JPH02137889A - Signal recording method - Google Patents

Abstract

PURPOSE:To obtain musical sound data from which nonintegral multiple sound components are removed, to reduce looping noises at the time of reproduction, and to perform looping smoothly by recording musical sound data on a recording medium through a comb-shaped filter. CONSTITUTION:A pitch detecting function block 12 detects the fundamental frequency f0 of an input signal such as a musical sound signal or an input digital signal corresponding to the analog signal. Then, the signal is filtered by the comb-shaped filter 14 whose passing bands are only the fundamental frequency band of the input analog signal and the frequency bands of its higher harmonic components to obtain an output analog or digital signal. The output analog or digital signal is recorded on the recording medium while controlled by a loop section detecting function block 16 so that only the fundamental frequency band of the output analog or digital signal and the frequency band of its higher harmonic components are the passing bands. Consequently, when components in the musical sound signal except noise components are attenuated and looping processing is performed, musical sound data wherein the noise components are attenuated are looped, so looping noises are suppressed.

Description

Detailed Description of the Invention [Field of Industrial Application] The present invention relates to a signal recording method for recording an analog signal such as a musical tone signal or a digital signal corresponding to the analog signal on a storage medium. ] The present invention supplies an input signal such as an analog signal such as a musical tone signal or a digital signal corresponding to the signal to a comb-shaped filter, passes only the vicinity of the fundamental frequency of the input signal and frequencies that are integral multiples of the fundamental frequency, and A signal recording method that reduces noise contained in an input signal by extracting an appropriate repetitive waveform section of an output signal and recording it on a storage medium, thereby suppressing the noise generated when the recorded waveform is repeatedly reproduced. This is what we propose.

[Conventional technology]

Generally, the sound sources used in electronic musical instruments, TV game machines, etc. are analog sound sources consisting of VCO, VCA, VCF, etc.
PSG (Programmable Sound Generator)
It is broadly classified into digital sound sources such as waveform ROM mW output type and so on. In recent years, as a type of digital sound source, sampler sound sources have become widely known, in which sound source data is sampled from raw fundamental tones and digitally processed and stored in a memory or the like (for example, (See JP-A-61-264099 and JP-A-62-267798).

Since this sampler sound source requires a large capacity of memory for storing sound source data, various methods have been proposed to save memory, such as looping processing that utilizes the periodicity of musical sound waves, A typical example of this is bit compression processing using nonlinear quantization.

The looping process described above is also a method for continuing to output a sound for a longer time than the original duration of the sampled musical tone. For example, when considering a musical tone signal waveform,
In general, immediately after the start of sound production, non-pitch components such as piano key tapping noise and wind instrument press noise are included, resulting in a formant part with unclear waveform periodicity. The same waveform will appear repeatedly with a fundamental period corresponding to high (high).

By setting n cycles (n is an integer) of this repetitive waveform as a looping section and reproducing it repeatedly as necessary,
This means that you can obtain long-lasting sounds with a small amount of memory.

[Problem to be solved by the invention]

However, when the looping method is used for a sampled sound source as described above, the waveform subjected to looping processing becomes a completely periodic function, and cannot have any non-integer overtone components with respect to the loop frequency. and,
These non-integer overtones that the sound source originally had become looping noise and appear in the looping waveform.

This looping noise has frequency components on the spectrum and is not pleasing to the sense of hearing, so it must be removed as much as possible.

Furthermore, since the musical tone data sampled and stored in the memory is the actual musical tone that is digitally processed and recorded on a storage medium, the tone quality at the time of sampling also determines the tone quality at the time of reproduction. For example, if the sound quality at the time of sampling contains many noise components, the musical tone signal read from the storage medium and reproduced will also contain the noise components as is. In addition, if the musical tone to be sampled has so-called vibrato, it will be subject to minute FM modulation, so during the above looping process, the sidewave components generated by this FM modulation will be converted into non-integer harmonics. As a result, it is played back as looping noise.

Therefore, the present invention has been proposed to solve the above-mentioned drawbacks, and it is possible to obtain musical sound data from which non-integer overtone components have been removed by recording musical sound data on a recording medium through a comb filter. It is an object of the present invention to provide a signal recording method that reduces looping noise during playback and allows smooth looping.

[Means to solve the problem]

The signal recording method according to the present invention has been made to achieve the above-mentioned object, and as shown in the flowchart of FIG. The fundamental frequency f of the input signal, which is the input digital signal corresponding to the signal. (pitch information), and then, in step S2, the output analog signal or digital signal is filtered by a comb-shaped filter whose pass band is only the fundamental frequency band of the input analog signal and the frequency band of its harmonic components. At the same time, in step S3, only the fundamental frequency band of the output analogue or digital signal and the frequency band of its harmonic components become the passband (so that they are extracted).
This feature is characterized in that the information is controlled and recorded on a storage medium in step S4.

[For production]

By passing a musical tone through a comb filter that passes only the fundamental tone of the musical tone and its harmonics, components other than pitch components (non-pitched components and some noise components) in the musical tone signal are attenuated.
The N ratio can be improved, and when looping processing is performed, since musical tone data with noise components etc. attenuated is looped, looping noise can be suppressed.

