CN111131141A - Voice encryption and decryption method and device - Google Patents
Voice encryption and decryption method and device Download PDFInfo
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- CN111131141A CN111131141A CN201910939850.0A CN201910939850A CN111131141A CN 111131141 A CN111131141 A CN 111131141A CN 201910939850 A CN201910939850 A CN 201910939850A CN 111131141 A CN111131141 A CN 111131141A
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- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04L—TRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
- H04L63/00—Network architectures or network communication protocols for network security
- H04L63/04—Network architectures or network communication protocols for network security for providing a confidential data exchange among entities communicating through data packet networks
- H04L63/0428—Network architectures or network communication protocols for network security for providing a confidential data exchange among entities communicating through data packet networks wherein the data content is protected, e.g. by encrypting or encapsulating the payload
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- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04L—TRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
- H04L65/00—Network arrangements, protocols or services for supporting real-time applications in data packet communication
- H04L65/80—Responding to QoS
Abstract
The invention discloses a voice encryption and decryption method and device, and belongs to the field of signal processing. The voice encryption method comprises the following steps: a transmitting end samples an analog voice signal to be encrypted to obtain a continuous digital signal; periodically intercepting a digital signal sequence with a preset length from the continuous digital signal at intervals of preset duration according to a sampling time sequence; respectively carrying out interleaving encryption processing on the frequency spectrums of the digital signal sequences, wherein amplitude information in the frequency spectrums of the digital signal sequences is not changed during the interleaving encryption processing; respectively constructing the digital signal sequence subjected to the interweaving encryption processing into corresponding encrypted continuous digital signals, wherein the encrypted continuous digital signals are matched with sampling frequency; and respectively converting each encrypted continuous digital signal into an analog signal and then sending the analog signal to a receiving end. The invention can avoid synchronization in the process of encrypting voice communication, reduce the calculation complexity and improve the effect of encrypting communication.
Description
Technical Field
The present invention relates to the field of signal processing, and in particular, to a voice encryption and decryption method and apparatus.
Background
Voice encryption technology is commonly used for telephony services. In the process of encrypting the call service, the sending party generally uses a fixed frame length to perform framing and encrypt each frame of data, and encryption parameters are changed along with the difference of time or communication nodes (designated as receiving parties). Correspondingly, in the process of decrypting the call service, the receiver needs to decrypt the data frame at the same framing position according to the consistent parameters in order to ensure the correct demodulation of the data frame, so the encryption method has higher requirements on the signal frame synchronization performance of the transmitter and the receiver. However, the synchronization between the transmitter and the receiver is also disrupted by the channel difference due to the different environments in which the transmitter and the receiver are located and the jitter of the synchronization position due to the time drift of the crystal oscillators of the transmitter and the receiver. When the signal frame of the receiving party is out of step, the quality of the decrypted voice is reduced, even the decrypted voice cannot be identified. Therefore, in the process of encrypted call, a synchronization process needs to be continuously performed to ensure the quality of encrypted call.
In the process of implementing the invention, the inventor finds that the prior art has at least the following problems:
adding a synchronization process increases the amount of computation during the call and results in a reduction in call efficiency. There is a conflict between the accuracy of the synchronization operation and the validity of the encrypted call.
Disclosure of Invention
The embodiment of the invention provides a voice encryption and decryption method and device, which can be used for reducing the calculation complexity, improving the encryption communication effect and obtaining higher communication quality without synchronization in the process of encrypting voice communication.
The technical scheme is as follows:
in a first aspect, a voice encryption method is provided, where the voice encryption method includes:
a transmitting end samples an analog voice signal to be encrypted to obtain a continuous digital signal;
periodically intercepting a digital signal sequence with a preset length from the continuous digital signal at intervals of preset duration according to a sampling time sequence;
respectively carrying out interleaving encryption processing on the frequency spectrums of the digital signal sequences, wherein amplitude information in the frequency spectrums of the digital signal sequences is not changed during the interleaving encryption processing;
respectively constructing the digital signal sequence subjected to the interweaving encryption processing into corresponding encrypted continuous digital signals, wherein the encrypted continuous digital signals are matched with sampling frequency;
and respectively converting each encrypted continuous digital signal into an analog signal and then sending the analog signal to a receiving end.
Optionally, the periodically intercepting a digital signal sequence of a preset length from the continuous digital signal at intervals of a preset time duration according to the sampling time sequence includes:
at the nth moment, windowing the continuous digital signal by adopting a low-pass filter to obtain a digital signal sequence with a preset length,
the N moment and the N-1 moment are distant from the preset duration, the preset duration comprises R sampling periods, the preset length comprises a sampling point at the N moment and 2LN-1 continuous sampling points which are before and closest to the N moment, the order of the low-pass filter is matched with the preset length, L is a positive integer larger than 1, R and N are positive integers, R is smaller than or equal to N, the frequency spectrum of the digital signal sequence is obtained through Discrete Fourier Transform (DFT), and the number of the DFT conversion points is N.
Optionally, the separately interleaving and encrypting the frequency spectrums of the digital signal sequences includes:
equally dividing the digital signal sequence intercepted at the nth moment into 2L vectors with the length of N, and adding elements at corresponding positions in the vectors to obtain a para-position superposed sequence at the nth moment;
and carrying out interleaving encryption processing on the frequency spectrum of the bit superposition sequence at the nth moment.
Optionally, the interleaving encryption processing on the spectrum of the bit-superposed sequence at the nth time includes:
calculating the frequency spectrum of the bit-aligned superposition sequence at the nth time by adopting DFT,
determining the number of spectrum points to be interleaved at the nth moment from the spectrum of the alignment superposition sequence at the nth moment, wherein the number of the spectrum points to be interleaved is less than half of N,
determining the interleaving frequency band of the nth moment based on the number of the frequency spectrum points to be interleaved at the nth moment,
constructing the interleaved and encrypted frequency spectrum at the nth time based on the interleaved frequency band at the nth time, wherein the interleaved and encrypted frequency spectrum at the nth time comprises the interleaved frequency band,
and transforming the interleaved and encrypted frequency spectrum at the nth time into the encrypted digital signal sequence at the nth time by using Inverse Discrete Fourier Transform (IDFT).
