JPS6139100A - Secret talk apparatus - Google Patents

Abstract

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

Description

[Detailed Description of the Invention] (Technical Field) The present invention relates to a privacy device for polarizing voice communication, particularly
BM (Composite 5inusoidal
The present invention relates to a confidential communication device based on the principle of Mode-11ng (composite sine wave model), and relates to an analog confidential communication device with high communication confidentiality.

(Prior art) In general, as a secret communication device, an audio signal is converted into a digital code by A/D conversion, the original audio information is transmitted in a confidential manner by eliminating specific code conversion, etc., and the receiving side converts the original audio information into a reverse code. Digital secret speech devices that reproduce the original audio signal by D/A conversion after conversion are widely used.

However, such digital methods require limited transmission capacity and
It has the disadvantage that high requirements are placed on the transmission quality of the transmission path, such as rate.

On the other hand, there are analog secret communication devices that use water droplets, such as inverting the spectrum of the audio signal or dividing it and switching the relative positions, but these generally do not meet the requirements for the transmission quality of the transmission path. Although it is low,
Since the spectral envelope of the original audio signal remains in some form, it generally has the disadvantage of low privacy.

(Object of the Invention) An object of the present invention is to provide a new type of analog secret speech device with high speech privacy based on the principles of speech analysis and speech synthesis using CSM, which will be described later.

(Structure of the Invention) The device of the present invention includes a CSM analysis means, a means for converting the distribution range of CSM frequency data calculated by the means at a constant ratio, and a frequency specified by the converted CSM data. a transmitting side having CSM spectrum generating means for generating an additively synthesized waveform of a plurality of sine waves having amplitudes; spectral parameter extraction means for extracting each parameter specifying the frequency and amplitude of a sine wave; means for inversely transforming the extracted parameters at the transmitting side; and specifying the inversely transformed parameters. a plurality of variable frequency oscillators with a phase reset function set to respective frequencies; and a variable gain amplifier correspondingly configured to set the output of each of the variable frequency oscillators with a phase reset function to an amplitude specified by the extracted parameter; A variable length window function generator and a random number generator are provided, and when synthesizing voiced sounds, the phase of each variable frequency oscillator with a phase reset function is reset in accordance with the pitch period, and when synthesizing unvoiced sounds, the phases of the variable frequency oscillators with phase reset function are reset. resets the phase of each of the variable frequency oscillators with a phase reset function in accordance with the period calculated from the random number output from the random number generator, and determines the starting point and time of the window function generated by the variable length window function generator. and a receiving side whose end point substantially coincides with the phase reset point.

(Principle) As mentioned above, the secret speech device of the present invention is based on the principle of CAM speech analysis and speech synthesis.

First, the principles of CSM speech analysis and speech synthesis will be explained.

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

Therefore, when performing CAM speech synthesis, it is first necessary to express the speech signal as a sum of a predetermined number of sine waves by CSM speech analysis.

CSM voice analysis will be explained in detail later, and only the main points will be explained here.

In CSM analysis, as in LPG 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.

That is, in CSM analysis, N low-order taps of the sample autocorrelation coefficient directly calculated from the speech waveform to be expressed for each analysis frame are used as N low-order taps of the autocorrelation coefficient of the composite wave. The purpose is to determine the frequency of each sine wave to be synthesized and its strength (power amplitude) so that it matches the individual sine waves.

Now, if the number of sine waves to be combined is n, each frequency of each sine wave is ω1 (i = 1.2...n), and the intensity of each sine wave is mi, then the CBM composite wave yt is yt-, Genden (ωit+φi), but the autocorrelation coefficient γ□ of this tap L is ωi, m
It is easily expressed using i, and is .

On the other hand, if the sample of the audio waveform to be expressed is Xt, then the sample autocorrelation coefficient t of tap t in a certain frame
lt is given as.

However, M is the number of samples in one analysis frame.

Now, in the CSM analysis, each mi, ω is set so that the above rt is equal to the given υ6 for N low-order
The purpose is to determine the value of i.

That is, the value of m2.machi is determined so that rt'''t'where t=0.1, 2.・N holds true.

This specific method will be explained in detail later.
Here, it is assumed that the above-mentioned sine waves ω and ωt are obtained one after another for each effort frame in response to a given audio signal.