〔Example〕

First, prior to a detailed explanation of the present invention, the looping process described above will be briefly explained with reference to the musical tone signal waveform shown in FIG. Generally, immediately after the start of sound generation, a formant part FR, which is a part where the periodicity of the waveform is unclear, occurs due to the inclusion of non-pitch components such as piano key tapping noise and wind instrument press noise, and after that,
The same waveform appears repeatedly with a fundamental period corresponding to the musical interval (pitch, pitch) of the musical tone. N periods (n is an integer) of this repetitive waveform are defined as a looping section LP, and this looping section LP is expressed between the looping points LP, the looping start point LP, and the looping end point LP. Then, the formant portion FR and the looping section LP are recorded on a storage medium, and during playback, the looping section LP is repeatedly played following the playback of the formant portion PR, thereby generating a musical tone over an arbitrary long period of time. can be done.

An embodiment of the present invention will be described below with reference to the drawings. It goes without saying that the present invention is not limited to the following examples.

FIG. 3 is a functional block diagram showing a specific example of each function from sampling an input musical sound signal to recording it on a storage medium when applying the sound source data compression encoding method of the embodiment of the present invention to a sound source data forming device. It is a diagram. Input terminal 1 in this case
As the input musical sound signal supplied to 0, for example, a signal directly picked up by a microphone, or a signal obtained by reproducing a digital audio signal recording medium, etc. can be used in the form of an analog signal or a digital signal. .

First, in the sampling processing function block 11 of FIG. 3, the input musical tone signal is sampled at a frequency of 38 kHz, for example, and extracted as digital data of 16 bits per sample.

This sampling process corresponds to A/D conversion process when the input musical tone signal is an analog signal,
Furthermore, when the input signal is a digital signal, it corresponds to processing of sampling rate conversion and bit number conversion.

Next, in the pitch detection function block 12, the fundamental tone frequency (
fundamental frequency) fo, that is, pitch information is detected.

The detection principle in this pitch detection function block 12 will be explained. Here, the musical sound signal that is the sampling sound source is
In many cases, the fundamental frequency is considerably lower than the sampling frequency fs, and it is difficult to identify the pitch with high precision just by detecting the peak of the musical tone on the frequency axis. It is necessary to utilize the spectrum of components.

First, let f(t) be the waveform of a musical tone signal whose pitch is to be detected, then convert this musical waveform f(t) to the amplitude a(
ω) and phase φ(ω), the musical sound waveform f (t
) is a Fourier expanded formula, f(t) = Σa (ω)cos (ωt +φ(
It can be expressed as ω))...■. Here, the phase shift of each overtone φ
If (ω) is all zero, f(t) = Σa(ω)C6Sωt ・ ・ ・ ・
・ ・ ・It can be expressed by the formula of ■. In this way, the points that are integral multiples of the period of all harmonics that the phase alignment has, and 1
=0 point. This is nothing but the period of the fundamental tone.

Based on this principle, the pitch detection procedure will be explained using the functional block diagram shown in FIG. 4.

In FIG. 4, musical tone data is supplied from a real part data input terminal 31 and °"0" is supplied from an imaginary part data input terminal 32 to a fast Fourier transform (FFT) functional block 33.

Here, in the fast Fourier transform performed by the fast Fourier transform function block 33, the musical tone signal whose pitch is estimated is x(t), and the harmonic components included in the musical tone signal X(0) are aacos(2πf, lt+θ )......■, then X(() is ω x(t)=Σa, 1cos(2πfllt+θ)...
■Rewriting this in complex representation, O x (t) = (1/2)Σa, 1exp(jθ, 1
)exp(jω1lt) −−■n;−ω However, cosθ= (exp (jθ)+exp(−jθ))
/2−・■ was used. When this formula is Fourier transformed, ω X(a+)=r x(t)exp(-jωt) d
t・Σanexp(j a n) δ(ω−ωn) ・
・■n=−, where δ(ω−ω7) is a delta function.

The next functional block 34 calculates the norm (absolute value, that is, the square root of the sum of the squares of the real and imaginary parts) of the data after the fast Fourier transform, that is, the absolute value Y(ω) of X(ω). ), the phase component is canceled and becomes Y(ω)・[X(ω)X(ω)! "" = (1/2) all δ (ω-ω, l) ・ ・ ・
-■ This is done in order to match the phases of all the high frequency components of the musical tone data, and by setting the imaginary part to zero, the phase components can be made equal.

Next, this calculated norm is supplied to the fast Fourier transformation tA (corresponding to inverse FFT in this case) function block 36 as real part data, and the imaginary part data input terminal 35 is supplied with "'0".
", and performs inverse FFT to restore musical tone data. In other words, the above inverse Fourier transform is as follows: y(t)=(1/2π)S Y(ω)exp(-,
iωt) dt0ko1lSa, ICO3ωllt -...(:[(1j
). The restored musical tone data after the inverse Fourier transform is extracted as a waveform that can be expressed by a combination of cosine waves in which all high-frequency components are aligned in phase.

Thereafter, a peak detection function block 37 detects the peak of the restored sound source data. Here, the peak is the point at which the extreme values (peaks) of all the high frequency components of the musical tone data are defeated, and in the next function block 38, the detected peak values are divided by 1 (sorted) from the largest value. )
do. By measuring the period of the detected peak, the pitch of the musical tone signal can be determined.

FIG. 5 is for explaining the configuration for detecting the local maximum value (peak) of musical tone data in the peak detection function block 37 of FIG. 4. In FIG.

In this case, the musical tone data has many peaks (extreme values) of different values, and by finding the maximum value of the musical tone data and detecting its period, the pitch of the musical tone can be determined.