Optionally, the respectively constructing the digital signal sequences after the interleaving encryption processing into corresponding encrypted continuous digital signals includes:
converting the encrypted digital signal sequence at the nth time into a prepared continuous digital signal at the nth time in a zero padding mode, wherein the prepared continuous digital signal at the nth time is matched with a target frequency, and the target frequency is R times of the sampling frequency,
and performing interpolation filtering processing on the prepared continuous digital signal at the nth time through an interpolation filter to generate an encrypted continuous digital signal at the nth time, wherein the encrypted continuous digital signal at the nth time is matched with a sampling frequency, and the interpolation multiple of the interpolation filter is R.
In a second aspect, a speech decryption method is provided, the speech decryption method comprising:
the receiving end samples the encrypted analog voice signal to obtain an encrypted continuous digital signal;
periodically intercepting a digital signal sequence with a preset length from the encrypted continuous signal at intervals of preset duration according to a sampling time sequence;
respectively carrying out de-interleaving processing on the frequency spectrums of the digital signal sequences;
respectively constructing the digital signal sequences subjected to the deinterleaving processing into corresponding decrypted continuous digital signals, wherein the frequency of the encrypted continuous digital signals is matched with the frequency of the decrypted continuous digital signals;
and respectively converting each decrypted continuous digital signal into an analog signal and outputting the analog signal.
Optionally, the periodically intercepting a digital signal sequence of a preset length from the encrypted continuous signal at intervals of a preset time duration according to the sampling time sequence includes:
at the nth moment, windowing the encrypted continuous digital signal by adopting a low-pass filter to obtain a digital signal sequence with a preset length,
the N moment and the N-1 moment are distant from the preset duration, the preset duration comprises R sampling periods, the preset length comprises a sampling point at the N moment and 2LN-1 continuous sampling points which are before and closest to the N moment, the order of the low-pass filter is matched with the preset length, L is a positive integer larger than 1, R and N are positive integers, R is smaller than or equal to N, the frequency spectrum of the digital signal sequence is obtained through Discrete Fourier Transform (DFT), and the number of the DFT conversion points is N.
Optionally, the separately deinterleaving the frequency spectrums of the digital signal sequences includes:
equally dividing the digital signal sequence intercepted at the nth moment into 2L vectors with the length of N, and adding elements at corresponding positions in the vectors to obtain a para-position superposed sequence at the nth moment;
and performing de-interleaving processing on the frequency spectrum of the bit superposition sequence at the nth moment.
In a third aspect, a voice encryption apparatus is provided, where the voice encryption apparatus is applied in a transmitting end, and the voice encryption apparatus includes:
the analog-to-digital converter ADC is used for sampling an analog voice signal to be encrypted to obtain a continuous digital signal;
the low-pass filter is used for periodically intercepting a digital signal sequence with a preset length from the continuous digital signal at intervals of preset duration according to a sampling time sequence;
the encryption module is used for respectively carrying out interweaving encryption processing on the frequency spectrums of the digital signal sequences, and amplitude information in the frequency spectrums of the digital signal sequences is not changed during the interweaving encryption processing;
the interpolation filter is used for respectively constructing the digital signal sequence subjected to the interweaving encryption processing into corresponding encrypted continuous digital signals, and the encrypted continuous digital signals are matched with the sampling frequency;
the digital-to-analog converter DAC is used for respectively converting the encrypted continuous digital signals into analog signals;
and the sending module is used for sending the analog signals converted by the DAC to a receiving end.
In a fourth aspect, there is provided a speech decryption apparatus, which is applied in a receiving end, the speech decryption apparatus comprising:
the analog-to-digital converter ADC is used for sampling the encrypted analog voice signal to obtain an encrypted continuous digital signal;
the low-pass filter is used for periodically intercepting a digital signal sequence with a preset length from the encrypted continuous signal at intervals of preset duration according to a sampling time sequence;
the decryption module is used for respectively carrying out de-interleaving processing on the frequency spectrums of the digital signal sequences;
the interpolation filter is used for constructing a decrypted continuous digital signal based on each digital signal sequence subjected to de-interleaving processing, and the frequency of the encrypted continuous digital signal is matched with the frequency of the decrypted continuous digital signal;
the digital-to-analog converter DAC is used for respectively converting each decrypted continuous digital signal into an analog signal;
and the sending module is used for outputting the analog signal converted by the DAC.
The technical scheme provided by the embodiment of the invention has the following beneficial effects:
sampling an analog voice signal through a sending terminal to obtain a digital signal, periodically intercepting a digital signal sequence with a preset length from the continuous digital signal at intervals of preset duration according to a sampling time sequence, respectively carrying out interweaving encryption processing on frequency spectrums of the digital signal sequences, and respectively constructing the digital signal sequence subjected to the interweaving encryption processing into a corresponding encrypted continuous digital signal, wherein the encrypted continuous digital signal is matched with a sampling frequency; finally, converting each encrypted continuous digital signal into an analog signal and then sending the analog signal to a receiving end; because the amplitude information in the frequency spectrum of the digital signal sequence is not changed during the interleaving encryption processing, the receiving end can recover the amplitude information of the voice signal by carrying out corresponding de-interleaving processing on the basis of the encrypted analog signal, so that when the receiving end and the transmitting end are asynchronous, the data decrypted by the receiving end according to the encryption step corresponding to the transmitting end and the original data only have a linear phase error possibly, and human ears are insensitive to the phase error, therefore, the recovery of the encrypted voice signal can be accurately finished after the amplitude information of the voice signal is recovered, the method does not need the transmitting end and the receiving end to be synchronous, and therefore, compared with an encryption and decryption mode of increasing a synchronous flow, the method can reduce the calculation complexity, improve the encryption conversation effect and obtain higher conversation quality.
Drawings
In order to more clearly illustrate the technical solutions in the embodiments of the present invention, the drawings needed to be used in the description of the embodiments will be briefly introduced below, and it is obvious that the drawings in the following description are only some embodiments of the present invention, and it is obvious for those skilled in the art to obtain other drawings based on these drawings without creative efforts.