An example of the speech feature vector pattern obtained in this manner using the 90 SM parameters mi, ωi is shown in FIG.

Also, the 9th order (N, = 9) CAM (number of sine waves n = 5) analyzed with the window length of 7 frames analyzed as 30m5EC.
9 obtained from the line spectrum and the same audio sample
FIG. 2 shows an example of correspondence with the following LPC spectrum envelope (frequency transmission characteristics of the LPC synthesis filter).

Note that there is a relationship of 1'J=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 fi and CSM contain information extracted from the features of the original speech to be expressed.

However, using the n sets of mt, ω values obtained as a result of the CSM analysis, we can calculate the If a sine wave is created and the sine waves are simply added and synthesized, the human ear can only hear it as a synthesized sound from the sine wave, and cannot hear it as the original sound at all. I'm doing it.

This means that even if sine waves are simply added, the spectrum of the generated optical signal is only 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, and unvoiced sounds have a mechanical spectral structure expressed by a beat probability process, and the spectral structure is completely different from the simply added CSM and speech signal. This is thought to be due to this.

Therefore, in order to synthesize audible speech using CSM, it is necessary to spread the line spectrum into a continuous spectrum using some special method. In other words, CSM speech synthesis means that a speech feature vector pattern expressed by a line spectrum as shown in FIGS. 1 and 2 is diffused using a special method to generate a speech spectrum pattern. If this advanced CSM speech synthesis is not performed, the simple addition synthesis waveform of multiple sine waves with the frequency and amplitude specified by CSM will not be able to acquire the information necessary to reproduce the original speech. Even though it is contained in a certain form, it cannot be heard as a sound at all.

The secret speech device of the present invention utilizes the above-mentioned characteristics of the CSM simple addition synthesis waveform.

That is, the CSM analysis of the input audio signal is performed on the transmitting side, and an analog signal of a simple addition composite waveform of a plurality of sine waves having the angular frequency and amplitude specified by the CSM is generated and sent to the transmission path. As mentioned above, although this synthesized waveform contains the information necessary to reproduce speech in the most basic form, it has a highly confidential nature in that it cannot be heard as speech at all. Furthermore, if particularly necessary, the confidentiality can be further improved by subjecting the CSM parameters to a predetermined specific conversion as will be described in detail later.

Now, on the receiving side, special CSM speech synthesis is performed to reproduce the original speech. The method of CSM speech synthesis using spread spectrum on the receiving side of the confidential communication device of the present invention is as follows.

That is, since voiced sounds have a clear pitch structure, each of the n sine waves of the CSM specified as described above is
The phase 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 function processing to the above-mentioned phase reset waveform as detailed in the explanation of the embodiment, discontinuity of the synthesized waveform at the time of phase reset is removed and continuity of the audio waveform is ensured. are doing.

Through the above implementation, the 2-in spectrum of 90SM shown in Figure 2 is diffused as shown in Figure 3) and changed to a spectrum having a spectral envelope and a fine structure of pitch, which is also audible enough. Experimental results have shown that sound quality that can withstand practical use can be obtained.

For reference, the spectrum of the CSM that is simply added without performing the above processing is shown in Figure 3 (flatly shown).As mentioned above, this is the spectrum of the signal transmitted via the transmission path, A waveform with such a spectrum can only be heard as a synthesized sound of sine waves, and cannot reproduce the original sound.

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 function processing performed for each pitch period are performed, whereas in the case of unvoiced sounds, pulses generated randomly by KM are used as a stochastic process instead of pitch periods. , the above-mentioned processing 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 since the above CSM synthesis is a synthesis method that does not use a filter, there is no need to consider stability on the synthesis side. Therefore, when using a communication means such as this device that transmits mi and ωi information to the synthesis side and reproduces the voice on the synthesis side, when the line quality is relatively poor, it is possible to obtain better sound quality than a vocoder. Another possible feature is that it can be obtained.

However, as mentioned above, simply adding sine waves will not be perceived as speech. The confidential communication device of the present invention is based on this principle, and further improves the confidentiality by means of converting the distribution range of the sine wave.

(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.

In this embodiment, -e is transmitted from the transmitting side 1 and the receiving side 2.