That is, in FIG. 5, the musical tone data string after inverse Fourier transform is supplied to an N-stage shift register 42 via an input terminal 41, and the registers a-N/□ . . . The signals are sequentially sent to the output terminal 43 via ao...aN/□. This N+1-stage shift register 42 acts as a window with a width of N+1 samples for the musical tone data string.
+1 samples are sent to the maximum value detection circuit 44 via the window. That is, the above musical tone data is first input to register a-N/2, and then sequentially transmitted to register aN/□, and is input to each register a-N/□...ao...
- The above-mentioned musical tone data of N+1 samples from aN/□ are sent to the maximum value detection circuit 44.

This maximum value detection circuit 44 detects the data of the register a0 as a peak value when the value of, for example, the central register a0 in the shift register 42 becomes the maximum among the data of the N+1 samples. , output terminal 45
It outputs more. In addition, the width N of the above window
+1 can be set arbitrarily.

Returning to FIG. 3, the envelope detection function block 13 obtains the so-called envelope waveform of the musical tone signal by performing envelope detection processing using the pitch information on the digital musical tone signal after the sampling process described above. ing.

For example, the waveform shown in Figure 6B is obtained by sequentially connecting the peak points of the musical tone signal waveform shown in Figure 6A, and the level (or volume) changes over time from immediately after the sound is produced. It represents change.

This envelope waveform is generally expressed by parameters such as ADSR (attack time/decay time/sustain level/release time). As an example of a musical sound signal, when considering a piano sound that is produced in response to a keystroke, the above attack time T is the time when a key on the keyboard is pressed (key-on) and the volume gradually increases to reach the target volume. The decay time TIl represents the time from the volume reached at the attack time T until reaching the next volume (for example, the volume of a sustained sound of a musical instrument), and the sustain level L It represents the volume of the sustained sound that is maintained until the key is released and the key is turned off, and the release time TII represents the time from when the key is turned off until the sound disappears. Note that the above-mentioned times TA, TD, and TI may also indicate the slope or rate of volume change. Furthermore, more envelope parameters may be used in addition to these four parameters.

Here, in the envelope detection function block 13, envelope waveform information represented by each parameter such as ADSR (attack time TA/decay time T, /sustain level L, /release time TI+) as described above is simultaneously detected. In order to extract the aforementioned formant portion with the attack waveform remaining, information indicating the overall decay rate of the signal waveform is obtained.

This decay rate information is, for example, as shown in FIG.
The waveform takes a reference value of "1" during the interval and then monotonically decays.

Here, an example of the configuration of the envelope detection function block 13 shown in FIG. 3 will be explained with reference to the functional block diagram shown in FIG. 8.

The principle of envelope detection is similar to envelope detection of so-called AM (amplitude modulation) signals. That is, the envelope is detected by considering the pitch of the musical tone signal as the frequency of the carrier of the AM signal. The envelope information is used when reproducing a musical tone, and the musical tone is formed based on the envelope information and pitch information.

For the musical tone data supplied to the input terminal 51 in FIG. 8, an absolute value of the peak value data of the musical tone is determined in an absolute value output function block 52. FIR this absolute value data
(finite impulse response) type digital filter function block 55.

Here, the FIR filter function block 55 acts as a low-pass filter, and the function block 5
By supplying the filter coefficients formed in step 4 to the FIR filter function block 55, the cutoff characteristic of the low-pass filter is determined.

Here, the filter characteristic is, for example, the characteristic shown in FIG. 9, and has zero points at the fundamental tone (frequency to) of the musical tone signal and the frequencies of its overtones. For example, from the musical tone signal shown in FIG. 6A, by attenuating the frequency of the first harmonic of the fundamental tone using the FIR filter, the musical tone signal shown in FIG.
Envelope information as shown in is detected. Note that the characteristics of the filter coefficients described above are shown by the following equation.

H(f) = k ・(sjn(πf/L))/f
−−・・■ro in this formula 0 is the fundamental frequency of the musical tone signal (
pitch).

Next, from the peak value data (sampling data) of the sampled musical tone signal mentioned above, the peak value data of the formant part FH signal shown in FIG. 2 and the peak value data of the signal in the looping section LP (loop data)
The process of generating .

In the first functional block 14 for generating the loop data, the peak value data of the sampled musical tone signal is divided (or multiplied by the reciprocal) by the data of the previously detected envelope waveform (FIG. 6B). By performing envelope correction, peak value data of a signal with a constant amplitude waveform as shown in FIG. 10 is obtained. By filtering (the peak value data of) this envelope-corrected signal, (the peak value data of) a signal in which components other than pitch components are attenuated or pitch components are relatively emphasized is obtained. Here, the pitch component refers to a frequency component that is an integral multiple of the fundamental frequency f0. Specifically, in order to remove low frequency components such as vibrato included in the envelope-corrected signal, it is passed through an HPF (bypass filter), and then the frequency characteristics as shown in the dashed line in FIG.
In other words, by passing through a comb-shaped filter whose passband frequency characteristic is an integral multiple of the fundamental frequency f, only the pitch components included in the HPF output signal are passed, and other non-pitch components and noise are filtered out. The noise component superimposed on the signal after passing through the comb filter 3IA is removed by attenuating the component and passing it through an LPF (low pass filter) as necessary.

That is, when considering a musical sound signal such as the sound of an instrument as the input signal, this musical sound signal usually has a constant pitch (pitch,
As shown by the solid line in Fig. 11, the frequency spectrum contains energy in the vicinity of the fundamental frequency r0 corresponding to the pitch of the musical note itself and in the vicinity of frequencies that are integral multiples thereof. A distribution is obtained in which .

