JPH051957B2 - - Google Patents

Description

DETAILED DESCRIPTION OF THE INVENTION (Technical Field) The present invention relates to a CSM type speech synthesizer, that is, at most four
~ CSM (Composite) expressed by 6 wave frequencies
This technology relates to a speech synthesizer that synthesizes speech using a composite sine wave model (Sinusoidal Modeling).

(Prior Art) Conventionally, LPC-type speech synthesizers have been widely used as speech synthesizers, but LPC-type speech synthesizers generally have a complicated structure. Also used for speech synthesis
The characteristic of the LPC filter is that its stability is compromised due to errors during parameter transmission.

For this, we perform speech synthesis using CSM.
As will be described in detail later, the CSM type speech synthesizer does not have a filter and has a very simple structure, and essentially does not cause stability problems during synthesis.

However, in order to synthesize speech using CSM, which is expressed by at most 4 to 6 frequencies, simply linearly combining these is not sufficient.
In addition to this, some special processing is required. Currently, these processes are not clearly known to the general public, and in particular, the spread of the spectrum, the method of generating silence, the method of interpolating parameters, etc. when performing speech synthesis using CSM have not been established. Therefore, it is difficult to say that CSM type speech synthesizers have been put into practical use yet.

(Object of the invention) The object of the present invention is to solve the above-mentioned various problems when performing speech synthesis using CSM,
The objective is to provide a practical CSM type speech synthesizer.

(Structure of the Invention) The synthesizer of the present invention includes a plurality of variable frequency oscillators with a phase reset function set to each frequency specified by a CSM, and correspondingly outputs of each of the oscillators to each frequency specified by the CSM. The oscillator is equipped with a plurality of variable gain amplifiers, a variable length window function generator, and a random number generator set to The phase of each oscillator is reset in accordance with the period set with the distribution width and lower limit value calculated from random numbers, and the starting point generated by the variable length window function generator is 1 and the ending point is 0. The start and end points of the variable-length window function with constant continuity are made to substantially coincide with the phase reset point.

(Principle) First, we will explain the principle of the CSM type speech synthesizer.

CSM expresses an audio signal as the sum of a specific number of sine waves whose amplitude and frequency are freely selectable parameters. A predetermined number of 4 to 6 sine waves is used at most.

Therefore, when performing CSM speech synthesis, first,
The audio signal must be expressed as the sum of a predetermined number of sine waves using CSM audio analysis.
CSM voice analysis will be explained in detail later.
Only the main points will be explained here.

In CSM analysis, as in LPC analysis,
An autocorrelation coefficient is used as an intermediate parameter for the purpose of ignoring phase information, averaging the influence of sound sources, and avoiding instability due to noise components.

In other words, in CSM analysis, N low-order taps of sample autocorrelation coefficients directly calculated from the speech waveform to be expressed for each analysis frame are combined with N low-order taps of autocorrelation coefficients of a synthesized wave. The purpose is to determine the frequency and intensity (power amplitude) of each sine wave to be synthesized so that they match the N sine waves.

Now, if the number of sine waves to be synthesized is n, the angular frequency of each sine wave is ω _i (i = 1, 2, ... n), and the intensity of each sine wave is m _i , then the CSM composite wave y _t becomes y _t = _o 〓 ⁱ⁼¹ √ _i sin (ω _i t + φ _i ), but the autocorrelation coefficient γ _l of this tap l can be easily expressed using ω _i , m _i , and γ _l = _o 〓 ⁱ⁼¹ m _i coslω _i .

On the other hand, the sample of the audio waveform to be expressed is
When X _t is assumed, the sample autocorrelation coefficient v _l of tap l in a certain frame is given as v _l =1/M _M-1 〓 ^t=l X _t X _t -l. However, M is the number of samples in one analysis frame.

Now, in the CSM analysis, the value of each m _i and ω _i is determined so that the above-mentioned γ _l is equal to the given v _l for N low-order values.

That is, the values of m _I and ω _i are determined so that γ _l =v _l where l=0, 1, 2, . . . N holds true.

This specific method will be explained in detail later, but here, m _i and ω _i of the n sine waves mentioned above are shall be obtained.

FIG. 1 shows an example of a speech feature vector pattern based on the CSM parameters m _i and ω _i obtained in this way.