FIG. 1 is a schematic diagram of an exemplary application scenario provided by an embodiment of the present invention;
fig. 2 is a flow chart of a voice encryption method according to an embodiment of the present invention;
fig. 3 is a flow chart of a voice decryption method provided by an embodiment of the present invention;
FIG. 4 is a schematic diagram of an interleaved encryption process provided by an embodiment of the present invention;
FIG. 5 is a schematic diagram of a de-interleaving process provided by an embodiment of the present invention;
fig. 6 is a block diagram of a voice encryption apparatus according to an embodiment of the present invention;
fig. 7 is a block diagram of a voice decryption apparatus according to an embodiment of the present invention.
Detailed Description
In order to make the objects, technical solutions and advantages of the present invention more apparent, embodiments of the present invention will be described in detail with reference to the accompanying drawings.
In this embodiment, the sending end or the receiving end may be a call terminal, such as an intercom, a mobile phone, and the like. The application scenario may be an analog voice communication scenario such as a short wave/ultra short wave voice communication system. Fig. 1 is a schematic diagram of an exemplary application scenario provided by an embodiment of the present invention. Referring to fig. 1, the transmitting end may be a first intercom 10, the receiving end may be a second intercom 20, the first intercom 10 encrypts the clear conversation x (n) to obtain a secret conversation s (n), and transmits the secret conversation s (n) to the second intercom 20 through a communication line, and the second intercom 20 decrypts the clear conversation x (n) after receiving the secret conversation s (n) to complete the encrypted conversation.
Fig. 2 is a flowchart of a voice encryption method according to an embodiment of the present invention. The execution body of the voice encryption method may be a transmitting end. Referring to fig. 2, the process flow includes the following steps.
And 103, the transmitting end respectively carries out interleaving encryption processing on the frequency spectrums of the digital signal sequences, and amplitude information in the frequency spectrums of the digital signal sequences is not changed during the interleaving encryption processing.
And step 104, the transmitting end respectively constructs each digital signal sequence subjected to the interleaving encryption processing into a corresponding encrypted continuous digital signal, and the encrypted continuous digital signal is matched with the sampling frequency.
And 105, the sending end converts each encrypted continuous digital signal into an analog signal and sends the analog signal to the receiving end.
Fig. 3 is a flowchart of a voice decryption method according to an embodiment of the present invention. The voice decryption method and the voice encryption method are the same inventive concept, and the execution main body of the voice decryption method can be a receiving end. Referring to fig. 3, the process flow includes the following steps.
In step 205, the receiving end converts each decrypted continuous digital signal into an analog signal and outputs the analog signal.
In this embodiment, an analog voice signal is sampled by a sending end to obtain a digital signal, a digital signal sequence with a preset length is periodically intercepted from a continuous digital signal at intervals of a preset duration according to a sampling time sequence, then, the frequency spectrum of each digital signal sequence is respectively subjected to interleaving encryption processing, then, each digital signal sequence subjected to interleaving encryption processing is respectively constructed into a corresponding encrypted continuous digital signal, and the encrypted continuous digital signal is matched with a sampling frequency; finally, converting each encrypted continuous digital signal into an analog signal and then sending the analog signal to a receiving end; because the amplitude information in the frequency spectrum of the digital signal sequence is not changed during the interleaving encryption processing, the receiving end can recover the amplitude information of the voice signal by carrying out corresponding de-interleaving processing on the basis of the encrypted analog signal, so that when the receiving end and the transmitting end are asynchronous, the data decrypted by the receiving end according to the encryption step corresponding to the transmitting end and the original data only have linear phase errors possibly, and human ears are insensitive to the phase errors, the recovery of the encrypted voice signal can be accurately finished after the amplitude information of the voice signal is replied, the method does not need the transmitting end and the receiving end to be synchronous, and therefore, compared with an encryption and decryption mode of increasing a synchronous flow, the method can reduce the calculation complexity, improve the encryption conversation effect and obtain higher conversation quality.
The steps 101 and 201 may be implemented by an Analog-to-Digital Converter (ADC), and the steps 105 and 205 may be implemented by a Digital-to-Analog Converter (DAC). The steps 102 and 202 can be implemented by using a low-pass filter. The encryption process of step 103 may apply Discrete Fourier Transform (DFT) and the decryption process of step 203 may apply Inverse Discrete Fourier Transform (IDFT). The steps 104 and 204 can be implemented by using an interpolation filter.
Specifically, in order to facilitate the encryption and decryption of the signal, parameters need to be determined according to the encrypted call effect, including the transform point value of DFT and the interpolation multiple of the interpolation filter, to further determine the orders of the low-pass filter and the interpolation filter. And simultaneously, the design of the two filters is completed according to the bandwidth requirements of the filters.
In the design of the low-pass filter, the bandwidth of the low-pass filter is required to be fsN, wherein fsThe sampling frequency of the ADC or DAC (in this embodiment, the sampling frequency of the ADC and the DAC is the same), N represents the number of discrete fourier transform points; at the same time, the passband of the low-pass filter is required to be close to the ideal low-pass filter, and the stopband is 40dB lower than the passband.
The low-pass filter can be represented as hL＝{hL(0),hL(1),...,hL(2LN-1) }, wherein the order of the filter is 2LN, and L is set to be a positive integer greater than 1, and is usually 3-5. Based on the fact that the passband of the low-pass filter is close to the ideal low-pass filter, the coefficient h of the low-pass filterL(o) can be represented by (o is an integer, 0. ltoreq. o < 2 LN):
wherein, aLThe magnitude of the energy compensation for the low-pass filter coefficients according to N can be expressed as:
in the design of the interpolation filter, the bandwidth of the interpolation filter is required to be fsand/N, the stop band is 40dB lower than the pass band, and the top end of the filter is as flat as possible. The interpolation filter can be denoted as hI＝{hI(0),hI(1),...,hI(2QR-1), the interpolation multiple is R, R is less than or equal to N, N represents the number of discrete Fourier transform points, the order is 2QR orders, Q is a positive integer which is more than 1, and 3-5 is selected according to the situation. Interpolation filter coefficient hI(p) can be represented by 0 ≦ p < 2QR (p is a positive integer):
wherein, aIAccording to RThe magnitude of the energy compensation for the interpolation filter coefficients can be expressed as:
the following describes the steps of the above method in detail from the encrypted part and the decrypted part, respectively.