On the other hand, it is known that general noise components have a −-like frequency distribution. Therefore, by passing the input musical tone signal through a comb-shaped filter having frequency characteristics as shown in the dashed line in FIG. 11, the fundamental frequency f0 of the musical tone signal is
Only frequency components (so-called pitch components) that are integral multiples of are passed through or emphasized, while other components (non-pitch components and part of noise) are attenuated, and as a result, the S/N ratio can be improved. Here, the frequency characteristic of the comb-shaped filter shown in the dashed line in FIG. 11 is expressed by the following equation. In this equation, fo is the fundamental frequency of the input signal (the frequency of the fundamental tone corresponding to the pitch), and N is the number of stages of the comb filter.

The musical tone signal whose noise components have been reduced in this way is sent to the repetitive waveform extraction circuit, and this repetitive waveform extraction circuit extracts an appropriate repeated waveform section such as the looping section LP in FIG. 2 described above. After that, it is sent to a storage medium such as a semiconductor memory and recorded. Since the musical tone signal data recorded on this storage medium has non-pitch components and some noise components attenuated, it is possible to reduce noise when repeatedly reproducing the above-mentioned repetitive waveform section, so-called looping start point. Can be done.

The frequency characteristics of the HPF, comb filter, and LPF are set based on the fundamental frequency f0, which is the pitch information detected by the pitch detection function block 12 in advance.

Next, the loop section detection function block 16 in FIG. 3 detects an appropriate repeating waveform section for the musical tone signal whose components other than the pitch components have been attenuated by the above-mentioned filter processing, thereby determining the looping starting point LP and the looping ending point. Set a looping point with point LP.

That is, the loop section detection function block 16 selects two looping points that are relatively separated by (an integral multiple of) the repetition period corresponding to the pitch (interval) of the musical tone signal, and the selection principle is explained below. Explain.

When performing looping processing on musical tone data, the looping interval must be an integral multiple of the fundamental frequency B (reciprocal of the frequency of the fundamental tone) of the musical tone signal. Therefore, if the pitch of the musical note is accurately identified, it can be determined easily.

That is, a looping interval is determined in advance, two points separated by the interval are taken out, and a looping point is set by evaluating the correlation or UQ characteristic of signal waveforms in the vicinity of the two points. As an example of this evaluation function, one that uses convolution (result, convolution) of samples of signal waveforms near each of the two points will be described. That is, the correlation or similarity of signal waveforms is evaluated by sequentially performing the above convolution operation on all sets of points. Here, the evaluation by the above-mentioned convolution is carried out by sequentially inputting the musical tone data to a shift register, for example, and inputting the musical tone data taken in by each register to a DSP, which will be described later.
(digital signal processing device), which calculates and outputs the above convolution. The set of two points for which the convolution obtained in this manner is maximum is defined as the looping start point Ps and the looping end point LP.

That is, in FIG. 12, the looping starting point LP,
Let the candidate point of the above looping starting point LP be a, and let the candidate point of the looping end point LP be 50, the candidate point a of the looping starting point LP,
The peak value data of a plurality of points, for example, 2N+1 points, in the vicinity before and after , are respectively a-s"+a-2+a-1
+aO+al+ az,” a,, % Same number of candidates before and after candidate point b0 of looping end point LPt (2N+1
b-m..., b-*, b
-+, bo, b+, bz, . . . bN, the evaluation function E (ao, bo) at this time can be determined by the following equation. This 0th equation is an equation for finding convolution centered on the point all+bo. Then, the set of candidate points aO+bo is sequentially changed to find the value of the evaluation function E for all looping point candidates, and the value is the maximum among all the evaluation functions E obtained. Let the point be the looping point.

Further, in addition to the method of finding the looping point from convolution as described above, it is also possible to find the looping point from the method of least squares of errors. That is, the candidate point a O+50 of the looping point by the least squares method is ε(aa, bo) '''Σ(a, -bk)t...
It can be expressed as a [phase] or uniaxial equation. In this case, aO+bO that minimizes the value of the evaluation function ε may be found.

Further, in the loop section detection function block 16 described above, the looping start point LP is determined as necessary.

The pitch conversion ratio is calculated based on the truching termination point LP. This pitch conversion ratio is determined by the following function block 17.
It is used as time axis correction value data during the time axis correction process. This time axis correction process is performed in order to align the pitches of various sound source data when actually recording the various sound source data in a storage means such as a memory, and is performed in order to align the pitches of the various sound source data when actually recording the various sound source data in a storage means such as a memory. The pitch information detected by the detection function block 12 may be used.

The pitch normalization operation in the time axis correction function block 17 will be explained with reference to FIG. 13.

FIG. 13A shows a tone signal waveform before time axis correction processing (mainly time axis companding process), and FIG. 13B shows a corrected waveform after the companding process. These Figures 13A and B
The time axis of is marked with a scale in units of blocks during quasi-instantaneous bit compression encoding processing, which will be described later.

In the waveform A before time axis correction, the looping section LP is normally unrelated to the above block, but the 13th
As shown in FIG. B, the time axis companding process is performed so that the looping section LP is an integral multiple (m times) of the block length (block period), and the block boundary position is the looping start point LP, and the looping end point LP, in the time axis direction. That is, the starting point LP and the ending point LP of the looping section LP.

Time axis correction (
time axis companding and shifting), an integer number (m
The looping process can be performed on a block-by-block basis, making it possible to normalize the pitch of sound source data during recording.