In addition, the 9th order (N = 9) CSM (number of sine waves n =
5) Figure 2 shows an example of the correspondence between the line spectrum and the 9th-order LPC spectrum envelope (frequency transmission characteristic of the LPC synthesis filter) obtained from the same audio sample.

Note that there is a function N=2n-1 between the above-mentioned order N and the number n of sine waves, as described later.

From these figures, it can be seen that the CSM contains information that extracts the features of the original speech that should be expressed.

However, using n sets of m _{i and} ω _i values obtained as a result of CSM analysis, the intensity specified by m _i and ω _i (the actual amplitude is √ _i as mentioned above)
If you create n sine waves with angular frequencies and simply add and synthesize them, the human ear will hear:
You can simply hear it as a synthesized sound of sine waves,
The purpose of reproducing the original sound cannot be achieved.

This means that even if you simply add sine waves, the spectrum of the generated signal is just a discretized n line spectrum, whereas the spectrum of the audio signal has a continuous spectral envelope, and In addition, voiced sounds have a pitch structure, while unvoiced sounds have a fine spectral structure that is represented by a stochastic process. This is because the spectral structure of a simply added CSM and a speech signal is completely different. Conceivable.

Therefore, to synthesize speech using CSM,
It is necessary to use some method to spread the line spectrum into a continuous spectrum. In other words
CSM speech synthesis can be thought of as generating a speech spectrum pattern from a speech feature vector pattern expressed by a line spectrum as shown in FIGS. 1 and 2.

In the present invention, the following method is used to perform the above-described spread spectrum in CSM speech synthesis.

That is, since voiced sounds have a clear pitch structure, the phase of each of the n sine waves specified as described above is reset every pitch period. This makes it possible to easily generate a spectral envelope and a fine pitch spectral structure.

Furthermore, by applying special time window processing to the above-mentioned phase reset waveform as detailed in the explanation of the embodiment, discontinuity in the synthesized waveform at the time of phase reset can be removed and continuity of the audio waveform can be ensured. ing.

Through the above implementation, the CSM line spectrum shown in Figure 2 is diffused as shown in Figure 3A, and changed to a spectrum having a spectral envelope and pitch fine structure, which is sufficiently practical for auditory purposes. Experimental results have shown that durable sound quality can be obtained.

For reference, the CSM spectrum obtained by simple addition without performing the above processing is shown in Figure 3B.
Shown below. As mentioned above, a waveform with such a spectrum simply sounds like a synthesized sound of sine waves, and the purpose of reproducing speech cannot be achieved.

The above is for voiced sounds, but in the case of unvoiced sounds, it is performed as follows. In other words, in the case of voiced sounds mentioned above, the phase reset and special time window processing performed for each pitch period are performed, whereas in the case of unvoiced sounds, the period, which is randomly generated as a stochastic process, is used instead of the pitch period. Using a pulse with a distribution width and a lower limit value set, the above-described process is performed every time this pulse is generated.

By using the above method, it is possible to perform CSM synthesis that is auditorily sufficient for practical use. Note that the CSM synthesis described above is a synthesis method that does not use a filter, so there is no need to consider stability on the synthesis side. For this reason, when using a communication method that transmits the information of m _i and ω _i to the synthesis side and reproduces the voice on the synthesis side, the line quality is relatively poor and errors may occur during transmission. In some cases, it may be possible to obtain better sound quality than a vocoder.

(Example) Next, the present invention will be explained in detail using examples.

FIG. 4 is a block diagram showing one embodiment of the present invention.

This embodiment consists of a transmitting side 1 and a receiving side 2.

The transmitting side 1 further includes an A/D converter 101, a Hamming window processor 102, and an autocorrelation coefficient measuring device 10.
3. After CSM analyzer 104 and CSM quantizer 10,
Power quantizer 106, pitch extractor 107, voiced/unvoiced sound determiner 108, and multiplexer 1
Including 09.

Further, the receiving side 2 further includes a demultiplexer and decoder 201, an interpolator 202, a voiced/
Unvoiced sound switch 203, period calculator 204, random number generator 205, n variable frequency oscillators with phase reset function 206-1, 206-2, 206-n,
n variable gain amplifiers 207-1, 207-2,
...207-n, addition combiner 208, variable length window function generator 209, multiplier 210, and multiplier 21
Contains 1.