The transmitting side 1 further includes an A/D converter 101, a Hamming window processor 102, an autocorrelation coefficient measuring device 103, a CSM analyzer 104, a pitch, 'v/UV analyzer 105, and a parameter converter 10Ln variable frequencies. Oscillators 107-1 to 10
7-n, n variable gain amplifiers 108-1 to 108-n
, addition combiner 109 , variable gain amplifier 110 , variable frequency 'Jls oscillator 111 , V/UVx switch 11
2, and an addition synthesizer 113.

In addition, the receiving side 2 further includes a 1-spectrum analyzer 201,
Power separator 202, parameter inverter 203, n variable frequency oscillators with phase reset function 2Q) 4f-1~
204-nSn variable gain amplification 1) 205-1 to 20
5-n, addition combiner 206, multiplier 207, multiplier 20
8, V/UV switch 209, variable length window function generator 210
, a period calculator 211 and a random number generator 212.

The operation of this embodiment is as follows. The audio waveform to be transmitted is sent to the A/D via input line 1000.
It is supplied to a converter 101, where the amplitude and time axis are converted to quantized digital data, and the outputs are sent to a normink window processor 102 and a pitch/'v
/Uv is supplied to the analyzer 105.

The digital data supplied to the Hamming window processor 102 is subjected to weight multiplication using a known Hamming window function for each predetermined frame, and is supplied to the autocorrelation coefficient measuring device 103 for each 7 frames of data. be done.

The autocorrelation coefficient measuring device 103 calculates the lowest N autocorrelation coefficients vt (where t=0.1...
. . , M-1), each of N Ut is obtained by performing the calculation process.

The utO set for each frame obtained in this way is used as the next CS
It is supplied to the M analyzer 104 and also to the parameter converter 106 as power information in this frame.

Now, the CSM analyzer 104, which has been supplied with the above-mentioned set of autocorrelation coefficients UtO for each frame, performs calculations K that will be detailed later. mi, ω, specifying the angular frequency
(where i=1.z, . . . n) is determined and supplied to the parameter converter 106.

Further, the pitch/V/UV analyzer 105, which receives the digital data of the original audio signal from the A/D converter 101, extracts information regarding the pitch period and voiced sound (V)/unvoiced sound (UV) and converts it into a parameter. Converter 106 is supplied.

Now, the parameter converter 106 is a converter that performs parameter conversion of this information and outputs it, but to simplify the explanation, initially this converter does not perform any conversion, but converts the input signal. shall be output as is.

Thus, the n pieces of angular frequency information ω obtained by the CSM analyzer 104 are supplied to the variable frequency oscillators 107-1 to 107-n via the converter 106, and specify the output angular frequency of each of these oscillators. and set it to 9ωi. Further, the n intensity information mi obtained by the CSM analyzer 104 is similarly transmitted to the variable gain amplifier 108-
1 to 108-n, and sets the output of each of the oscillators 107-1 to 107-H to a designated intensity mi.

In this way, the output of the additive synthesizer 109 is a waveform obtained by simple additive synthesis of a plurality of sine waves having amplitudes and frequencies specified by the CSM.

The above composite waveform is further processed by the variable gain amplifier 110.
The total power is the electric power obtained by the correlator 103. After being controlled so that it is proportional to , it is supplied to the synthesizer 113.

On the other hand, the pitch period information obtained by the analyzer 105 is supplied to the variable frequency oscillator 111, and the frequency of this oscillator 111 is controlled to become the fundamental frequency of the given pitch. Furthermore, analyzer 105 or bo voiced/lt, voice (V/
UV) tHH: is supplied to the V/UV switch 112,
In the case of a voiced sound (V), the output of the oscillator 111 is supplied to the synthesizer 113, and in the case of an unvoiced sound (UV), it is controlled to be turned off.

In this way, the pitch information and the summed composite waveform of each sine wave of the power-controlled CSM are transmitted to the transmission line 1200 connected to the synthesizer 113 in the form of an analog signal, but as described above, this signal is It has a confidential nature in that it cannot be heard as audio even if it is converted to IF sound.

Now, on the receiving side 2, the signal thus transmitted is subjected to spectrum analysis by a spectrum analyzer 201, and each angular frequency information ω specified by the line spectrum of the CSM and each intensity weighted by the total power are Information m is obtained, the latter being the total power information ν in the power separator 202. and intensity information mi of each sine wave of the CSM.

Furthermore, pitch period information and V/UV information are extracted in the spectrum analyzer 201, and each of the above information is supplied to a parameter inverse transformer 203.

Now, the parameter inverse transformer 203 is a converter that performs the inverse transformation of the parameter converter 106 on the transmitting side described above, but in order to make the explanation easier to understand as before, first, the parameter converter 106 on the transmitting side Corresponding to the case where the parameter inverter 203 outputs the input signal as it is without performing any conversion, it is assumed that the parameter inverse transformer 203 also outputs the input signal as it is.

Thus, the output of spectrum analyzer 201, CSM
ωi (W□~wn
) is added to the frequency control input of the n variable frequency oscillators with phase reset function 204-1 to 204-n,
The output angular frequency of these oscillators is the specified angular frequency W
Set to 0 to Wn.

In addition, a door for specifying the strength (power amplitude) of the n can liquids of the CSM is connected to the n variable gain amplifiers 205-1.
The signal is supplied to the gain control terminal of 205-H, thereby controlling the oscillation power of each frequency to a specified value.

The n outputs obtained in this manner are subjected to addition and combination in the addition and combination unit 206, and then supplied to the next multiplier 207.

Now, the pitch period data and V/UV information (voiced sound/unvoiced sound information) extracted by the spectrum analyzer 201 are as follows.
v/Uv through the parameter inverse transformer 203, respectively.
The signal is supplied to the switch 209.

On the other hand, the random numbers generated by the random number generator 212 are supplied to the period calculator 211, where they are converted so that the random number distribution width and its lower limit value become specific values, and the phase reset time interval for unvoiced sounds is determined. It is supplied to the V/UV switch 209 as a data string to be determined.

Thus, when the above-mentioned V/UV information of the spectrum analyzer 201 specifies an unvoiced sound (V), the switch 209 selects the above-mentioned pitch period data side and sends it to the variable length window function generator 210. supply

Furthermore, if the aforementioned V/UV information specifies unvoiced sound (UV), the switch 209 selects the data string side representing a random time interval generated in the stochastic process of the output of the aforementioned period calculator 211. , which is supplied to the window function generator 210 as a digital data string representing the pitch period described above.

Now, the window function generator 210 is used to generate a window function that ensures the continuity of the audio waveform by eliminating discontinuities that occur in the output waveform due to phase reset, and also to generate an odd time relationship with this window function. It also generates a phase reset pulse.

As mentioned above, the window match generator 210 includes a switch 209.
A data string specifying the interval between the next phase reset pulses is inputted via the window function generator 210.
Impulses having time intervals specified by this data are generated one after another, and are supplied to the phase reset terminals of the variable frequency oscillators with phase reset function 204-1 to 204-H through the 2-in 2101. The phase of these oscillators Now, the window function generation a21 o is of the order 4fi mentioned above.
A variable length window function -(,) as shown below is generated in synchronization with the generation of the reset pulse.

In other words, if the value of the interval between phase reset pulses at that point specified by the input data is T, and the elapsed time since the previous phase reset pulse occurs, then W (g) = 0.5 + 0.5 am (π〒) However, a window function as expressed by o<x<'r is generated. This window function W(
1) is shown in FIG. 5(A). The above-mentioned value of T represents the pitch period in the case of a voiced sound, and represents a variable generated in a stochastic process in the case of an unvoiced sound, so it is 1, and changes over time in either case. Therefore, this window function Wω has a variable length, and the above-mentioned phase reset pulse generation and Fig. 5 (B
) The window function thus generated is sent to the multiplier 207 via a line 2102 (the start and end points of the window function are for phase reset). Supplied. As a result, the multiplier 207 generates n sine waveforms whose phases are reset for each phase reset pulse synthesized by the addition synthesizer 206, and the above-mentioned window function generated in synchronization with each phase reset pulse. The product with W(b) is obtained. The waveform obtained in this way is that each sine wave is continuously converged to 0 by multiplication by the window function W (b) immediately before the phase reset, and each sine wave rises from 0 at the time of the phase reset, so the waveform The continuity of is ensured, and thus the discontinuity that occurs in the phase reset waveform by multiplication by the window function W- can be removed.

The output of the multiplier 207 with discontinuities removed is the following power: i
ii, weighted by the power information v0 of each frame sent from the transmitter 1, separated by the power separator 202, and output as synthesized speech on line 2000.

As explained above, on the receiving side 2 of this embodiment, the CSM speech synthesis necessary for the above-mentioned speech synthesis is executed, and as a result, the original input to the transmitting side 1, which was hidden on the transmission path 1200, is The sound will be reproduced with good sound quality.

In the above explanation, it is assumed that the parameter converter 106 on the transmitting side 1 and the parameter inverse converter 203 on the receiving side 2 output the input data as is to the output side and do not perform any transformation. explained. Of course, even in this case, confidentiality on the transmission path is maintained as described above.

That is, such a parameter converter 106 on the transmitting side 1 and a parameter inverse converter 20 on the receiving side 2
3 may be completely omitted to form a confidential communication device.

However, in order to obtain an even higher degree of confidentiality, the parameter converter 106 performs pagemeter conversion, and the parameter inverse converter 203 performs inverse conversion, so that each data input to the parameter converter 106 has parameters. It is sufficient if it can be obtained on the output side of the inverse transformer 203.

As an example of parameter conversion, the distribution range of CSM frequency data is converted at a constant ratio. With α and θi as constants, ωt=α,・Machi 10 (i=1.2.−・・−・N
) can be used. In addition, O
In the case of <a<1, there is also an audio band compression transmission effect.

The conversion on the receiving side is ω,; (Tachi - θ)/α (i=x, 2...N
).

Furthermore, in the embodiment described above, the pitch frequency information is transmitted from the transmitting side 1, but instead, the transmission of pitch frequency information from the transmitting side 1 may be omitted as follows. can.

In other words, by utilizing the property of voice that when the voice energy is high, the pitch frequency becomes high, and when the voice energy is low, the pitch frequency is low.
A correspondence table between voice energy and pitch frequency is created empirically. A pseudo pitch cycle generating means is provided on the receiving side 2 to generate a pitch period from the total audio power information transmitted from the transmitting side 1 using this correspondence table, and the pseudo pitch period generated thereby is used on the receiving side 2. It is also possible to omit the transmission of pitch frequency information from the transmitting side.

Next, a circuit example of the variable frequency oscillator with phase reset function 204 is shown in FIG. A voltage applied to the frequency control terminal 2041 causes a current to flow to the constant current power supplies 2o42 and 2043. The charging/discharging current value for the capacitor 2044 is controlled, thereby making the oscillation frequency variable. The oscillation voltage waveform at point υ 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 2045, point V is momentarily grounded and forced to 0
The voltage is pulled back to the potential, and oscillation is restarted from there to perform a phase reset. This triangular wave oscillation output at point V is input to the sine wave converter 2046 and converted to a sine wave.
47 and used as the output of the oscillator 204. The sine wave converter 2646 can be easily realized, for example, by reading out a sine function value stored in a ROM as 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 205 is shown in FIG. The signal to be amplified is applied to terminal 2051, and the control signal is applied to terminal 2
052 to control the amount of negative feedback and obtain an output having a controlled amplitude at the output terminal 2053.

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

Variable gain amplifier 108 can also be implemented in a similar manner.

Next, an example of a circuit of the random number generator 212 is shown in FIG. 15
stage shift register 2121 and one half adder 2122
A 15th order M-sequence of pseudo-random numbers having a period of 21!11 is generated. clock terminal 212 at the required time.
By adding a shift pulse to 3, the next random value can be obtained.

Next, a block diagram of the period calculator 211 is shown in FIG. 9 (6). This is because the random numbers uniformly distributed in the range from 0 to 2-1 output from the random number generator 211 described above are suitable for use as random numbers to specify the time interval of the phase reset pulse during unvoiced speech. It converts into a distribution, and consists of a constant multiplier 2121 and a constant adder 2122. As a result, the distribution width and lower limit value of one random number can be set to appropriate values as shown in FIG. 9(B).

Next, an embodiment of the window function generator 210 is shown in FIG.

It includes a register 2101, a presettable down counter 2102, a counter 2103, and a read only memory (ROM) 2104.

Data T specifying the phase reset pulse interval supplied from the switch 209 is stored in the register 2101. The down counter 2102 is a high-speed clock C with a constant period.
First, the register 210 is a counter that counts LK.
The content T of 1 is preset and counted down using the clock CLK. The content of the counter 2102 is O
When this happens, a pulse is generated from the output terminal, which presets the contents of the register 2101 again and starts counting down this value. Thus, the down counter 210
A pulse train having a period proportional to T (for example, T/&) is generated at the output 2102-1 of No. 2. This pulse train is added as a clock to counter 2103.

