume gradually increases from zero, modulate the ending part so that the volume gradually decreases, and add the sounds of the beginning and end parts to create an endless sound.
In this method, by cutting near the middle of the above-mentioned section, sample values with good waveform continuity can be obtained even when repeatedly read out. FIG. 4 is a conceptual diagram of the second method.

When using the section A-F from the signal played in unison, the volume of the rising part A-B is modulated to gradually increase as shown in Figure 4 (,), and the volume of the falling part is reversed. Modulate to gradually decrease. Then, overlap the sections A to B and E-F as shown in Figure 4), cut at points C and D, and
Let C be the starting point and C be the ending point.

Then, sampling is performed between D and 0. In this way, A-B and E-F will be connected smoothly, and point D and 0
Since points are originally continuous parts, they naturally connect smoothly.

FIG. 6 further gives the impression of a smooth ensemble. FIG. 6 [present]) is a continuous signal of -periods created as shown in FIG. 4(b). On the other hand, in Figure 6 (C
) is the same type of musical instrument, for example, a violin ensemble, and the other time parts are connected smoothly as shown in Figure 4), but the amplitude is modulated and mixed. The point is T2. Figures 6(d) and (e) are also violin ensembles, but they are mixed at T3 and T4.

Based on these four different sound source waveforms, continuous signals of - frequency mixed at four different times are mixed to create the signal shown in FIG. 6 (-). In this way, amplitude modulation and mixing almost always occur, and since there are four types of ensembles, an even smoother ensemble sound can be obtained. It is better that the times T1 to T4 are not at constant time intervals. Also, the slope of modulation per time is
Better change it.

It would be better if the tones and pitches of each of the four ensembles were slightly different. It is also possible to combine completely different instruments.

For the rise of the sound, create and memorize the waveform of the rise as shown in Figure 6, and write it as K - + D - + C-).
It may be read as D→C...

Also, D −+ (1! −+ ]) −+ C...
You can also read out ``...'' and add an envelope at the beginning of the sound.

This is known as a conventional method of applying envelopes for musical tone generation.

These processes may be performed on the analog signal, or may be performed once the signal is digitized. Of course, A-B and E- in FIGS. 4(a) and (b)
F processing methods are amplitude modulation and delay modulation.

This may be performed by a method such as spectral conversion.

The sample values obtained as described above are stored in the memory 3.

The address counter 2 includes, for example, a programmable divider that performs variable frequency division according to the pitch code DK;
It can be configured by cascading a counter with the bit length of the address.

In this case, by making the read clock of the memory 3 variable, musical tones of a predetermined pitch can be generated. As for octave switching, it may be realized by interlaced reading of 1 sample, 3 samples, etc., or it may be a method of accumulating pitch codes DK corresponding to frequencies and using the accumulated result as address code ADR. .

Note that the signal stored in the memory 3 may be a natural musical instrument sound such as a violet, an electronic sound such as a music synthesizer, or a human voice.

Effects of the Invention As described above, the present invention stores sample values of a signal waveform obtained by mixing a plurality of musical tone signals having different fundamental frequencies in a memory, and stores sample values of multiple periods, which are several times or more of the period determined by the fundamental frequency. It stores a series of cut sample values and repeatedly reads out address codes according to the musical scale, so it has a very simple configuration.
It is possible to read out a musical tone signal that closely resembles a natural unsampled feeling.

In particular, in the present invention, a mixture of a plurality of waveforms with different sound joints is used, so that a smoother ensemble effect can be achieved.

[Brief Description of the Drawings] Figure 1 is a block diagram of an embodiment of the present invention, Figures 2 to 6
The figure shows the signal processing method of the above embodiment, and FIG. 6 is a waveform diagram including the rising edge of the sound. 1... Scale information generator, 2... Address counter, 3... Memory. Name of agent: Patent attorney Toshio Nakao and 1 other person No. 1
Figure 3 Figure 4 EF Figure 5 Figure 6

Claims (1)

[Scope of Claims] (1) A scale information generator that generates information for determining a scale using an operating device such as a keyboard, and a clock signal that is frequency-divided according to the output of the scale information generator to output an address code. a memory that stores sample values of musical tone signal waveforms and from which the sample values are repeatedly read out according to the address code; and a digital-to-analog converter that converts the output of the memory into an analog signal; Using a series of sample values cut from sample values of a signal waveform obtained by mixing a plurality of musical tone signals having different fundamental frequencies, the sample values having a multiple period of several times or more the period determined by the above fundamental frequency. By storing sample values that are a mixture of these,
An unsampled sound source device characterized in that a musical tone having a natural unsampled feeling is read out from the memory. (2. In claim 1, the sample value is cut out so that the beginning and end of the sample values stored in the memory are smoothly connected, and the times at the joints are made to be different. Unsampled sound source device. (3) In claim 2, the starting part and the ending part of the sample value are made to substantially match in form so that the starting part and the ending part of the sample value are smoothly connected. An unsampled sound source device characterized by: