JP2776175B2 - Method of generating original tone signal used for synthesis of tone signal - Google Patents

Description

DETAILED DESCRIPTION OF THE INVENTION

[0001]

BACKGROUND OF THE INVENTION 1. Field of the Invention The present invention relates to a method for synthesizing a tone signal.

[0002]

2. Description of the Related Art In recent years, data obtained by digitizing a musical tone signal generated from an actual musical instrument or the like is stored in a memory, and the memory storing the data is used to generate a musical tone. Electronic musical instruments having such a configuration as provided in a sound source and other various devices have been widely used. In order for a device such as the above-described electronic musical instrument to perform a music performance close to a music performance of an actual musical instrument, data corresponding to a large number of timbres is required as the data. When a device such as an electronic musical instrument having data corresponding to timbres is configured, a large-capacity memory is required, so that the configuration of the device becomes large and the cost increases.

The above problem can be solved by, for example, efficiently encoding data for each tone color, compressing the amount of data, and then storing the data in a memory. According to Japanese Patent Application No. Hei 4-30076 "Phase prediction method of audio signal", the first and second respective Fourier transform frames are sequentially cut out from the audio signal so as to have a predetermined fixed time length. The same predetermined number of discrete frequency data obtained as a result of the Fourier transform for each of the first and second Fourier transform frames by performing a discrete Fourier transform on the Fourier transform frame using the same window function. Thus, for each of the first and second Fourier transform frames, phase information for each discrete frequency is obtained, and in each of the first and second Fourier transform frames, And determining the aspect of the change of the phase information for each of the same discrete frequencies, and assuming that the aspect of the change of the phase information for each of the discrete frequencies is constant on the time axis, the first time position Phase information of a predetermined number of discrete frequencies in a Fourier transform frame at a third time position existing at a time position that is an integral multiple of the time difference between the second time position and the third time position, A method for predicting phase information of a Fourier transform frame of a position has been proposed. By applying the above-described phase information prediction method, highly efficient encoding of an acoustic signal is performed, thereby transmitting (recording) an acoustic signal, musical instrument, and the like. The structure of the sound source and so on were simplified.

FIG. 4 is a block diagram of an encoder which can be used also as a means for compressing data for a musical sound source, which is one of applications of the above-described phase information prediction method proposed by the present applicant company. FIG. 5 is a block diagram of the decoder. FIG. 6 is also used to explain the configuration principle and the operation principle of the above-mentioned proposed phase information prediction method. The AU described in the uppermost part in FIG.
The waveform indicated by DIO is an audio signal targeted for high-efficiency encoding. The above acoustic signal in FIG.
.., which are shown in a certain portion of the display of “me Window”, are cut out so as to have a certain time length, and are sequentially formed as Fourier transform frames. A state in which each of the above-mentioned Fourier transform frames is multiplied by a window function so that each frame is successively overlapped with each other and gradually connected so that each of the frames has, for example, 1024 sample points from the acoustic signal. The Fourier transform frame for each one frame is subjected to an orthogonal transform by, for example, a Fourier transform (DFT) of a discrete finite sequence or a fast Fourier transform (FFT). It is. In the following description, it is described that the orthogonal transform is performed by a fast Fourier transform (FFT).

When an FFT operation is performed on a Fourier transform frame for each frame, the number of data (the number of samples) in the Fourier transform frame for each frame is N, and the sampling frequency is fs. , F
= Fs / N The FFT operation result for each discrete frequency (total N frequencies) for each frequency interval of f indicated by fs / N is obtained, but the FFT operation result is a real number for each discrete frequency. Part (Real) amplitude and the imaginary part (Ima
g) amplitude. Using the real part (Real) amplitude and the imaginary part (Imag) amplitude of each discrete frequency described above, by the following equations 1 and 2, for each discrete frequency described above, The synthetic amplitude term (Am
p) and a phase term (Phase) are obtained. By the way, FF
In N real part (Real) amplitudes and N imaginary part (Imag) amplitudes obtained for each discrete frequency as a result of the T operation, the N real part (Real) amplitudes have the same value. Two appear, and N imaginary parts (I
mag) Since two amplitudes having the same value appear in absolute value, the number of effective combined amplitude terms (Amp) becomes の of the total number, and the effective phase term (Phase) term Since the number is also と of the total number, the number of FFT output data is
It can be equal to the number of real input data of FT.

[0006]

(Equation 1)

(Equation 2)

[0007] Now, regarding a continuous Fourier transform frame (frame) continuous on the time axis, the real part (Real) amplitude and the imaginary part (Ima) of each discrete frequency are provided.
g) By using the amplitude and the equations (1) and (2), the composite amplitude term (Am
When p) and the phase term (Phase) are obtained, it is natural to predict that two adjacent frames on the time axis will have the same amplitude if the most common sense is taken. In fact, for example, if the sampling frequency is 44.1
Assuming that the number of samples is 1024 points at KHz and the frame length (the frame length is about 1/40 second), the spectrum obtained by performing the FFT operation on the signal of the sound of the piano also has a certain 1 Comparing the 512 amplitudes in one frame with the 512 amplitudes in the next frame of that frame or the 512 amplitudes in the next frame of the next frame, It has been confirmed that the amount of change between spectra is extremely small.

On the other hand, the fact that it is difficult to predict the phase in successive frames continuous on the time axis is that there is no relation between the repetition time of the successive frames and the frequency of the signal. It is clear from the beginning and end of the frame regardless of the frame, and it has been conventionally considered that signal prediction by orthogonal transform is difficult. Then, assuming that the sampling frequency is 44.1 KHz and the number of samples is 1024 as a frame length,
Even if the distribution of 512 phases in one frame based on the FFT operation result obtained by actually performing the FFT operation on the signal of the piano sound is seen, the distribution of the phases is random. 5 in the frame of
It turns out that the distribution of the 12 phases cannot be predicted.

Mr. Takahashi, the inventor of the above-mentioned “method of predicting the phase of an acoustic signal” by the present applicant, has stated that even if the phase information of one frame in a sequential frame on the time axis is used, the other Cannot predict the phase information of the two frames. However, for two frames, the phase information between the same predetermined number of discrete frequency phase information obtained as a result of Fourier transform for each Fourier transform frame is calculated. Focusing on the fact that if the change mode is constant on the time axis, it is possible to predict the phase information of another frame by using the relationship, and as described above, "for each of the two frames, The change mode of the phase information between the phase information for each of the same predetermined number of discrete frequencies obtained as a result of the Fourier transform for each transform frame is one on the time axis. , Which is referred to as Takahashi's hypothesis, and in fact, a single frequency sine wave signal, a composite signal of multiple sine wave signals, a musical instrument (piano) sound signal, When experiments were performed using various signals such as the above, a correct prediction result was obtained to the extent that it was recognized that the phase of the frame predicted according to the above hypothesis and the phase of the actual frame were practically identical. Various experimental results confirm that Takahashi's hypothesis holds in practice.

According to Takahashi's hypothesis, N / 2 phase term data θi (1) obtained for each discrete frequency in frame 1 shown in FIG. The data θi (2) of N / 2 phase terms obtained for each discrete frequency and the data θi (3) of N / 2 phase terms obtained for each discrete frequency in frame 3, for example. ) And, for example, N / 2 phase term data θi (4) obtained for each discrete frequency in frame 4, the phase term data in frame 1 Is, for example, θ1
Also, if the data of the phase term in frame 2 is, for example, θ2, the data of the phase term in frame 3 is, for example, θ3, and the data of the phase term in frame 4 is, for example, θ4, ,
Since the amount of phase change between frames is substantially the same as θ2−θ1 ≒ θ3−θ2 ≒ θ4−θ3 ≒ Δθa, if this hypothesis holds, then two frames If the phase difference between the data of the phase terms of all the frequency values corresponding to each other in the N / 2 phase terms obtained for each discrete frequency is known, the above-mentioned two frames are obtained. It is possible to predict the phase of another different frame, and specifically,
As in the above example, when the data of the phase term of a specific frequency value fa in frame 1 is θ1, and the data of the phase term of a specific frequency value fa in frame 2 is θ2, The data θ3 of the phase term of a specific frequency value fa in the frame next to 2 is
.theta.3.noteq.2 + (. theta.2-.theta.1) = 2.theta.2-.theta.1... (a), and the above-described phase prediction is individually performed for all discrete frequencies in frames 1 and 2. Thus, the phase of the signal of frame 3 can be predicted.

According to the Takahashi hypothesis described above, even if the phase information of one frame, for example, the phase information of only frame 1 is known, it is impossible to predict the phase information of another frame using the phase information. However, if the phase information of two frames is known, it is possible to predict the phase information of another frame, and it is possible to predict the phase information of two adjacent frames, for example, the phase information of frame 1 and the phase information of frame 2. This indicates that it is possible to predict the phase information of frames other than the two frames described above, and that K is two times as long as the time length of one frame.
If the phase information of one frame, for example, frame 1 and the phase information of frame 4 are known, the phase information of another frame, for example, frame 7, which is separated from frame 4 by K times the time length of one frame is predicted. It is also possible to express the above-mentioned Takahashi's hypothesis in general, as follows: “First, first, second, and third in each sequential Fourier transform frame cut out from the audio signal so as to have a predetermined time length.
The same predetermined number of discrete numbers obtained as a result of the Fourier transform for each of the first and second Fourier transform frames by performing a Fourier transform discretely on each of the second Fourier transform frames using the same window function. From the data for each frequency, phase information for each discrete frequency is obtained for each of the first and second Fourier transform frames, and the phase information is obtained for each of the first and second Fourier transform frames. The manner of change of the phase information for each identical discrete frequency is determined, and the manner of change of the phase information for each discrete frequency is assumed to be constant on the time axis, and the first time position and the The individual phase information of a predetermined number of discrete frequencies in the Fourier transform frame at the third time position existing at a time position that is an integer multiple of the time difference from the second time position is determined. Fourier It can be to be able to predict the phase information of the conversion frame ".

The block diagram shown in FIG. 4 is based on the above-described acoustic signal phase prediction technology based on the Takahashi hypothesis.
When performing recording and transmission by compressing the information amount of a signal to be recorded and transmitted, the residual signal Ai (m) -Ai (m-1) of the amplitude and the residual of the phase for each frame. Difference signal Δθi
(m) = θi (m) − {2θi (m−1) −θi (m−2)}, showing a configuration example of an encoder on the transmission side (encoding side) configured to be able to record and transmit. FIG. 5 is a diagram showing a configuration example of a receiving side (decoding side) decoder. Each of the residual signals becomes zero when prediction is performed, but since there is usually a slight deviation from the predicted value, the residual signal rarely becomes zero, but the original signal has a small value. The information amount of the residual signal is much smaller than the information amount.

In FIG. 4, reference numeral 1 denotes an input terminal of a digital audio signal to be recorded and transmitted. The digital audio signal supplied to the input terminal 1 of the digital audio signal is overlapped by a block 2 by a block 2. The sequential Fourier transform frames extracted so as to have a predetermined fixed time length in the state as described above are multiplied by a window function for each period having, for example, N sample points, and each successive frame is obtained. After the window function is multiplied in block 3 so as to form a sequential one frame period in which the seams of are overlapped and connected gently, a fast Fourier transform operation (FFT operation) is performed in block 4. It is. As a result of the FFT operation, for each Fourier transform frame, the same constant frequency interval f ｛
The number of data in the Fourier transform frame for each frame The number of samples is N, the sampling frequency is fs, and f = fs /
N} for each of the N discrete frequencies
l) The data of the FFT operation result including the amplitude and the imaginary part (Imag) amplitude is obtained.

As described above, the data at each of the N discrete frequencies obtained as a result of the FFT operation is subjected to signal processing by a different signal processing device for each of the discrete frequency data. However, FIG. 4 illustrates only the configuration of one of the N signal processing devices. In FIG. 4, the signal processing device is a component between the multiplexer 6 and the block 6 which is displayed as a rectangular coordinate → polar coordinate conversion.
Data of a specific discrete frequency including a real part and an imaginary part for each of N discrete frequencies obtained as a result of the FFT operation is subjected to polar coordinate conversion by a rectangular coordinate → polar coordinate conversion unit 6 to obtain an amplitude term and After being separated into the phase term, the calculation of the amplitude according to the above-described equation (1) and the calculation of the phase according to the above-described equation (2) are performed. For each frequency, the composite amplitude term Ai (m)
And the phase term θi (m) are obtained.

The synthesized amplitude term Ai (m) of a specific discrete frequency obtained as a result of the calculation performed by the rectangular coordinate → polar coordinate converter 6 is supplied to a latch circuit 7, a subtractor 8, and a data selector 9. The data selector 9 selects one of the setting data based on the composite amplitude term Ai (m) and the residual data output from the subtractor 8 according to the switching signal supplied to the terminal 34. And output it to the multiplexer 18. The switching operation of the data selector 9 is performed in conjunction with the switching operation of the data selector 16 described later. The phase term θi (m) of a specific discrete frequency obtained as a result of the calculation performed by the orthogonal coordinate → polar coordinate conversion unit 6 is determined by the latch circuit 10 and the subtractor 14.
And the data selector 16. As a result of calculating the phase of a specific discrete frequency (here, supposed to be fa) in the m-th frame, a rectangular coordinate → polar coordinate converter 6
Data of the phase θi (m) output from the latch circuit 1
The data of the phase held in the latch circuit 10 before being held at 0, that is, the calculation result of the phase of the specific discrete frequency fa in the (m-1) -th frame is output from the rectangular coordinate → polar coordinate converter 6. Phase θi (m-1)
Is stored in the latch circuit 12 in the phase prediction unit PFC.
Is held. The phase predicting unit PFC includes a latch circuit 12, an amplifier 11 having a gain of 2, and a subtractor 13 in the illustrated configuration example.

The data of the phase held in the latch circuit 12 before the data of the phase θi (m−1) is held in the latch circuit 12, that is, a specific discrete data in the (m−2) -th frame. The phase θ output from the rectangular coordinate → polar coordinate converter 6 as the calculation result of the phase of the frequency fa
The data of i (m−2) is supplied to the subtracter 13 as a subtraction signal, and the data of the i (m−2) is supplied to the subtracter 13 as a subtrahend signal. , The data of the predicted phase output from the subtractor 13, that is, the phase prediction unit PF
The predicted phase data output from C is 2θi (m-1) −
θi (m−2). The phase θ of a specific discrete frequency obtained as a result of the calculation performed by the rectangular coordinate → polar coordinate converter 6
i (m) is supplied to the latch circuit 10, the subtractor 13, and the data selector 16. The data of the phase θi (m) output from the rectangular coordinate → polar coordinate converter 6 as the calculation result of the phase of the specific discrete frequency (supposed to be fa) in the m-th frame is held in the latch circuit 10. The data of the phase held in the latch circuit 10 before the latching, that is, the calculation result of the phase of the specific discrete frequency fa in the (m-1) -th frame is used as a result of the orthogonal coordinate → polar coordinate conversion unit 6.
The data of the phase θi (m−1) output from the PFC is held in the latch circuit 12 in the phase prediction unit PFC composed of the latch circuit 12, the amplifier 11 having a gain of 2, and the subtractor 13. The terminal 15 is a supply terminal for the shift clock signal.

The data of the phase held in the latch circuit 12 before the data of the phase θi (m−1) is held in the latch circuit 12, that is, a specific discrete data in the (m−2) th frame. The phase θ output from the rectangular coordinate → polar coordinate converter 6 as the calculation result of the phase of the frequency fa
The data of i (m−2) is supplied to the subtractor 13 as a decrement signal, and the data of i (m−2) is supplied to the subtractor 13 as a subtrahend signal. , The data of the predicted phase output from the subtractor 13, that is, the phase prediction unit PF
The predicted phase data output from C is 2θi (m-1) −
θi (m−2). Predicted phase data 2θi (m-1) -θi (m-2) output from the phase prediction unit PFC
Is subtracted from the actual phase data θi (m) by the subtractor 14, so that the phase residual signal Δθi (m) = θi (m) − {2θi (m−1) − θ
i (m−2)}, which is supplied to the quantization scaling 17 via the data selector 16, where the quantization size is set according to the frequency, and then supplied to the multiplexer 18.

In the multiplexer 18, the amplitude residual signal ΔAi (m) having a specific discrete frequency (hereinafter, supposed to be fa) and the phase residual signal Δθi (m) having a specific discrete frequency fa are used. Is supplied to the output terminal 19.
It should be noted that the output terminal 19 has at least once an original amplitude component Ai (m), a phase component θi (m),
Needless to say, it is supplied. All other signal processing circuits that generate the amplitude residual signal ΔAi (m) and the phase residual signal Δθi (m) for each of a predetermined number of discrete frequencies in the Fourier transform frame are provided in the multiplexer 18. Since the output data is also supplied from the multiplexer 18, the audio signal data in a state where the amount of information is compressed is output from the multiplexer 18 and transmitted to the transmission line via the output terminal 19.

Next, an example configuration of the receiving side (decoding side) of the transmission system will be described with reference to FIG. In FIG.
Reference numeral 20 denotes an input terminal of a decoder provided on the receiving side,
The input terminal 20 receives compressed audio signal data transmitted from the transmission side to the reception side via a transmission path (not shown) from the transmission side described with reference to FIG. That is, a demultiplexer (not shown) is used to convert the audio signal data including the amplitude residual signal ΔAi (m) for each of a predetermined number of discrete frequencies in the Fourier transform frame and the phase residual signal. , A signal including an amplitude residual signal ΔAi (m) and a phase residual signal Δθi (m) for each specific discrete frequency.
The amplitude residual signal ΔAi (m) and the phase residual signal Δθi (m) for each specific discrete frequency supplied to the input terminal 20 described above.
Is subjected to predetermined signal processing by a signal processing device that performs signal processing on data of a specific discrete frequency. FIG. 5 exemplarily shows one signal processing device that performs signal processing on data of a specific discrete frequency.

The input terminal 20 includes an original amplitude component Ai (m) and phase component θi (m) of a specific discrete frequency, an amplitude residual signal ΔAi (m), a phase residual signal Δθi (m), and the like. In the signal processing circuit supplied with the signal composed of the following components, the original amplitude component A of a specific discrete frequency (here, supposed to be fa) is specified by the demultiplexer 21.
i (m), the phase component θi (m), and the amplitude residual signal ΔA
i (m) and the phase residual signal Δθi (m) are separated and supplied to the amplitude component signal processing circuit and the phase component processing circuit. In FIG. 4, the data supplied from the latch circuit 24 to the adder 25 is the amplitude residual signal ΔAi (m) supplied from the demultiplexer 21 to the adder 25.
Is the amplitude residual signal ΔAi (m) in the m-th frame.
In the case of, since the data is the data Ai (m-1) of the combined amplitude term of the (m-1) th frame, the adder 25 calculates the data Ai (m-1) of the combined amplitude term of the (m-1) th frame. And the amplitude residual signal ΔAi (m) in the m-th frame are added, and the adder 25 outputs data Ai (m) of the synthesized amplitude term of the m-th frame, which is
4 and supplied to the polar → rectangular coordinate converter 27 via the data selector 26.

Further, as described above, the demultiplexer 2
1 at a particular discrete frequency (now supposed to be fa
The original amplitude component Ai (m), the phase component θi (m), and the phase component θi (m) and the phase residual in the amplitude residual signal ΔAi (m) and the phase residual signal Δθi (m). The difference signal Δθi (m) is supplied to the adder 29 and the data selector 30 after being requantized by the requantizer 22. The fact that the data output from the adder 29 becomes the data θi (m) of the phase term of the specific discrete frequency in the m-th frame means that the two data added by the adder 29 are , The phase residual signal Δ provided to the adder 29 from the demultiplexer 21.
θi (m) is the phase residual signal Δ in the m-th frame
θi (m) and 2θi output from the phase prediction unit PFC
This is because (m-1) -θi (m-2). The data θi (m) of the phase term of the specific discrete frequency in the m-th frame output from the adder 29 is supplied to the polar coordinate → rectangular coordinate conversion unit 27 via the data selector 30. It is supplied to a latch circuit 28. The terminal 23 is supplied with a shift clock signal.

The phase θ output from the adder 29
Before the data of i (m) is held in the latch circuit 28,
The phase data held in the latch circuit 28, that is, the specific discrete frequency f in the (m-1) -th frame
The data of the phase θi (m−1) of “a” is held in the latch circuit 12 in the phase prediction unit PFC composed of the latch circuit 12, the amplifier 11 having a gain of 2, and the subtractor 13. The data of the phase θi (m−1) is stored in the latch circuit 1
2, the data of the phase θi (m−2) of the specific discrete frequency fa in the (m−2) -th frame is stored in the subtractor 13. Is supplied as a subtraction signal to the subtractor 13, and is supplied to the subtracter 13 as a subtrahend signal because the output from the amplifier 11 having the gain of 2 is provided from the subtractor 13. The predicted phase data output, that is, the predicted phase data θi (m) output from the phase prediction unit PFC is 2θi (m−1) −θi (m
-2). Therefore, the prediction phase data θi (m) = 2θi (m−1) − output from the phase prediction unit PFC.
θi (m−2) and the phase residual signal Δθi of the m-th frame
When (m) = θi (m) − {2θi (m−1) −θi (m−2)} is added by the adder 29, the adder 29 outputs a specific discrete signal in the m-th frame. The data θi (m) of the phase term of the frequency fa is output.

As described above, m output from adder 25
The data Ai (m) of the combined amplitude term of the n-th frame and the data θi (m) of the phase term having a specific discrete frequency in the m-th frame output from the adder 29 are each a predetermined data selector. The polar coordinate → rectangular coordinate conversion unit 27 supplies the data Ai (m) of the composite amplitude term of the m-th frame and the adder 29 The real part (Real) amplitude and the imaginary part (Imag) at the above-mentioned specific discrete frequency fa by the data θi (m) of the specific discrete frequency phase term in the m-th frame output from The amplitude is calculated and output, and is supplied to the inverse FFT operation unit 31. Since the output data from all the signal processing circuits provided for each of the predetermined number of discrete frequencies in the Fourier transform frame is supplied to the inverse FFT operation unit 31, the inverse FFT operation unit 31 Is restored to the original digital tone signal by applying the window function multiplication shown in block 32 and the overlap addition shown in block 33. Output terminal 35
Will be sent to

[0024]

As described above, data obtained by digitizing a tone signal generated from an actual musical instrument or the like is stored in a memory, and the memory storing the data is stored in a memory. In an electronic musical instrument having a configuration such as that provided in a sound source for generating a musical tone, and other devices, if a musical tone signal to be stored in a memory has a substantially uniform amplitude, a memory The data amount of the amplitude component in the tone signal to be stored in the tone signal can be made extremely small, and the phase component in the tone signal described above is the same as that of the previously proposed "phase prediction method of acoustic signal" of the present applicant. As described above, the same window is used for each of the first and second Fourier transform frames in each of the sequential Fourier transform frames cut out from the audio signal so as to have a predetermined fixed time length. Fourier transform is performed discretely using numbers, and the first first data is obtained by the same predetermined number of discrete frequency data obtained as a result of the Fourier transform for each of the first and second Fourier transform frames. , Phase information for each discrete frequency for each second Fourier transform frame, and phase information for the same discrete frequency corresponding to each other in the first and second Fourier transform frames. The mode of change of the phase information for each discrete frequency is constant on the time axis,
That is, a predetermined number in the Fourier transform frame of the third time position existing at a time position that is an integral multiple of the time difference between the first time position and the second time position, assuming that it is in the form of an equal term. Means for determining the individual phase information of the discrete frequency and predicting the phase information of the Fourier-transformed frame at the third time position "for compressing the data amount of the phase component in the musical tone signal. Can be made smaller,
Therefore, it is possible to easily configure a sound source for generating a musical sound in the memory storing the data.