The count output 2103-1 of the counter 2103, which is incremented by this clock, is applied to the ROM 2104 as an address designation signal, and the data of the rotation function W- written therein is sequentially read out and output to the 2-in 2200. When the contents of the counter 2103 become negative, the ROM 2104
The last data of is read out and outputs a reset pulse on line 2100. This reset pulse is generated by the oscillator 20
It is used as the above-mentioned phase reset pulse supplied to the phase reset terminals 4-1 to 204-H, and is also used to set the next input data in the register 2101. Thus, the phase reset pulses whose inter-pulse intervals have the specified values one after another, and the fifth
As shown in Figure (B), a synchronized variable length window function Wω is generated.

Finally, CSM analysis will be explained.

As mentioned above, CAM analysis uses, for each analysis frame, N low-order tap values of sample autocorrelation coefficients directly calculated from the speech waveform to be represented and a composite wave (n sine waves). The purpose is to determine the angular frequency ωi of each sine wave to be synthesized and its intensity (power amplitude) rrLi so that the N low-order tap values of

Now, if the autocorrelation coefficient of tap t of the composite wave is rt, then rL=, Σ m6 (Ill Lωi 1 1) as described above.

On the other hand, from the sample Xt of the audio waveform to be expressed, the sample autocorrelation coefficient tit of tap t of a certain frame is.

From this point...(2
) t = 0, 1, 2, -...-N, however, N
;2 n-1, the following matrix expression is obtained.

However, the above equation cannot be solved by simple matrix operations because the town and 7ni are unknown. Therefore, W, ;(2)X, ...(4)
Toki, (Xll"! "" (Xli L Co! -"z
: TI-("z)"' (5). This T coefficient is a Tchebycheff polynomial. If we perform this substitution, (3)
The expression is converted as follows.

However, in general, 1 is TO←), T1)...
・T IH can be expressed as a linear combination.

i.e. It is a coefficient.

Using this B<a, the sample autocorrelation coefficient vj The linear combination At is defined as shown below.

However, /, = 0.1, 2. ..., 2n-1 Then, by using the relationships of equations (7) and (8) on the left and right sides of equation (6), respectively, the following relational expression is established.

Now, here, xl, x2. ......, define an n-th degree polynomial with a zero point at Xi, and use this Pn (x)) to create an equation similar to the left side of equation (9). . It is clear that the above formula is 0, but it can be further changed as follows.

From the above, the following formula is obtained with A=011, 21...n.

holds true. The matrix formed by Ai on the left side is generally called a Hankcl matrix. As mentioned above, each Ai is given by equation (8) from the sample autocorrelation coefficient vj of the speech waveform to be expressed and is known.

...The value of P(n) can be found.

When each P(4) is obtained, (J 1t 22..., x4) is obtained as the cancellation of the n-dimensional equation (%).

From this, each CSM frequency ωi is ωi4 in equation (4).
-L's, and the CAM strength mi is obtained using the following equation derived from equation (9).

Note that the matrix on the left side of the above equation is generally Vander Mon
This is called a -de (7are del monde) matrix.

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

(1) Calculate the sample autocorrelation coefficient.

(2) Define At using the inverse Chebyshev coefficient (3)
Pjn) is obtained by solving the Hankel (/Nk) matrix equation by At.

(4) Find n Ri by solving an n-dimensional algebraic equation with pj'' as a coefficient.

x” (”) s”-” (n) Jn-m +
, , , , +Pn (J) :E +p, -8
+pn-2p"'x+p=.

(5) (2) Perform inverse transformation to obtain CSM angular frequency (town).

11 ωi-Maki xi (6) N'an d@r Monde (77y Del Monde) Solve the matrix equation to obtain the CSM intensity (-,).

By executing each step above, each angular frequency of the CSM ('1 #'!...&1In) and the strength of the can liquid (N t t m x t...-) are determined. be able to.

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 Nieton-2Pso/method or the like.

Further, as an efficient method for solving the Vand@r Monde (77 Nder Monde) matrix equation, it is possible to use a method of performing triangular matrix formation and obtaining a solution in the first order.