First, encrypted part
In step 101, the analog voice signal to be encrypted is sampled by the ADC to obtain a digital voice signal x (n) with a sampling frequency fs。
Illustratively, step 102 may include: and at the nth moment, windowing the sampled continuous digital signals by adopting a low-pass filter to obtain a digital signal sequence with the preset length at the nth moment. The distance between the nth time and the (n-1) th time is a preset time, the preset time comprises R sampling periods (the interpolation multiple is also R), and the preset length comprises the sampling point of the nth time and 2LN-1 continuous sampling points which are before the nth time and are closest to the nth time. The order of the low-pass filter is matched with the preset length, the order of the low-pass filter is 2LN, L is a positive integer larger than 1, R and N are both positive integers, R is smaller than or equal to N, the frequency spectrum of the digital signal sequence is obtained through Discrete Fourier Transform (DFT), and the number of transform points of the DFT is N.
Based on the preset duration of R sampling periods, assuming that the nth time is the starting time of the rR (R is a non-negative integer) sampling periods, the designed low-pass filter H is utilized at the n timeLWindowing the sampled signal sequence x (n), calculating the windowed sequence a at that momentn(i) (preset length of digital signal sequence):
an(i)＝x(n-2LN+i)hL(2LN-i),0≤i＜2LN,n＝rR
where x (n-2LN + i) represents the input digital voice signal amplitude (i is an integer) at the n-2LN + i sample point before the sample point at the n-th time, hL(2LN-i) 2LN-i coefficient representing a low pass filter; a isn(i) After corresponding multiplication of low-pass filter coefficients for the input speech sequenceThe result, representing the windowed signal sequence, is a column of vectors of length 2 LN.
Here, n ≠ rR, r is a non-negative integer, and when n ≠ rR, it is intended to reduce the amount of calculation. Because under the condition of R < ═ N, due to the effect of the passband (pi/N) of the front low-pass filter, based on the sampling theorem, the sequence generated by taking a value once every R sampling period can restore all information of the original sequence (including all sampling points of the R sampling periods), namely, the intercepted sequence can still completely describe all information of the original sequence, in order to reduce the calculation amount, the sampling points between the sampling time R (the starting time of the R-th sampling period) and the 2R (the starting time of the 2R-th sampling period) can not be calculated, namely, the calculation of the section (R +1 to 2R-1) is omitted.
Illustratively, step 103 may include the following steps.
Step 103a, equally dividing the digital signal sequence intercepted at the nth time into 2L vectors with the length of N (N represents the number of discrete Fourier transform points), and adding elements at corresponding positions in each vector to obtain a positioning superposition sequence at the nth time.
A is ton(i) The sequence is equally divided into 2L vectors with the length of N, and the values with the same index in each vector are added to the phase to obtain un(q)：
In the formula, an(q + rN) represents an(i) Wherein q is the result of i taking the remainder of N, and represents the position of the ith point spectrum in the range of [0,2 pi), and q is more than or equal to 0 and less than N; x (n-Nr + q) represents the input digital voice signal amplitude at time (n-Nr + q); h isL(Nr-q) represents the (Nr-q) th coefficient of the ideal low-pass filter; u. ofn(q) is a vector of length N.
And 103b, carrying out interleaving encryption processing on the frequency spectrum of the bit superposition sequence at the nth moment.
Wherein step 103b may comprise the following steps.
Firstly, calculating the frequency spectrum of the alignment superposition sequence at the nth time by adopting DFT (Discrete Fourier Transform), wherein the number of Transform points of DFT is N.
The signal spectrum Y at the nth time can be calculated by discrete fourier transform according to the following formulan(k)：
{Yn(k),0≤k＜N}＝DFT{un(mod(q-n,N)),0≤q＜N}
In the formula, Yn(k) Indicating the frequency spectrum of the bit-aligned superposition sequence at the nth moment, and DFT indicating discrete Fourier transform; mod (q-N, N) denotes that q-N takes the remainder of N, 0 ≦ ((q-N))N＜N；un(mod (q-N, N)) is a vector of length N, embodied as vector u in step 103an(q) as a result of performing a cyclic shift, when the time scale N is an integer multiple r of N, (q-N) as a result of taking the remainder of N to be q, in which case a shift operation is not necessary; y isn(k) The spectrum of N points corresponding to the signals at the N moments is represented, and the spectrum characteristics of the sampling point signals corresponding to the current moment and the previous 2LN-1 sampling point signals can be embodied.
And secondly, determining the number of frequency spectrum points to be interleaved at the nth moment from the frequency spectrum of the alignment superposition sequence at the nth moment, wherein the number of the frequency spectrum points to be interleaved is less than half of N.
Wherein, assume ICFor the number of spectral points to be interleaved, the frequency bandwidth f of the speech signalxAnd the sampling rate fsDetermination of IC＞N·fx/fs(ii) a Due to the fact that the DFT calculation result has a conjugate symmetry relation, 0 is less than IC＜N/2。
And thirdly, determining the interleaving frequency band at the nth moment based on the number of the frequency spectrum points to be interleaved at the nth moment.
The number of the frequency spectrum points to be interleaved at the nth moment Is 0 to (N-1), the interleaved frequency band Is Ic frequency points (the number of the frequency spectrum points Is to (Is + Ic)) from Is and a frequency band which Is symmetrical about N/2, and only the two frequency bands are in the interested effective frequency band of the voice. Also, because the values of other bins are not of interest in the speech processing, they can be zeroed out.
And fourthly, constructing an interleaved and encrypted frequency spectrum at the nth moment based on the interleaved frequency band at the nth moment, wherein the interleaved and encrypted frequency spectrum at the nth moment comprises the interleaved frequency band.
In the fourth step, the signal spectrum Y is measured at the time of nn(k) And carrying out encryption processing, wherein the encryption operation can be completed by an interleaving method. In this embodiment, interleaving only changes the position of each frequency point data, and does not change its amplitude (ensuring that the amplitude information Is recovered when decrypting, and then the voice signal Is recovered), that Is, (Is to (Is + Ic)) and its part data symmetric about N/2 exchange positions with each other. Fig. 4 is a schematic diagram of an interleaving encryption process provided by an embodiment of the present invention. Referring to fig. 4, the upper row represents the portions Is to (Is + Ic) to be interleaved, and the lower row represents the positions of Is to (Is + Ic) after interleaving. The specific process is to input a sequence YnSequentially storing the part Is to be interleaved (Is + Ic) into an interleaved array, wherein the array coordinate of the corresponding storage input value in the interleaved array increases inv when the sequence count increases by 1, and the array coordinate value needs to be opposite to ICThe remainder is taken.