However, data obtained by digitizing a tone signal generated from an actual musical instrument or the like is stored in a memory, and the memory storing the data is provided in a sound source for generating a tone. When a continuous tone signal is generated using the tone signal data to be stored in the memory in the electronic musical instrument having such a configuration, the tone signal actually synthesized is represented by (b) in FIG. In some cases, such as the swell of the Fourier transform frame as in the waveform illustrated in FIG. 1), a solution to the problem has been sought.

[0026]

SUMMARY OF THE INVENTION The present invention discretely Fourier-transforms a signal of each sequential Fourier transform frame cut out from a tone signal so as to have a predetermined fixed time length by using the same window function. And converting the amplitude component and the phase component for each discrete frequency obtained using the same predetermined number of data for each discrete frequency obtained as a result of the Fourier transform for each Fourier transform frame into a high-efficiency code. And
A method of generating an original tone signal used in a tone signal synthesizing method in which a Fourier inverse transform is performed after inverse quantization to synthesize a tone signal, wherein the original tone signal is generated in correspondence with a tone generated from an instrument. In the synthesis of a tone signal in which the original tone signal is obtained by simultaneously adding together signals for each time length that is a predetermined integer multiple of the fundamental period of the tone signal in the stationary part of the tone signal and arithmetically averaging the signals, A method for generating an original musical tone signal to be used is provided.

[0027]

The time length of a predetermined integral multiple of the fundamental period of the tone signal in a stationary portion of the tone signal generated corresponding to the tone generated by the musical instrument (at least required for performing phase prediction) N time signals (time length of two or more Fourier transform frames) is sequentially stored in the memory. After simultaneously adding the N signals stored in the memory, arithmetic averaging is performed to obtain an original musical tone signal, which is used for synthesizing the musical tone signal.

[0028]

BRIEF DESCRIPTION OF THE DRAWINGS FIG. 1 is a block diagram of a method for synthesizing a musical tone signal according to the present invention. FIG. 1 is a block diagram showing an example of a configuration of an original sound signal generating apparatus which implements an original musical sound signal generating method used for synthesizing a musical sound signal according to the present invention, and FIG. 2 is an original musical sound used for synthesizing a musical sound signal according to the present invention. FIG. 3 is a diagram illustrating a waveform example for explaining a signal generation method, FIG. 3 is a diagram used for describing an operation, FIG. 4 is a block diagram illustrating a configuration example of an encoder used for a proposed transmission method of an acoustic signal, FIG. 5 is a block diagram showing an example of the configuration of a decoder used in the previously proposed method of transmitting an audio signal, and FIG. 6 is a diagram for explaining the configuration principle and operation of the previously proposed method of transmitting an audio signal. FIG. 1 is a block diagram showing an example of a configuration of an original sound signal generating apparatus which implements a method of generating an original musical sound signal used for synthesizing a musical sound signal according to the present invention. In FIG. (Microphone),
The musical tone generated (radiated) from an unillustrated musical instrument is converted into an electric signal (tone signal). The tone signal output from the sound-to-electric converter 36 is amplified by an amplifier 37, converted to a digital signal by an analog / digital converter 38, and sequentially written to a memory 39.

Since the fundamental period of the tone signal written in the memory 39 as described above is known, the tone signal written in the memory is divided by a fixed length equal to an integral multiple of the fundamental period. It is easy to read out from the memory 39 in a state where the tone signal is divided on the time axis. Each individual latch circuit 40 (1), 4
0 (2), 40 (3)... 40 (N). FIG. 2A exemplifies a tone signal written in the memory 39, and a section T in FIG. 2 shows a basic period of the tone signal written in the memory 39. 2 (b) and 2 (c)
.. Are signals obtained by dividing a musical tone signal that is continuous on the time axis into predetermined constant lengths as described above. , 40 (N), the signals stored in the individual latch circuits 40 (1), 40 (2), 40 (3),..., 40 (N). Each of the aforementioned latch circuits 40 (1),
The N signals latched at 40 (2), 40 (3)... 40 (N) are added by an adder circuit 41 and then divided by 1 by a divider circuit.
/ N, which is output to the output terminal 43 as an original tone signal used for synthesis of tone signals.