In the 90 SM analysis described above, this example uses a method of solving an equation in which the sample autocorrelation coefficient and the CSM autocorrelation coefficient are equal.
It is also possible to use a method of calculating the line spectrum frequency by making the coefficients lossless and calculating the number of tones.

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.

In the above embodiments, mi was used as the quantity that specifies the intensity (power amplitude) of each sine wave of the CSM, but mi, which is proportional to the amplitude, is used as the signal that actually controls the variable gain amplifier. It is clear that this may be done.

(Effects of the Invention) As described above, by using the present invention, it is possible to provide a new type of confidential communication device that transmits analog signals based on the principle of speech analysis and synthesis using CAM. This privacy device has relatively low requirements for the quality of the transmission path, and is characterized by being able to provide high privacy.

Furthermore, this confidential communication device also has the effect of band pressure line transmission of voice signals.

[Brief explanation of drawings]

Fig. 1 is a diagram showing an example of a speech feature vector pattern based on CSM parameters, Fig. 2 is a diagram showing an example of the correspondence between the C8J inspectrum and the LPC spectrum envelope obtained from the same speech sample, and Fig. 3 (A ) is a diagram showing the spread CSM spectral envelope and pitch fine structure, FIG. 3(B) is a diagram showing a CSM spectrum obtained by simple addition, and FIG. 4 is a block diagram showing an embodiment of the present invention. Figure 5 (6) is a diagram showing the functional form of the variable length window function, Figure 5 (B)
is a diagram showing the relative time relationship between the variable length window function and the phase reset pulse, FIG. 6 is a diagram showing an example of a circuit of a variable frequency root generator with a phase reset function, and FIG. 7 is a diagram showing a circuit example of a variable frequency root generator with a phase reset function. FIG. 8 is a diagram showing an example of a circuit of a random number generator. FIG. 9 (A) is a block diagram of a period calculator.
FIG. 10B is a diagram showing the random number distribution of the output of the period calculator, and FIG. 10 is a block diagram showing an example of the variable length window function generator. In the figure, 1... transmitting side, 2... receiving side, 101... A/D converter, 02...
... Hamming window processor, 103 ... Autocorrelation coefficient measuring device, 104 ... CSM analyzer, 105
--------Pitch-V/UV analyzer, 106...
...Parameter converter, 107-1 to 107-n...
...Variable frequency oscillator, 108-1 to 108-n...
...Variable gain amplifier, 109...Summing combiner, 11o...Variable gain amplifier, 111...
...Variable frequency oscillator, 112...V/UV
Switch, 113, old... Addition synthesizer, 2o1...
... Spectrum analyzer, 202 ... Power separator, 203 ... Parameter inverter, 204-1
~204-n... Variable frequency oscillator with phase reset function, variable gain amplifier for 205-1 to 205-n, 206... Addition synthesizer, 207...
... Multiplier, 208 ... Multiplier, 209 ...
...V/UV switch, 21o...Variable length window function generator, 21st...Period calculator, 212...
...Random number generator. Fig. 2 Hari] fL number k Torashiwa @ k Yuru

Claims (1)

[Scope of Claims] A privacy device for polarizing voice communication includes: a CSM analysis means; a means for converting a distribution range of CSM frequency data calculated by the means at a constant ratio; A transmitting side comprising a CSM spectrum generating means for generating an additive composite waveform of a plurality of sine waves having a frequency and amplitude specified by CSM data, and performing spectrum analysis of a signal from the CSM spectrum generating means on the transmitting side. spectral paratota extracting means for extracting each paratota specifying the frequency and amplitude of each of the plurality of additively synthesized sine waves; and means for inversely transforming the extracted parameters at the transmitting side; A plurality of variable frequency oscillators with a phase reset function are set to frequencies specified by the inversely converted parameters, and correspondingly outputs of the variable frequency oscillators with a phase reset function are specified by the extracted parameters. a variable gain amplifier, a variable length window function generator, and a random number generator to set the amplitude to When resetting and synthesizing unvoiced sound, the phase of each variable frequency oscillator with a phase reset function is reset in accordance with the period calculated from the random number output from the random number generator, and the variable length window function generator and a reception side, wherein the start and end times of the window function generated in the above-mentioned phase reset substantially coincide with the time of the phase reset.