Here, the peak value data may be filled with "0'° between the deviation ΔT from the block boundary which occurs at the beginning of the musical tone signal waveform due to the time shift.

The 14th session shows the Pro-2 structure when the peak value data of the waveform after time axis correction is divided into blocks for the focus compression encoding process described later. The number (number of samples, number of words) is h. In this case, the above-mentioned pitch normalization means - In boat fishing, the number of words in the above-mentioned block is calculated by the number of words in the looping section LP for n periods of the waveform of the constant period Tw of the musical tone signal waveform shown in FIG. More preferably, the start point LP and the end point LP of the looping section LP are made to coincide with the block boundary position on the time axis. The process is to perform time axis processing (shift processing). In this way, each point LP
If s and LPi match the block boundary positions, errors caused by block switching during decoding in a bit compression encoding system can be reduced.

Here, the words WLPS and WLP in the shaded part in one block of FIG. This is a word indicating a sample of point). Note that if the above shift processing is not performed, the looping start point LPs and the looping end point LPt do not necessarily coincide with the block boundary, so as shown in FIG. 14B, the setting positions of the words WLP, WLPE, is set to any position. However, the above word WLP
The number of words between , and word WLP is an integral multiple (m times) of the number of words in one bronc, and the pitch is normalized.

Here, as mentioned above, various methods can be considered for time-axis companding processing of the musical tone signal waveform in order to make the number of words in the looping section LP an integral multiple of the number of words in one block.
For example, this can be realized by interpolating the peak value data of a sampled waveform, and as a specific example, a filter configuration for oversampling processing, etc. can be used.

By the way, the looping period of an actual musical sound waveform has a fraction of the sampling period unit, and the looping start point L
Sampling peak value at P s and looping termination point L
If there is a deviation from the sampling peak value at P, interpolation processing using oversampling etc.
A non-integer number of sampling cycles including interpolated samples is obtained by finding a peak value that matches the sampling peak value of the looping start point LPs at a position near the looping end point LP (a position at a distance shorter than the sampling cycle). It is conceivable to realize double the looping period (with a fraction). Such a looping period which is a non-integer multiple of the sampling period can also be made into an integer multiple of the block period by the above-described time axis correction processing. For example, when performing time-axis companding processing using 256 times oversampling, ,
The error in the peak value between the looping start point LP and the end point LP can be reduced to 1/256, thereby achieving smoother looping playback.

In the next functional block 21, the waveform whose looping section LP has been determined and which has been subjected to time axis correction (companding) processing as described above is converted into a looping section L as shown in FIG.
Loop data is generated by connecting P in front and behind each other. In other words, FIG. 15 shows the musical sound waveform (the first
A loop data waveform is shown in which only the looping section LP is cut out from FIG. 3B) and a plurality of these looping sections LP are arranged.
and the other looping start point LP are connected and arranged in sequence. This loop data waveform is generated by the loop data generation function block 21.

Since this loop data is formed by connecting looping sections LP many times, a word WLP corresponding to each looping start point LP of the connected loop data waveform is
Immediately before the start block containing 5, the looping end point L
The data of the ending bronc including the word WLP corresponding to PE (more precisely, the point immediately before point LP) will be placed as is. In principle, when performing bit compression encoding processing, it is sufficient that at least the end block exists immediately before the start block of the looping section LP to be stored. In other words, during bit compression encoding in block units, the parameters of the start block (information on bit compression encoding for each compressed block, such as range information and filter selection information described later) are the data of the start block and end block. It may be formed based on the following. This is a technique that can also be used when the sound source is a musical tone signal consisting only of loop data without a formant part, which will be described later.

By doing this, during the above encoding, the looping start point L
Regarding P and the terminal point LP, the same data is lined up over a plurality of samples before and after each of them. Therefore, the parameters for bit compression encoding for each point LP and each block immediately before LP are the same, and errors (noise) during looping playback during decoding processing can be reduced. can.

That is, the musical tone data that is looped and reproduced becomes stable without any connection noise. In this embodiment, the number of samples of data in the looping section LP placed immediately before the start process is approximately 500 samples.

Next, in the process of generating data for the signal of the formant portion FR, envelope correction processing is first performed in the functional block 18, similar to the functional block 14 in generating the loop data.

However, in this case, the envelope correction is performed by dividing the sampled musical tone signal by the envelope waveform (see Fig. 7) containing only the decay rate information mentioned above, resulting in a waveform as shown in Fig. 16. The signal (peak value data) is obtained.

That is, in the output signal of FIG. 16, the envelope of the attack portion (between time TAO) remains, and the other portions remain at a constant amplitude.

This envelope-corrected signal is subjected to filter processing in a functional block 19 as necessary. For filter processing in this functional block 19, the above-mentioned functional block 1
For example, a comb-shaped filter having a frequency characteristic as shown in the dashed line in FIG. 11 is used. In other words, this comb-shaped filter has a frequency characteristic that emphasizes frequency band components that are integral multiples of the fundamental frequency r corresponding to the pitch and relatively attenuates non-pitch components. The frequency characteristic is set based on the pitch information (fundamental frequency ".") detected by the pitch detection function block 12.Such a signal is used as sound source data that is finally recorded in a storage medium such as a memory. It is used to generate data for the formant part of the signal.

In the next functional block 20, the above functional block 1
The same time axis correction as in step 7 is also performed on the formant part generation signal. This is the function block 16 above.
The pitch conversion ratio found in or the function block 1 above
By compressing and expanding the time axis based on the pitch information detected in step 2, the pitches of each sound source can be made equal (normalized).