Now, the operation of this embodiment is as follows. The audio waveform to be transmitted is fed via an input line 1000 to an A/D converter 101, where it is converted into digital data whose amplitude and time axis are quantized, and whose outputs are each subjected to Hamming window processing. 102, pitch extractor 107, and voiced/unvoiced sound determiner 108.

The digital data supplied to the Hamming window processor 102 is subjected to weighted multiplication using a known Hamming window function for each predetermined frame, and is supplied to the autocorrelation coefficient measuring device 103 for each frame of data. Ru.

The autocorrelation coefficient measuring device 103 calculates the lowest N autocorrelation coefficients v _l (where l=
1, 2,...N).

In other words, the data of one frame sentence is X _t (however,
t ⁼ 0, 1, ..., M _{-1 )} , then _each of N _{v l} is demand.

The set of v _l for each frame obtained in this way is supplied to the next CSM analyzer, and the
v _p (that is, v _p =1/M _M-1 〓 ^t=0 X ² _t ) is supplied to the power quantizer 106 as power information in this frame.

Now, the autocorrelation coefficient v _l for each frame mentioned above
The CSM analyzer 104, which has been supplied with the set m _i , ω i (where ω _i i=1,2...n)
and supplies this to the CSM quantizer 105.

The CSM quantizer 105 quantizes the set of values of these m _i and ω _i with an appropriate coarseness determined by taking into consideration the requirements for reproduction sound quality and the transmission capacity of the line, and then supplies the quantizer to the multiplexer 109 .

In addition, the power quantizer 10 supplied with the above-mentioned v _p
6 also quantizes v _p with an appropriate coarseness determined from the above-mentioned viewpoint, and then similarly supplies it to the multiplexer 109 .

The pitch extractor 10 receives digital data of the original audio signal from the A/D converter 101.
7 extracts the pitch period from this digital data and supplies it to the multiplexer 109 as appropriately quantized data.Similarly, the voiced/unvoiced sound determiner 108 also determines voiced/unvoiced sound from the supplied digital data. and supplies it to the multiplexer 109 as a binary signal.

The multiplexer 10 supplied with the above signals
9 combines these signals into a form that can be easily separated on the receiving side and is suitable for transmission over a given transmission path, and transmits the combined signals to the receiving side 2 via a transmission path 1200.

Now, on the receiving side 2, the thus transmitted signals are decoded and separated in the demultiplexer and decoder 201, thereby restoring each signal on the input side of the multiplexer 109 on the transmitting side 1.

Each signal thus restored is supplied to an interpolator 202 having a memory function, and after performing necessary interpolation, it is used as follows.

First, ω _i (ω ₁ to _{ω o} specifying the angular frequency of each of the n waves of the CSM are applied to the frequency control inputs of the n variable frequency oscillators with phase reset function 206-1 to 206-n). , set the output angular frequencies of these oscillators to the specified angular frequencies ω ₁₂ to ω _o .

Also, the intensity (power amplitude) of each of the n waves of CSM
m ₁ to m _o that specify do.

The n outputs obtained in this way are
After addition and synthesis are performed in step 08, the signals are supplied to the next multiplier 210.

Now, the demultiplexer and decoder 201
The pitch period information outputted from the interpolator 202 including a memory performs interpolation as necessary, and is supplied to the voiced/unvoiced sound switch 203 as digital data representing the pitch period.

On the other hand, the random number generated by the random number generator 205 is
It is supplied to the period calculator 204, where it is converted so that the random number distribution width and its lower limit become specific values, and is used as a data string for determining the phase reset time interval for unvoiced sounds by the voiced/unvoiced sound switcher 203. fed to the other input.

Further, a binary signal (V/U) that distinguishes between voiced and unvoiced sounds output from the demultiplexer and decoder 201 is supplied as a switching control signal to the above-mentioned switch 203, and in the case of voiced sounds, the switch 20
3 selects the digital data representing the aforementioned pitch period output from the interpolator 202 and supplies it to the window function generator 209 .

If the binary signal (V/U) specifies an unvoiced sound, the switch 203 selects the data string side representing a random time interval generated in the stochastic process of the output of the period calculator 204. This is then supplied to the window function generator 209 instead of the digital data string representing the pitch period described above.

Now, the window function generator 209 is for generating a window function that ensures the continuity of the audio waveform except for discontinuities that occur in the output waveform due to phase reset, and is also closely related to this window function. A time-related phase reset pulse is also generated.

As mentioned above, the window function generator 209 includes a switch 2.
03, a data string specifying the interval between the next phase reset pulses is input, and the window function generator 209 successively generates impulses having the time intervals specified by this data. , is supplied via line 2090 to the phase reset terminals of variable frequency oscillators with phase reset function 206-1 to 206-n, thereby resetting the phases of these oscillators. Also add this to line 2090
and is used as a timing signal for interpolating the angular frequency data ω _i and the intensity data m _i .

That is, if T is the value of the interval between phase reset pulses at that point specified by the input data, and x is the elapsed time since the previous phase reset pulse was generated, W _(x) = 0.5 + 0. .5cos(xx/T) However, it generates a window function expressed as 0<xT. This window function W _(x) is shown in FIG. 5A. The above-mentioned value of T represents the pitch period in the case of a voiced sound, and represents a variable that occurs in a stochastic process in the case of an unvoiced sound, so it changes over time. Therefore, this window function W _(x) has a variable length and is synchronized with the generation of the phase reset pulse described above in the relative time relationship shown in FIG. 5B (the start and end points of the window function (This almost coincides with the time point at which the phase reset pulse is generated.)

The window function thus generated is provided to multiplier 210 via line 2091. As a result, the multiplier 210 generates n sine waveforms whose phase is reset for each phase reset pulse synthesized by the summing synthesizer 208, and the above-mentioned window function generated in synchronization with each phase reset pulse. The product with W(x) is obtained. The waveform obtained in this way is continuously converged to 0 by multiplying the window function W(x) just before each sine wave is phase reset, and since each sine wave rises from 0 at the time of phase reset, the waveform Continuity is ensured, and thus discontinuity occurring in the phase reset waveform by multiplication by the window function W(x) can be removed.

The output of the multiplier 210 from which discontinuities have been removed is supplied to the next multiplier 211, where it is weighted by the power information of each frame sent from the transmitting side 1, and is output from line 2000 as synthesized speech. be done.

As explained above, on the receiving side 2 of this embodiment, the CSM synthesis necessary for the above-mentioned speech synthesis is executed, and as a result, the reproduction of the original voice inputted to the transmitting side 1 is transmitted using the information on the transmission path 1200. This is done with relatively good sound quality despite volume compression and transmission errors.

The interpolation for each transmission data by the interpolator 202 described above is performed using various combinations (for example, only ω _i , or only ω _i , m _i , etc.), and as for the interpolation method, it is also possible to use linear interpolation or interpolation using a higher-level function. Regarding the interpolation for ω _i , m _i , it is advantageous to select the interpolation points so that interpolated data can be obtained at each point in time when the above-mentioned phase reset pulse occurs, that is, the temporal distortion of ω _i , m _i is In order to update the values of ω _i and m _i at this timing, the phase reset pulse is supplied to the interpolator 202 via the line 2090 as described above.

To perform such an interpolation, after the desired later data arrives or occurs,
Since interpolated data is obtained, actual processing such as resetting the phase and setting the frequency ω _i for the oscillator 206, and setting the intensity m _i for the amplifier 207 must be performed with a delay of a necessary fixed time from real time. become. For this reason, interpolator 202 includes a memory for storing necessary information until a necessary point in time.

Next, variable frequency oscillator 2 with phase reset function
A circuit example of 06 is shown in FIG. Frequency control terminal 2
The voltage applied to 061 causes constant current power supply 206
2 and 2063. The charging/discharging current value for the capacitor 2064 is controlled, thereby making the oscillation frequency variable. The oscillation voltage waveform at point v is a triangular waveform that linearly rises and falls between +Vr and -Vr of the reference voltage. When an impulse is applied to the phase reset terminal 2065, point v is momentarily grounded and forcibly pulled back to 0 potential, and oscillation is restarted from there to perform a phase reset. this. The triangular wave oscillation output at point v is input to a sine wave converter 2066, converted to a sine wave, outputted from a terminal 2067, and used as the output of the oscillator 206. The sine wave converter 2066 can be easily realized, for example, by reading out a sine function value stored in a ROM using an input waveform.