Wherein inv is an interleaving parameter, according to ICDetermining that the requirement of 1 < inv < I for ensuring the interweaving effectCAnd inv and ICAll factors except 1 are relatively prime and adopt a value close to IC2 interleaving parameter to increase the dispersion degree of frequency spectrum; i isSIndicating the position of the starting point of the spectrum to be interleaved, obviously with IC+IS< N/2, and I should be used to ensure the integrity of the voice spectrumSCorresponding frequency IS·fsN is lower than the starting position of the voice spectrum; i isinitIndicating the starting position of data put in the interleaving array, I is more than or equal to 0init＜IC；mod(i·inv+Iinit,Ic) Represents (I. inv + I)init) To ICTaking the result of the remainder; conj [. C]Representing a conjugate calculation; the frequency spectrum outside the frequency band to be interleaved does not contain the frequency domain information of voice, so the zero setting is directly carried out; { Yn' } denotes { YnAnd (6) interleaving and encrypting the result.
And fifthly, transforming the interleaved and encrypted frequency spectrum at the nth time into a digital signal sequence encrypted at the nth time by using IDFT (Inverse Discrete Fourier Transform).
The encrypted signal spectrum { Yn' is restored to a time domain signal yrR(k)：
{yrR(k),k＝0,1,...N-1}＝DFT-1{YrR(k),k＝0,1,...N-1}
Wherein DFT-1Indicating an IDFT.
Illustratively, step 104 may include the following steps.
And 104a, converting the encrypted digital signal sequence at the nth time into a prepared continuous digital signal at the nth time in a zero padding mode, wherein the prepared continuous digital signal at the nth time is matched with a target frequency, and the target frequency is R times of the sampling frequency.
And 104b, carrying out interpolation filtering processing on the prepared continuous digital signal at the nth time through an interpolation filter to generate an encrypted continuous digital signal at the nth time, wherein the encrypted continuous digital signal at the nth time is matched with the sampling frequency, and the interpolation multiple of the interpolation filter is R.
To match the sampling rate, a time-domain signal y is obtainedrR(k) Firstly, zero filling is carried out according to the R times of up-sampling rate (step 104a), and v is obtained after the zero filling is finishedn(k)。
In step 104a, v is obtained after zero paddingn(k) Is a matrix of N rows, k is its abscissa N is the ordinate, and N is rR to (R +1) R-1. Where the value of the nth-rR column is N points after the current time IDFT, and zero padding is performed at rR +1 to (R +1) R-1 in each row of the matrix, so that each row is incremented by R0 s.
vn(k) And then extracting corresponding data through an interpolation filter to obtain an encrypted complete time domain voice signal s (n). s (n) comprises s (rR), s (rR +1), … and s ((R +1) R-1), so that the length of s (n) is consistent with the sampling length (the number of sampling points obtained by R sampling periods), and the sampling rate of the input and output signals is consistent.
hI(n-m) represents the n-m-th coefficient of the interpolation filter in step S1; and s (n) is the voice signal after encryption processing.
The above encryption process is only taken as an example at the nth time, and the encryption process will be repeatedly performed at other times, for example, at the (n +1) th time, which is not described herein again.
Second, decryption part
Step 202 may include: and at the nth moment, windowing the encrypted continuous digital signal by adopting a low-pass filter to obtain a digital signal sequence with the preset length at the nth moment. The method comprises the steps that the distance between the nth moment and the (N-1) th moment is preset duration, the preset duration comprises R sampling periods, the preset length comprises a sampling point at the nth moment and 2LN-1 continuous sampling points which are before the nth moment and are closest to the nth moment, the order of a low-pass filter is matched with the preset length, the order of the low-pass filter is 2LN, L is a positive integer larger than 1, R and N are both positive integers, R is smaller than or equal to N, the frequency spectrum of a digital signal sequence is obtained through Discrete Fourier Transform (DFT), and the number of transform points of the DFT is N.
At the nth time (the initial time of rR sampling periods, r is a non-negative integer), the designed low-pass filter H is utilizedLWindowing a sequence s' (n) of sampled encrypted speech (encrypted speech) digital signals, calculating a windowed signal b at that momentn(i)，
bn(i)＝s'(n-2LN+i)hL(2LN-i),0≤i＜2LN
Wherein n is a time scale; s '(n-2LN + i) represents the amplitude of the input digital voice signal s' (n) at the corresponding time instant of n-2LN + i; bn(i) The result of multiplying the input speech sequence s' (n) by the low-pass filter coefficients is a series of arrays of length 2 LN. To reduce the amount of calculation, n ≠ rR, r is a non-negative integer, and counting is not performed when n ≠ rRAnd (4) calculating.
Illustratively, step 203 may include the following steps.
Step 203a, equally dividing the digital signal sequence intercepted at the nth time into 2L vectors with the length of N, and adding elements at corresponding positions in the vectors to obtain a positioning superposition sequence at the nth time.
B is ton(i) The sequence is equally divided into 2L vectors with the length of N, and the values with the same index in each vector are added to the phase to obtain wn(q)，
Where s' (n-Nr + q) represents the encrypted digital voice signal amplitude at time (n-Nr + q); w is an(q) is a vector of length N.
Step 203b, deinterleaves the spectrum of the bit-superposed sequence at the nth time.
Step 203b may include the following steps.
A, obtaining a frequency spectrum corresponding to the n-th time alignment superposition sequence through DFT,
{Mn(k),0≤k＜N}＝DFT{wn(mod(q-n),N),0≤q＜N}
in the formula, DFT represents discrete fourier transform; mod (q-N, N) represents that q-N takes the remainder of N, and mod (q-N, N) is more than or equal to 0 and less than N; w is an(mod (q-N, N)) is a vector of length N, embodied as vector w for step 203an(q) performing a cyclic shift, wherein when the time scale N is an integer multiple of N (q-N), the remainder of N is taken as q, and no shift operation is required; mn(k) Representing the N-point spectrum corresponding to the N-time signal.