Next, in a functional block 22, the loop data and formant part generation data, both of which have been time-base corrected using the same pitch conversion ratio or pitch information, are mixed. In this mixing, a Hamming window is applied to the formant part generation signal from the functional block 20 to form a fade-out type signal that attenuates over time in the part to be mixed with the loop data. On the other hand, a similar Hamming window is applied to the loop data from the functional block 20, and in this case, a fade-in type signal that increases with time is formed in the portion to be mixed with the formant signal, and these Mix the signals of (
(crossfading) to obtain musical tone signals that ultimately become sound source data. Here, as the loop data to be recorded in a storage medium such as a memory, by extracting the data of one looping section that is a certain distance from the above-mentioned crossfade part, noise during looping playback can be eliminated.
looping noise) can be reduced. In this way, peak value data of the sound source signal is obtained, which consists of the formant portion FR, which is a waveform portion including non-pitch components from the time of sound generation, and the looping section LP, which is a repeating waveform portion containing only pitch components.

In addition, a process of cutting and connecting each part such that the start point of the loop data signal is connected to the position of the looping start point in the formant part generation signal may also be considered.

By the way, when actually performing loop section detection, looping processing, and even mixing loop data and formant parts, a rough mix is performed by repeating trial and error testing by human hands, and this Loop points (looping start point LPs and looping end point L)
P,) performs more accurate processing based on information etc.

That is, prior to detecting a loop section with high accuracy in the function block 16, loop section detection and the above-mentioned mixing etc. are performed manually while repeatedly listening to samples in the procedure shown in the flowchart of FIG. 17, and then the above-mentioned Highly accurate processing (from step S26 onward) is performed.

In FIG. 17, in the first step S21, the loop point is detected with relatively rough accuracy, for example, by using the zero-crossing point of the signal waveform or visually checking the display of the signal waveform, and in step S22. A looping process is performed to repeatedly reproduce the waveform between the loop points, and in the next step S23, a human listens to the sample and determines whether it is good or not.

If it is defective, the process returns to the first step S21 and the loop point is detected again. If a good listening result is obtained by repeating this process, the process proceeds to the next step S24, where it is mixed with the above-mentioned formant part signal by cross-fading, etc. In the next step S23, a human listens to the signal from the formant part to the looping part. Determine whether the transition is good or not. If the mixture is defective, the process returns to step S24 and the above mixing is repeated.

Thereafter, the process proceeds to step S26, where the loop section detection function block 16 performs highly accurate loop section detection. Specifically, loop section detection including the above-mentioned interpolated samples is performed, for example, in the case of 256 times oversampling, loop section detection is performed with an accuracy of 1/256 of the sampling period.
In the next step S27, a pitch conversion ratio for the pinch normalization is calculated. Based on this pitch conversion ratio, the time axis correction process in the function block 17.20 is performed in the next step 528, and loop data generation in the function block 21 is performed in the next step S29. and,
In step 330, the mixing process in the functional block 22 is performed. In the processing after step S26, the loop point information etc. obtained from steps S21 to 325 are used. Note that steps S21 to 325 may be omitted to fully automate the looping process and the like.

The formant part F obtained by such a mixing process
The peak value data of the signal consisting of R and the looping section LP is subjected to bit compression encoding processing in the next functional block 23.

(Left below) Various bit compression encoding methods can be considered as the above-mentioned bit compression encoding method.
The quasi-instantaneous companding type proposed in Japanese Patent Application No. 08629 and Japanese Patent Application Laid-Open No. 62003516, etc. is highly efficient, in which the peak value data is divided into blocks for each fixed number of words (h samples) and bit compression is applied in units of blocks. The high-efficiency via compression encoding method will be briefly described with reference to FIG. 18.

In FIG. 18, the high-efficiency bit compression encoding system is composed of an encoder 70 on the recording side and a decoder 90 on the reproduction side. data x (n)
is supplied.

This input signal (peak value data) x(n) is obtained by the predictor 7
The predicted signal (peak value data) x(n) from the predictor 72 is supplied to the FTP (finite impulse response type) digital filter 74, which is composed of 2 and an adder 73, and the predicted signal (peak value data) x(n) from the predictor 72 is sent to the adder 73 as a subtraction signal. being sent. In the adder 73, the predicted signal 7(n) is subtracted from the input signal x(n), thereby outputting a prediction error signal or a broadly defined difference output d(n). Predictor 72 generally calculates past p inputs x (n-pLx(
Predicted value by 1 frozen combination of n-p+IL...x(n-1)? (n) is calculated. Note that the FIR filter 74 is hereinafter referred to as an encode filter.

In the high-efficiency bit compression code system, data within a certain time period of the sound source data, that is, input data of a certain number of words, is divided into blocks, and the encoding filter 74 having the optimum characteristics is selected for each block. I try to do that. In this method, multiple (for example, four) encoding filters with different characteristics are provided in advance, and the filter with the optimal characteristics, that is, the one that can obtain the highest compression rate, is selected. This can be achieved by doing so. However, in terms of the configuration of the digital filter -a, a plurality of M (for example, 4 sets) of coefficient sets of the predictor 72 of one encode filter 74 shown in FIG. 18 are stored in a coefficient memory or the like.
By time-divisionally switching and selecting these sets of coefficients, one of the plurality of encoding filters is substantially selected.
This is often the equivalent of selecting one.