Further, such a variable frequency oscillator with a phase reset function can be easily realized using a computer program.

Next, a circuit example of the variable gain amplifier 207 is shown in FIG. By applying a signal to be amplified to a terminal 2071 and a control signal to a terminal 2072, the amount of negative feedback is controlled, and an output having a controlled amplitude is obtained at an output terminal 2073.

In addition, it can also be realized by using an analog multiplier, or by using an analog waveform input of the reference voltage of a D/A converter and using control information expressed in digital quantities as a digital input. It can also be easily realized by a method.

Next, an example of a circuit of the random number generator 205 is shown in FIG. The 15-stage shift register 2051 and one intermediate adder 2052 have synchronization of 2 ¹⁵ -1.
Generate the next M series of pseudo-random numbers. By applying a shift pulse to clock terminal 2053 at the required time, the next random value is obtained.

Next, the block diagram of the period calculator 204 is shown in FIG. 9A.
Shown below. This is suitable for using the random numbers uniformly distributed in the range of 0 to ²¹⁵ -1 output from the random number generator 205 mentioned above as random numbers for specifying the time interval of the phase reset pulse during unvoiced speech. It is composed of a constant multiplier 2041 and a constant adder 2042. Thereby, as shown in FIG. 9B, the random number distribution width D and the lower limit L can be set to appropriate values.

Next, an embodiment of the window function generator 209 is shown in FIG. This includes a register 2091, a presettable down counter 2092, and a counter 209.
3. Contains read-only memory (ROM) 2094.

Data T specifying the phase reset pulse interval supplied from the switch 203 is stored in the register 209.
It is stored in 1. The down counter 2092 is a counter that counts the high speed clock CLK of a fixed period. First, the content T of the register 2091 is preset, and this is down counted using the clock CLK. When the contents of the counter 2092 reach 0, a pulse is generated from the output terminal, thereby presetting the contents of the register 2091 again and starting counting down this value. Thus, a pulse train having a period proportional to T (for example, T/k) is generated at the output 2092-1 of the down counter 2092. This pulse train is added as a clock to counter 2093. The count output 2093-1 of the counter 2093, which is incremented by this clock, is
The data of the window function w(x) that is added to the ROM 2094 as an address designation signal and written therein is sequentially read out and output to the line 2091. When the content of the counter 2093 reaches k, the ROM 20
The last data of the 94 window functions w(x) is read out, and at the same time, the counter 2093 is reset and outputs a reset pulse on line 2090. This reset pulse is generated by the oscillators 206-1 to 206-
It is used as the aforementioned phase reset pulse supplied to the phase reset terminal of n and the interpolator 202, and is also used to set the next input data in the register 2091. Also ROM2
The data of the window function w(x) previously stored as k samples in 094 is read out on line 2095 and supplied to multiplier 210. Thus,
Phase reset pulses whose inter-pulse intervals have successively specified values and a variable length window function w(x) synchronized with these pulses as shown in FIG. 5B are generated.

Finally, we will explain CSM analysis.

As mentioned above, CSM analysis uses, for each analysis frame, N low-order tap values of sample autocorrelation coefficients calculated directly from the speech waveform to be represented;
The frequency ω _i of each sine wave to be synthesized and its intensity (power amplitude) m _i are determined so that the N low-order tap values of the synthesized wave (sum of n sine waves) match. That's true.

Now, if the autocorrelation coefficient of tap l of the composite wave is γ _l , then as mentioned above, γ _l = _o 〓 ⁱ⁼¹ m _i coslω _i .

On the other hand, _the sample of the voice waveform to be expressed
Therefore, the sample autocorrelation coefficient v _l of tap l in a certain frame is v _l =1/M _M-1 〓 ^t=l X _t X _t -l (1).

From this, if γ _l =v _l ...(2) l=0, 1, 2,...N, where N=2n-1, the following matrix expression is obtained.

However, the above equation cannot be solved by simple matrix operations because ω _i and m _i are unknown. Therefore, we set ω _i = cos ^-1 X _i ...(4) and perform the substitution coslω _i = cosl cos ^-1 X _i ≡T _l (x _i ) ...(5). This T _l (x) is a Tchebycheff polynomial. If you do this substitution
Equation (3) is converted as follows.