Since the original tone signal transmitted from the output terminal 43 as described above is more likely to be phase predicted than the tone signal used to generate it, see, for example, FIG. Using the encoder having the above-described configuration, data obtained by performing high-efficiency encoding of a musical tone signal is stored in a memory of a tone generator of the electronic musical instrument, and the tone signal read out from the memory of the tone generator of the electronic musical instrument. A tone signal synthesized by generating a sustained tone signal using the data shown in FIG. 3A has a waveform like a swell in the unit of a Fourier transform frame as shown in the waveform illustrated in FIG. The result is a tone signal of good waveform that has not been performed.

[0031]

As is apparent from the above description, the method of generating an original tone signal used for synthesizing a tone signal according to the present invention has a predetermined length of time. The same predetermined number of discrete frequencies obtained as a result of the Fourier transform for each Fourier transform frame are obtained by discretely Fourier transforming the signals of the respective Fourier transform frames cut out from using the same window function. A method of synthesizing a tone signal in which the amplitude component and the phase component of each discrete frequency obtained using the data of each of the discrete frequencies are encoded with high efficiency, and the inverse quantization is performed and the inverse Fourier transform is performed to synthesize the tone signal. When generating the original tone signal used in the above, the basics of the tone signal in the stationary part of the tone signal generated corresponding to the tone generated from the musical instrument Since the original musical tone signal is obtained by adding the signals for each time length of a predetermined integral multiple of the period at the same time and then performing arithmetic averaging, according to the present invention, as shown in FIG. When a tone signal having a good waveform can be restored, and the tone signal data obtained by the conventional method is stored in the tone generator memory of the electronic musical instrument, and a continuous tone signal is generated by the tone signal data. It is also possible to prevent a tone signal actually synthesized from generating a tone signal having a swell in the unit of a Fourier transform frame, such as the waveform illustrated in FIG. 3B.

[Brief description of the drawings]

FIG. 1 is a block diagram showing an example of the configuration of an original sound signal generating apparatus that implements an original musical sound signal generating method used for synthesizing a musical sound signal according to the present invention.

FIG. 2 is a waveform example diagram for explaining a method of generating an original tone signal used for synthesizing a tone signal according to the present invention.

FIG. 3 is a diagram used for explaining the operation.

FIG. 4 is a block diagram illustrating a configuration example of an encoder used in a proposed transmission method of an audio signal.

FIG. 5 is a block diagram showing a configuration example of a decoder used in a previously proposed audio signal transmission method.

FIG. 6 is a diagram for explaining the configuration principle and operation of a previously proposed sound signal transmission method.

[Explanation of symbols]

1: input terminal, 6: rectangular coordinate to polar coordinate converter, 6A: amplitude calculator, 6P: phase calculator, 7, 10, 12, 24,
28, 40 (1) to 40 (N) ... Latch circuit, 8, 13, 14
... Subtractor, 11 Amplifier with gain of 2, 42, 43 Adaptive quantizer, 18 Multiplexer, 19 Output terminal, 2
0: input terminal of decoder, 21: demultiplexer,
29: adder, 27: polar coordinate → rectangular coordinate converter, 31:
Inverse FFT operation unit, 36: acoustic-electric converter, 37: amplifier, 38: analog / digital converter, 39: memory,
41 ... addition circuit, 42 ... division circuit,

Claims (1)

(57) [Claims] 1. A discrete Fourier transform is applied to signals of respective sequential Fourier transform frames cut out from a tone signal so as to have a predetermined fixed time length using the same window function. The amplitude component and the phase component for each discrete frequency obtained using the same predetermined number of data for each discrete frequency obtained as a result of the Fourier transform for each transformed frame are respectively highly efficiently coded, and after the inverse quantization, the Fourier transform is performed. A method for generating an original tone signal used in a tone signal synthesizing method in which a tone signal is synthesized by performing an inverse conversion, wherein a stationary tone in a tone signal generated corresponding to a tone generated from an instrument is used. A signal in which the original tone signal is obtained by simultaneously adding the signals for each time length that is a predetermined integer multiple of the fundamental period of the tone signal in the above-mentioned general portion and then performing arithmetic averaging. A method of generating an original musical sound signal used for synthesizing a sound signal.