Next, the difference output d (n) as the prediction error is sent via an adder 81 to a bit compressor consisting of a shifter 75 with a gain G and a quantizer 76, and is sent to a bit compressor consisting of a shifter 75 with a gain G and a quantizer 76, for example, a floating point Compression processing or ranging processing is performed such that the exponent part in the display form corresponds to the gain G and the mantissa part corresponds to the output from the quantizer 76, respectively. That is, the shifter 75 shifts the input data by the number of bits corresponding to the gain G to switch the range, and the quantizer 76 performs requantization to extract a fixed number of bits from the bit-shifted data. There is. Here, the noise shaping circuit (noise shaper) 77 uses an adder 78 to obtain a so-called quantization error corresponding to the error between the output and the input of the quantizer 76, and applies this quantization error to a shifter with a gain G-'. 79 to the predictor 80, so-called error feedback is performed in which the predicted signal of the quantization error is fed back to the adder 81 as a subtraction signal.

In this way, requantization by the quantizer 76 and error feedback by the noise shaping circuit 77 are performed, and the output power (n) is taken out from the output terminal 82.

By the way, the output d'' (++) from the adder 81
is the above-mentioned difference output d (from n>, the above-mentioned noise shaper 7
Predicted signal of quantization error from 7? (n), and the output d°″(n
) is the gain G and the output d' (
n). Further, the output cuff (n) from the quantizer I6 is the quantization error e (n) in the quantization process and the output d'' (n) from the shifter 75.
The adder 78 of the noise shaper 77 extracts the quantization error e (n). This quantization error e (n) is the gain Q
The predicted signal 1(n) of the quantization error is obtained by passing through the -1 shifter 79 and the predictor 80 which takes the 1 frozen sum of one past input.

The sound source data is subjected to the encoding process as described above, and is outputted from the quantizer 76 as an output ↑(n) through the output terminal 82.

Next, the prediction/range adaptation circuit 84 outputs mode selection information as optimal filter selection information and sends it to the predictor 72 and output terminal 87 of the encoding filter 74, and also outputs the gain G and the gain G-1 or the range information for determining the above bit shift amount is output to each thick 75.79 and output terminal 8.
It has been sent to 6.

Next, the output ↑(tl) from the output terminal 82 of the encoder 70 is transmitted to the input terminal 91 of the decoder 90 on the reproduction side, or the signal 7'(n) obtained by recording and reproduction is transmitted. is supplied. This input signal”
a"' (n) is sent to the adder 93 via the shifter 92 with a gain of G-1. The output x' from the adder 93
(n) is sent to the predictor 94 and the predicted signal T' (n)
This predicted signal 'i' (n) is sent to the adder 93
and is added to the output ↑" (n) from the shifter 92. This added output becomes the decode output '5i" (n
) from the output terminal 95.

Further, each output terminal 86 and 87 of the encoder 70
The range information and mode selection signal output, transmitted, recorded, and reproduced are input to input terminals 96 and 97 of the decoder 90, respectively. Then, the range information from the input terminal 96 is sent to the shifter 9.
2 to determine the gain G-', and mode selection information from input terminal 97 is sent to the predictor 94 to determine prediction characteristics. The prediction characteristic of this predictor 94 is selected to be equal to the characteristic of the predictor 72 of the encoder 70.

In the decoder 90 having such a configuration, the shifter 9
The output from 2 ↑” (n) is the above input signal 7′ (n)
is multiplied by the gain G-1. Further, the output `(n) of the adder 93 is the sum of the output IT(n) from the shifter 92 and the prediction signal x'(n).

Next, FIG. 19 shows the bit compression encoding encoder 7.
This figure shows an example of the above one block of output data from 0. This one block of data consists of 1 byte of header information (parameter information related to compression or attached information) RF and 8 bytes of sample data DAO~ D113
It is made up of.

The header information RF includes the 4-bit range information,
The above mode selection information or filter selection information of 2 bits and two flag information of 1 bit each, for example, information Ll indicating the presence or absence of a loop, and information El indicating whether there is a terminal block (end block) of the waveform. It is made up of. Here, the peak value data of one sample is compressed and expressed in 4 bits, and the above data I) Ae
~D0 contains 16 samples of 4-bit data DAO
Contains N to D 13L.

Next, FIG. 20 shows each block of the quasi-instantaneous (blocked) bit compression encoded peak value data corresponding to the leading portion of the musical tone signal waveform as shown in FIG.

In this FIG. 20, the header is omitted and only the peak value data is shown, and for convenience of illustration, the l block is divided into 8
Although this is a sample, it goes without saying that it can be set arbitrarily, such as 16 samples per block. this is,
The same applies to the case shown in FIG. 14 above.

Here, the above-mentioned quasi-instantaneous bit compression code system has a mode in which the above-mentioned input musical tone signal is directly output, that is, a straight P
Select the mode in which a signal with the highest compression rate is obtained from the CM mode and the mode in which the musical tone signal is output through a filter, that is, the first-order or second-order differential filter mode, and then output the musical tone data as the output signal. It is designed to transmit.

When sampling a musical tone and recording it in a storage medium such as a memory, the waveform of the musical tone signal waveform of the musical tone starts to be captured at the sounding start point KS.
When a filter mode that requires an initial value, such as the first-order or second-order difference filter mode, is selected in the first block from S, it is necessary to prepare this initial value in advance. It is desirable to have a format that does not require an initial value. Therefore, in the period preceding the sound generation start point KS, the straight PCM mote (input, specifically, in FIG. All “′0
'' is arranged, and all data ``0'' from the beginning of this block are bit-compressed and taken in as sampling peak value data. This can be done, for example, by creating a block in advance in which all the data in one block is "'0'°" and storing it in a memory etc., or by using the sound generation starting point KS when sampling a musical tone. It can be obtained by starting sampling from the input signal of the part where all data is "0°" (that is, the silent part before the start of sound generation), and the number of blocks of the above pseudo input signal is at least one block. .