However, in general, x ^l can be expressed as a linear combination of T ₀ (x), T ₁ (x)...T _l (x).

That is, x ^l = _l 〓 ^j=0 S ⁽¹⁾ _j T _j (x) ... (7) where S(l) _j is an inverse Tchebycheff function.

Using this S(l) _j , the sample autocorrelation coefficient v _j
Define the linear combination A _l as shown below.

A _l = _l 〓 ^j=0 S ⁽¹⁾ _j v _j ……(8) However, l=0, 1, 2, ……, 2n−1 In this way, on the left and right sides of equation (6), respectively
By using the relationships of equations (7) and (8), the following relational expression is established.

Now, here, the n _- th degree polynomial P _o (x ₎ ≡ _o 〓 ^k=0 p ⁽ⁿ⁾ _k ^{X k} _{= o} Π ⁱ⁼¹ ( x−x _i ), and using this P _o (x), create _o 〓 ⁱ⁼¹ m _i P _o (x _i )x ^l _i , which is similar to the left side of equation (9), and Let's consider. It is clear that the above equation is 0, but it can be further rewritten as follows.

0= _o 〓 ⁱ⁼¹ m _i P _o (x _i ) x ^l _i = _o 〓 ⁱ⁼¹ m _io 〓 ^k=0 p ⁽ⁿ⁾ _k X ^k+l _i ＝ _o 〓 ^k=0 p ⁽ⁿ⁾ _ko 〓 ⁱ⁼¹ m _i x _i ^k+l = _o 〓 ^k=0 p ⁽ⁿ⁾ _k A _k+l From the above, the following formula can be obtained with l=0, 1, 2, . . . n.

However, p ⁽ⁿ⁾ _o =1.

holds true. The matrix formed by A _i on the left side is generally called the Hankel matrix. As described above, each A _i is given by equation (8) from the sample autorelation coefficient v _j of the expressed speech waveform and is known.

Therefore, by solving equation (10), p ⁽ⁿ⁾ ₀ , p ⁽ⁿ⁾ ₁ , ...
The value of p ⁽ⁿ⁾ _o-1 can be found.

When each p ⁽ⁿ⁾ _i is found, as a solution to the n-dimensional equation P _o (x)=x ⁿ +p ⁽ⁿ⁾ _o-1 x ^n-1 +...p ⁽ⁿ⁾ ₀ =0, {x ₁ , x ₂ ..., x _o } are obtained.

From this, each CSM frequency ω _i is obtained from ω _i =cos ⁻¹ x _i in equation (4), and the CSM intensity m _i is obtained using the following equation derived from equation (9).

Note that the matrix on the left side of the above equation is generally Van der
This is called the Monde (Juan Del Monde) matrix.

To summarize the above, the analysis algorithm for CSM analysis is as follows.

(1) Calculate the sample autocorrelation coefficient v _l =1/M _M-1 〓 ^t=l X _t X _t −l (2) Define A _l using the inverse Tievisiev coefficient.

A _l = _l 〓 ^j=0 S ⁽¹⁾ _j v _j (3) Solve the Hankel matrix equation by A _l to find p ⁽ⁿ⁾ _i (4) p ⁽ⁿ⁾ _Find n x _i by solving an n-dimensional algebraic equation with i as a coefficient.

P _o (x)≡x ⁿ ＋p ⁽ⁿ⁾ _o-1 x ^n-1 ＋p ⁽ⁿ⁾ _o-2 x ^n-2 ＋……＋p ⁽ⁿ⁾ ₁
x+
p ₀ =0 (5) Perform inverse cos transform to find CSM angular frequency {ω _i }.

ω _i =cos ^-1 x _i (6) Solve the Van der Monde matrix equation to find the CSM intensity {m _i }.

By performing each of the above steps, CSM
Each frequency {ω ₁ , ω ₂ ...ω _o } and the intensity of each wave {m ₁ , m ₂ , ... m _o } can be obtained.

Note that, as an efficient method for solving the above-mentioned Hankel matrix equation, a method is known in which initial conditions are given and solutions are sequentially obtained.

Furthermore, since it has been proven that the above nth-order algebraic equation has only real roots, the roots can be found using the Newton-Raphson method or the like.