B, obtained for n timen(k) Carry on the corresponding decryption process, this operation can be completed through the method of deinterleaving, fig. 5 is a schematic diagram of the deinterleaving process provided in the embodiment of the present invention. Referring to fig. 5, the upper row represents the portions Is to (Is + Ic) to be deinterleaved, and the lower row represents the positions of Is to (Is + Ic) after deinterleaving. The specific flow is that the input sequence { Mn(k) Store solutions in turnInterleaving array, and sequentially counting each time to increase inv when storing the coordinate value which needs to be paired with ICTaking the remainder; the array coordinates in the interleaved array corresponding to the stored input values are incremented by 1,
number of spectral points I in which interleaving is requiredCInterleaving parameter inv, and initial position I of frequency spectrum to be deinterleavedSData is put into the starting position I of the de-interleaving arrayinitWhen the parameters are consistent with the parameters selected in the step 103 during encryption, the transmitting and receiving parties complete negotiation on the encryption parameters before entering the encryption communication; the frequency spectrum outside the frequency band to be interleaved does not contain the frequency domain information of voice, so the zero setting is directly carried out; { Mn' } denotes MnThe result after deinterleaving (decryption).
C, the decrypted signal spectrum { Mn' restore to time domain signal mrR(k)：
{mrR(k),k＝0,1,...N-1}＝DFT-1{MrR(k),k＝0,1,...N-1}
Wherein DFT-1Representing an inverse discrete fourier transform.
In step 204, to match the sampling rate, the obtained time domain signal m is usedrR(k) And (4) firstly carrying out zero filling according to the sampling rate of R times, and then completing the decryption of the signal through an interpolation filter after completing the zero filling.
In the formula, hI(n-m) represents the n-m-th coefficient of the interpolation filter in step S1; x' (n) is a string of clear-text signals with the length of R recovered after decryption processing.
The decryption process is only performed at the nth time, and the decryption process is repeated at other times, for example, at the (n +1) th time, which is not described herein again.
As calculated at the nth time (the starting time of the rR-th sampling period), the obtained encrypted signal s (n) is a signal that contains information of all voice signals of 2LN sampling points that are continuous before the nth time and is summarized into a length of signal as long as the number of sampling points of the R sampling periods after interleaving, and is restored into a length of signal as long as 2LN sampling points after de-interleaving and is to be superimposed with a signal coming later — this means that amplitude information of the signal can be completely restored. If there is a synchronization error of i sampling periods between the transmitter and the receiver, the signal received by the receiver at a certain time is s' (rR + i), and the result of DFT after windowing by the low-pass filter and the result of DFT after windowing at the right time of synchronization only have a phase difference exp (j (2pi/N) i), but the human ear is insensitive to the phase difference. Thus, the recovery of the encrypted voice signal can be accomplished more accurately.
In the embodiment of the invention, an analog voice signal is sampled by a sending terminal to obtain a digital signal, a digital signal sequence with a preset length is periodically intercepted from a continuous digital signal at intervals of preset duration according to a sampling time sequence, then the frequency spectrums of all the digital signal sequences are respectively subjected to interweaving encryption processing, then all the digital signal sequences subjected to interweaving encryption processing are respectively constructed into corresponding encrypted continuous digital signals, and the encrypted continuous digital signals are matched with sampling frequency; finally, converting each encrypted continuous digital signal into an analog signal and then sending the analog signal to a receiving end; because the amplitude information in the frequency spectrum of the digital signal sequence is not changed during the interleaving encryption processing, the receiving end can recover the amplitude information of the voice signal by carrying out corresponding de-interleaving processing on the basis of the encrypted analog signal, so that when the receiving end and the transmitting end are asynchronous, the data decrypted by the receiving end according to the encryption step corresponding to the transmitting end and the original data only have linear phase errors possibly, and human ears are insensitive to the phase errors, the recovery of the encrypted voice signal can be accurately finished after the amplitude information of the voice signal is replied, the method does not need the transmitting end and the receiving end to be synchronous, and therefore, compared with an encryption and decryption mode of increasing a synchronous flow, the method can reduce the calculation complexity, improve the encryption conversation effect and obtain higher conversation quality.
Fig. 6 is a block diagram of a voice encryption apparatus according to an embodiment of the present invention, which is applied to a transmitting end, and referring to fig. 6, the voice encryption apparatus includes an ADC 61, a low-pass filter 62, an encryption module 63, an interpolation filter 64, a DAC 65, and a transmitting module 66.
And the ADC 61 is used for sampling the analog voice signal to be encrypted to obtain a continuous digital signal.
And a low-pass filter 62 for periodically intercepting a digital signal sequence of a preset length from the continuous digital signal at intervals of a preset duration in the sampling time sequence.
And the encryption module 63 is configured to perform interleaving encryption processing on the frequency spectrums of the digital signal sequences respectively.
And the interpolation filter 64 is used for respectively constructing each digital signal sequence subjected to the interleaving encryption processing into corresponding encrypted continuous digital signals, and the encrypted continuous digital signals are matched with the sampling frequency.
And a DAC 65 for converting each of the encrypted continuous digital signals into analog signals, respectively.
And a sending module 66, configured to send the analog signal converted by the DAC 65 to a receiving end.
Illustratively, the low-pass filter 62 is configured to, at an nth time, perform windowing on the continuous digital signal to obtain a digital signal sequence with a preset length, where the nth time is separated from an nth-1 th time by a preset time duration, the preset time duration includes R sampling periods, the preset length includes a sampling point at the nth time and 2LN-1 continuous sampling points before the nth time and closest to the nth time, an order of the low-pass filter matches the preset length, L is a positive integer greater than 1, R and N are both positive integers, R is less than or equal to N, a frequency spectrum of the digital signal sequence is obtained by discrete fourier transform DFT, and the number of transform points of the DFT is N.
Exemplarily, the encryption module 63 is configured to equally divide the digital signal sequence intercepted at the nth time into 2L vectors with a length of N, and add elements at corresponding positions in the vectors to obtain an alignment superposition sequence at the nth time; and carrying out interleaving encryption processing on the frequency spectrum of the bit superposition sequence at the nth moment.