The H14m input input sound data formed as described above is subjected to signal compression processing using a high-efficiency bit compression encoding system as shown in FIG. and reproduce this compressed signal.

Therefore, when playing musical tone data including the above-mentioned pseudo input signal, the P-CM mode (stray) is selected for the filter at the start of playback (block portion of the pseudo input signal), so the first or second difference filter is There is no need to set initial values in advance.

Here, there is a concern about a delay in the sound generation start time due to the above-mentioned pseudo input signal (there is no sound because all the data is "O") at the start of playback. However, for example, when the sampling frequency is 32 kHz and one block has 16 samples, the delay in the sound generation time is about 0.55 seconds, which is not an audible delay and does not pose a problem.

By the way, the above-mentioned bit compression encoding process and other digital signal processing for generating sound source data are often implemented in software using a digital signal processing device (DSP). A software-like configuration using a DSP is often adopted for the reproduction of sound source data. As an example, FIG. 21 shows an audio processing unit (APU) 107 as a sound source unit that handles sound source data and its surroundings. An example of the overall system configuration including the following is shown.

In FIG. 21, a host computer 104 installed in, for example, a general personal computer device, a digital electronic musical instrument, a TV game machine, etc. is connected to the APUI07 as the sound source unit, and from the host computer 104, Sound source data etc. are APU1
It is designed to be loaded in 07. This APU10
7 is a CPtJ (central processing unit) 103 such as a microprocessor, and a DSP (digital signal processing unit) 1
01, and a memory 102 in which the above-mentioned sound source data and the like are stored.

That is, this memory 102 stores at least sound source data, and the DSPIOI performs various processing including readout control of the sound source data, such as looping processing, bit expansion (restoration) processing, pitch conversion processing, addition of an envelope, and echo processing. (Reverberation) processing etc. are applied.

The memory 102 is also used as a buffer memory for these various processes. The CPU 103' is a DSPI
It controls the operations and contents of these various processes of OI.

Furthermore, D is applied to the sound source data from the memory 102.
The digital musical sound data finally obtained by performing the above various processes using SPIOI can be stored on a digital/analog CD.
/A) The signal is converted into an analog signal by the converter 105 and supplied to the speaker 106.

Note that the present invention is not limited to the embodiments described above; for example, in the embodiments described above, the formant part and the looping section were connected to form the sound source data, but the sound source data consists only of the looping section. It can also be easily applied to the case of forming sound source data. Further, the decoder side configuration and the external memory for sound source data may be supplied as a ROM cartridge or an adapter, and can be applied not only to the sound source of musical tone signals but also to voice synthesis.

〔Effect of the invention〕

According to the present invention, by passing sound source data containing noise components through a comb filter, only the fundamental tone of a musical tone and frequencies that are integral multiples thereof can be extracted, thereby making it possible to cut noise components. Similarly, even if the sound source has minute FM modulation such as vibrato, looping noise can be removed by passing it through a comb filter.

[Brief explanation of the drawing]

FIG. 1 is a flowchart showing the principle of the signal recording method of the present invention, FIG. 2 is a musical tone signal waveform diagram, and FIG. 3 is a functional block diagram for explaining a specific example of the signal recording method of the present invention.
Fig. 4 is a functional block diagram for explaining pitch detection operation, Fig. 5 is a block diagram for explaining peak detection operation, Fig. 6 is a waveform diagram of musical tone signal and envelope, and Fig. 7 is a functional block diagram for explaining pitch detection operation.
The figure is a waveform diagram of the decay rate information of the musical tone signal, Figure 8 is a functional block diagram for explaining the envelope detection operation, Figure 9 is a characteristic diagram of the FIR filter, and Figure 10 is the envelope correction of the musical tone signal. A waveform diagram showing subsequent peak value data, Figure 11 is a characteristic diagram of the comb filter, Figure 12 is a waveform diagram to explain the operation of setting the optimal looping point, and Figure 13 is a musical tone signal before and after time axis correction. FIG. 14 is a schematic diagram showing the structure of a block for quasi-instantaneous bit compression for peak value data after time axis correction.
Figure 5 is a waveform diagram showing loop data obtained by repeatedly connecting the waveforms of the looping section, Figure 16 is a waveform diagram showing formant part generation data after envelope correction based on decay rate information, and Figure 17 is the actual waveform. A flowchart to explain the operation before and after the looping process,
FIG. 18 is a block circuit diagram showing a schematic configuration of a quasi-instantaneous bit compression encoding system, FIG. 19 is a schematic diagram showing a specific example of one block of data obtained by quasi-instantaneous bit compression encoding, and FIG. 21 is a schematic diagram showing the contents of a block at the beginning of a musical tone signal, and FIG. 21 is a block diagram showing an example of the configuration of a system including an audio processing unit (APU) and its surroundings.

Claims (1)

[Claims] An output analog signal or digital signal is obtained by supplying an input analog signal or an input digital signal corresponding to the analog signal to a comb-shaped filter whose pass band is only the fundamental frequency band of the input analog signal and the frequency band of its harmonic components. A signal recording method characterized in that a suitable repetitive waveform section of the output analog or digital signal is extracted and recorded on a storage medium.