Furthermore, as an efficient method for solving the Van der Monde matrix equation, a method can be used in which triangular matrixing is performed and solutions are sequentially obtained. The above analysis method is based on the paper "Speech spectrum analysis using a composite sine wave model" by Mr. Sagayama et al.
Journal of the Institute of Electronics and Communication Engineers '81/2Vol.J64-ANo.2P.105~
112 is detailed.

The above shows one embodiment of the present invention, and the present invention is not limited to the above embodiment.

For example, in the CSM analysis on the transmitting side, in this example, a method of solving an equation in which the sample autocorrelation coefficient and the CSM autocorrelation coefficient are equal is used. It is also possible to use a method based on calculation of the line spectrum frequency and residue calculation. in any case,
Good sound quality can be achieved by supplying the appropriate CSM parameters obtained by CSM analysis and other necessary parameters to the receiving side of the above embodiment.
CSM speech synthesis becomes possible. In this way, a CSM type speech analysis and synthesis device can also be configured.

Further, in this embodiment, the interpolator performs parameter interpolation at the time of phase reset, but this may be omitted.

Further, in this embodiment, a variable length window function having a specific function form is used, but this function form is merely an example, and it is clear that other function forms may be used.

Furthermore, the random number generator, period calculator, etc. are also shown as examples, and there is no need to be limited thereto.

(Effects of the Invention) As described above, by using the present invention, it is possible to provide a CSM type speech synthesizer that synthesizes speech signals with good sound quality using CSM parameters.

This synthesizer has a simple structure and does not include a filter.
Therefore, it has advantages such as no stability problem on the synthesis side, and can be used to improve the performance of speech transmission devices, speech analysis and synthesis devices, etc.

[Brief explanation of the drawing]

Figure 1 shows an example of a voice feature vector pattern based on CSM parameters, and Figure 2 shows a CSM line spectrum and a pattern obtained from the same voice sample.
Diagram showing an example of correspondence with LPC spectrum envelope, 3rd
Figure A is a diagram showing the spectral envelope of the diffused CSM and the fine structure of the pitch, Figure 3B is a diagram showing the CSM spectrum obtained by simple addition, and Figure 4 is a block diagram showing an embodiment of the present invention. , FIG. 5A is a diagram showing the functional form of the variable length window function, FIG. 5B is a diagram showing the relative time relationship between the variable length window function and the phase reset pulse, and FIG. 6 is a diagram showing the function form of the variable length window function. 7 is a diagram showing an example of a circuit of a variable gain amplifier, FIG. 8 is a diagram showing an example of a random number generator circuit, and FIG. 9A is a block diagram of a period calculator. 9B is a diagram showing the distribution of random numbers output from the period calculator, and FIG. 10 is a block diagram showing an example of a variable length window generator. In the figure, 1...sending side, 2... receiving side, 1
01... A/D converter, 102... Hamming window processor, 103... Autocorrelation coefficient measuring device, 104...
...CSM analyzer, 105 ...CSM quantizer, 10
6...Power quantizer, 107...Pitch extractor,
108... Voiced/unvoiced sound determiner, 109... Multiplexer, 201... Demultiplexer and decoder, 202... Interpolator, 203... Voiced/unvoiced sound switcher, 204... Period calculator, 20
5... Random number generator, 206-1 to 206-n...
Variable frequency oscillator with phase reset function, 207-
1 to 207-n...variable gain amplifier, 208...
Addition synthesizer, 209...Variable length window function generator, 2
10,211...multiplier.

Claims (1)

[Claims] 1. A plurality of variable frequency oscillators with a phase reset function set to each frequency specified by the CSM, and a plurality of variable frequency oscillators with a phase reset function correspondingly set to each intensity specified by the CSM. It is equipped with a variable gain amplifier, a variable length window function generator, and a random number generator, and resets the phase of each of the oscillators according to the pitch period when synthesizing voiced sounds, and converts the output of the random number generator into random numbers when synthesizing unvoiced sounds. The phase of each oscillator is reset in accordance with the period set with the distribution width and the lower limit calculated by 1. A CSM type speech synthesizer, characterized in that the start and end points of the variable length window function substantially coincide with the phase reset point. 2. The CSM type speech synthesizer according to claim 1, wherein interpolation of the CSM parameters is performed at the time of the phase reset.