Illustratively, the encryption module 63 is configured to calculate a spectrum of the bit-aligned superposition sequence at an nth time by using DFT, determine a number of spectrum points to be interleaved at the nth time from the spectrum of the bit-aligned superposition sequence at the nth time, where the number of spectrum points to be interleaved is less than half of N, determine an interleaved frequency band at the nth time based on the number of spectrum points to be interleaved at the nth time, construct an interleaved and encrypted spectrum at the nth time based on the interleaved frequency band at the nth time, where the interleaved and encrypted spectrum at the nth time includes the interleaved frequency band, and transform the interleaved and encrypted spectrum at the nth time into the encrypted digital signal sequence at the nth time by using inverse discrete fourier transform IDFT.
Illustratively, the interpolation filter 64 is configured to convert the encrypted digital signal sequence at the nth time into a preliminary continuous digital signal at the nth time by zero padding, where the preliminary continuous digital signal at the nth time matches a target frequency, the target frequency is R times the sampling frequency, and the preliminary continuous digital signal at the nth time is subjected to interpolation filtering processing by the interpolation filter to generate an encrypted continuous digital signal at the nth time, where the encrypted continuous digital signal at the nth time matches the sampling frequency, and an interpolation multiple of the interpolation filter is R.
Fig. 7 is a block diagram of a speech decryption apparatus according to an embodiment of the present invention, which is applied to a receiving end, and referring to fig. 7, the speech decryption apparatus includes an ADC 71, a low-pass filter 72, a decryption module 73, an interpolation filter 74, a DAC 75, and a transmission module 76.
The ADC 71 is configured to sample the encrypted analog voice signal to obtain an encrypted continuous digital signal.
And a low-pass filter 72 for periodically intercepting a digital signal sequence of a preset length from the encrypted continuous signal at intervals of a preset duration in accordance with the sampling time sequence.
And the decryption module 73 is configured to perform deinterleaving processing on the frequency spectrums of the digital signal sequences respectively.
And an interpolation filter 74 for constructing a decrypted continuous digital signal based on each digital signal sequence subjected to the deinterleaving processing, the frequency of the encrypted continuous digital signal matching the frequency of the decrypted continuous digital signal.
The DACs 75 are used for respectively converting each decrypted continuous digital signal into an analog signal.
And a sending module 76 for outputting the analog signal converted by the DAC.
Illustratively, the low-pass filter 72 is configured to, at an nth time, perform windowing on the encrypted continuous digital signal by using the low-pass filter to obtain a digital signal sequence with a preset length, where the nth time is separated from an nth-1 th time by a preset time, the preset time includes R sampling periods, the preset length includes a sampling point at the nth time and 2LN-1 continuous sampling points before the nth time and closest to the nth time, an order of the low-pass filter is matched with the preset length, L is a positive integer greater than 1, R and N are both positive integers, R is less than or equal to N, a frequency spectrum of the digital signal sequence is obtained by discrete fourier transform DFT, and a number of transform points of the DFT is N.
Illustratively, the decryption module 73 is configured to equally divide the digital signal sequence intercepted at the nth time into 2L vectors with a length of N, and add elements at corresponding positions in the vectors to obtain an alignment superposition sequence at the nth time; and performing deinterleaving processing on the frequency spectrum of the bit superposition sequence at the nth time.
Since voice signals (voice signals are transmitted in input and output) are transmitted between the encryption and decryption devices, the encryption and decryption devices can be voice terminals, such as smart phones, interphones and the like. In hardware implementation, the encryption apparatus may be implemented by using the first ADC, the first processor, the first DAC, and the sending module. The first processor is used to implement the functions of the low pass filter 62, the encryption module 63, and the interpolation filter 64 described previously. The decryption device may employ a receiving module, a second ADC, a second processor, and a second DAC. The second processor is used to implement the functions of the low pass filter 72, decryption module 73, and interpolation filter 74 described previously. The transmit module is responsible for transmitting the analog signal into a communication link (such as a telephone line). The final output of the decryption device is a time-continuous analog signal.
It should be noted that: the voice encrypting apparatus provided in the above embodiment only exemplifies the division of the above functional modules when encrypting the voice and the voice decrypting apparatus decrypts the voice, and in practical applications, the above functions may be distributed by different functional modules according to needs, that is, the internal structure of the device may be divided into different functional modules to complete all or part of the above described functions. In addition, the embodiments of the voice encryption method and the voice decryption method provided by the above embodiments belong to the same concept, the embodiments of the voice encryption device and the voice encryption method belong to the same concept, the embodiments of the voice decryption device and the voice decryption method belong to the same concept, and the specific implementation process thereof is detailed in the embodiments of the methods and will not be described herein again.
It will be understood by those skilled in the art that all or part of the steps for implementing the above embodiments may be implemented by hardware, or may be implemented by a program instructing relevant hardware, where the program may be stored in a computer-readable storage medium, and the above-mentioned storage medium may be a read-only memory, a magnetic disk or an optical disk, etc.
The above description is only for the purpose of illustrating the preferred embodiments of the present invention and is not to be construed as limiting the invention, and any modifications, equivalents, improvements and the like that fall within the spirit and principle of the present invention are intended to be included therein.
Claims (10)
1. A voice encryption method, characterized in that the voice encryption method comprises:
a transmitting end samples an analog voice signal to be encrypted to obtain a continuous digital signal;
periodically intercepting a digital signal sequence with a preset length from the continuous digital signal at intervals of preset duration according to a sampling time sequence;
respectively carrying out interleaving encryption processing on the frequency spectrums of the digital signal sequences, wherein amplitude information in the frequency spectrums of the digital signal sequences is not changed during the interleaving encryption processing;
respectively constructing the digital signal sequence subjected to the interweaving encryption processing into corresponding encrypted continuous digital signals, wherein the encrypted continuous digital signals are matched with sampling frequency;
and respectively converting each encrypted continuous digital signal into an analog signal and then sending the analog signal to a receiving end.
2. The voice encryption method according to claim 1, wherein said periodically intercepting a digital signal sequence of a preset length from the continuous digital signal at intervals of a preset duration in the order of sampling time comprises:
at the nth moment, windowing the continuous digital signal by adopting a low-pass filter to obtain a digital signal sequence with a preset length,
the N moment and the N-1 moment are distant from the preset duration, the preset duration comprises R sampling periods, the preset length comprises a sampling point at the N moment and 2LN-1 continuous sampling points which are before and closest to the N moment, the order of the low-pass filter is matched with the preset length, L is a positive integer larger than 1, R and N are positive integers, R is smaller than or equal to N, the frequency spectrum of the digital signal sequence is obtained through Discrete Fourier Transform (DFT), and the number of the DFT conversion points is N.
3. The voice encryption method according to claim 2, wherein said separately interleave-encrypting the spectrum of each of the digital signal sequences comprises:
equally dividing the digital signal sequence intercepted at the nth moment into 2L vectors with the length of N, and adding elements at corresponding positions in the vectors to obtain a para-position superposed sequence at the nth moment;
and carrying out interleaving encryption processing on the frequency spectrum of the bit superposition sequence at the nth moment.
4. The voice encryption method according to claim 3, wherein the interleaving encryption processing of the spectrum of the bit-superimposed sequence at the nth time comprises:
calculating the frequency spectrum of the bit-aligned superposition sequence at the nth time by adopting DFT,
determining the number of spectrum points to be interleaved at the nth moment from the spectrum of the alignment superposition sequence at the nth moment, wherein the number of the spectrum points to be interleaved is less than half of N,
determining the interleaving frequency band of the nth moment based on the number of the frequency spectrum points to be interleaved at the nth moment,
constructing the interleaved and encrypted frequency spectrum at the nth time based on the interleaved frequency band at the nth time, wherein the interleaved and encrypted frequency spectrum at the nth time comprises the interleaved frequency band,
and transforming the interleaved and encrypted frequency spectrum at the nth time into the encrypted digital signal sequence at the nth time by using Inverse Discrete Fourier Transform (IDFT).
5. The voice encryption method according to claim 4, wherein said separately constructing each interleaved encrypted processed digital signal sequence into a corresponding encrypted continuous digital signal comprises:
converting the encrypted digital signal sequence at the nth time into a prepared continuous digital signal at the nth time in a zero padding mode, wherein the prepared continuous digital signal at the nth time is matched with a target frequency, and the target frequency is R times of the sampling frequency,
and performing interpolation filtering processing on the prepared continuous digital signal at the nth time through an interpolation filter to generate an encrypted continuous digital signal at the nth time, wherein the encrypted continuous digital signal at the nth time is matched with a sampling frequency, and the interpolation multiple of the interpolation filter is R.
6. A voice decryption method, characterized in that the voice decryption method comprises:
the receiving end samples the encrypted analog voice signal to obtain an encrypted continuous digital signal;
periodically intercepting a digital signal sequence with a preset length from the encrypted continuous signal at intervals of preset duration according to a sampling time sequence;
respectively carrying out de-interleaving processing on the frequency spectrums of the digital signal sequences;
respectively constructing the digital signal sequences subjected to the deinterleaving processing into corresponding decrypted continuous digital signals, wherein the frequency of the encrypted continuous digital signals is matched with the frequency of the decrypted continuous digital signals;
and respectively converting each decrypted continuous digital signal into an analog signal and outputting the analog signal.
7. The voice decrypting method according to claim 6, wherein the periodically intercepting a digital signal sequence of a preset length from the encrypted continuous signal at intervals of a preset duration in the order of sampling time comprises:
at the nth moment, windowing the encrypted continuous digital signal by adopting a low-pass filter to obtain a digital signal sequence with a preset length,
the N moment and the N-1 moment are distant from the preset duration, the preset duration comprises R sampling periods, the preset length comprises a sampling point at the N moment and 2LN-1 continuous sampling points which are before and closest to the N moment, the order of the low-pass filter is matched with the preset length, L is a positive integer larger than 1, R and N are positive integers, R is smaller than or equal to N, the frequency spectrum of the digital signal sequence is obtained through Discrete Fourier Transform (DFT), and the number of the DFT conversion points is N.
8. The speech decryption method according to claim 7, wherein the separately deinterleaving the frequency spectrums of the respective digital signal sequences comprises:
equally dividing the digital signal sequence intercepted at the nth moment into 2L vectors with the length of N, and adding elements at corresponding positions in the vectors to obtain a para-position superposed sequence at the nth moment;
and performing de-interleaving processing on the frequency spectrum of the bit superposition sequence at the nth moment.
9. A voice encryption apparatus, wherein the voice encryption apparatus is applied in a transmitting end, the voice encryption apparatus comprising:
the analog-to-digital converter ADC is used for sampling an analog voice signal to be encrypted to obtain a continuous digital signal;
the low-pass filter is used for periodically intercepting a digital signal sequence with a preset length from the continuous digital signal at intervals of preset duration according to a sampling time sequence;
the encryption module is used for respectively carrying out interweaving encryption processing on the frequency spectrums of the digital signal sequences, and amplitude information in the frequency spectrums of the digital signal sequences is not changed during the interweaving encryption processing;
the interpolation filter is used for respectively constructing the digital signal sequence subjected to the interweaving encryption processing into corresponding encrypted continuous digital signals, and the encrypted continuous digital signals are matched with the sampling frequency;
the digital-to-analog converter DAC is used for respectively converting the encrypted continuous digital signals into analog signals;
and the sending module is used for sending the analog signals converted by the DAC to a receiving end.
10. A speech decryption apparatus, wherein said speech decryption apparatus is applied in a receiving end, said speech decryption apparatus comprising:
the analog-to-digital converter ADC is used for sampling the encrypted analog voice signal to obtain an encrypted continuous digital signal;
the low-pass filter is used for periodically intercepting a digital signal sequence with a preset length from the encrypted continuous signal at intervals of preset duration according to a sampling time sequence;
the decryption module is used for respectively carrying out de-interleaving processing on the frequency spectrums of the digital signal sequences;
the interpolation filter is used for constructing a decrypted continuous digital signal based on each digital signal sequence subjected to de-interleaving processing, and the frequency of the encrypted continuous digital signal is matched with the frequency of the decrypted continuous digital signal;
the digital-to-analog converter DAC is used for respectively converting each decrypted continuous digital signal into an analog signal;
and the sending module is used for outputting the analog signal converted by the DAC.
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