EP0729678A1 - Process and device for speech scrambling and unscrambling in speech transmission - Google Patents
Process and device for speech scrambling and unscrambling in speech transmissionInfo
- Publication number
- EP0729678A1 EP0729678A1 EP95900687A EP95900687A EP0729678A1 EP 0729678 A1 EP0729678 A1 EP 0729678A1 EP 95900687 A EP95900687 A EP 95900687A EP 95900687 A EP95900687 A EP 95900687A EP 0729678 A1 EP0729678 A1 EP 0729678A1
- Authority
- EP
- European Patent Office
- Prior art keywords
- signal
- phase
- complex
- preamble
- transmission
- Prior art date
- Legal status (The legal status is an assumption and is not a legal conclusion. Google has not performed a legal analysis and makes no representation as to the accuracy of the status listed.)
- Granted
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Classifications
-
- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04K—SECRET COMMUNICATION; JAMMING OF COMMUNICATION
- H04K1/00—Secret communication
-
- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04K—SECRET COMMUNICATION; JAMMING OF COMMUNICATION
- H04K1/00—Secret communication
- H04K1/006—Secret communication by varying or inverting the phase, at periodic or random intervals
Definitions
- the invention relates to a method and a device for speech concealment and unveiling during speech transmission or in devices for speech transmission which, on the one hand, and / or for digitizing a, with a front end unit for digitizing a speech signal and adapting a transmission signal to a predetermined transmission channel Reception signals and on the other hand are equipped to adapt the processed reception signal to a speech reproduction device.
- Frequency band inversion i.e. Exchange of high and low frequencies of the low-frequency spectrum to be transmitted with a fixed or variable split point (image frequency method).
- Ad 1. The same channels are generally used for the transmission of the digital data as for the unveiled language. Since these channels only provide a limited bandwidth, data reduction methods are necessary. After this (reduced) data has been reconstructed on the receiving side, it is not possible to reliably identify the speaker.
- Ad 2 For physiological reasons, the number and temporal length of the partial intervals can only be changed within narrow limits. This leads to a simple decipherability of the transmitted signal.
- transitions between interchanged subintervals cannot generally be reconstructed on the reception side in a phase-pure manner, so that a reduction in the signal quality can be achieved compared to the unveiled signal.
- Ad 3 For physiological reasons, the number and bandwidth of the spectral subintervals are strictly limited. This leads to a simple decipherability of the transmitted signal. Unavoidable bandwidth overlaps of the filters required for generating and reconstructing the partial spectra lead to a deterioration in the Transmission quality.
- Ad 4. Decryption of the transmitted signal is possible with relatively little technical effort.
- the residual intelligibility of the skewed signal is high; trained listeners can also listen to broadcasts without technical aids.
- Combinations of the different methods generally increase security against decryption; however, they also lead to a summation of the disadvantageous properties, such as a deterioration in the signal-to-noise ratio and limitation to a few simple constellations of transmission channels.
- the invention is therefore based on the object of providing a method and a device for disguising and unveiling speech during speech transmission which can be produced in a compact design as a module (which can also be retrofitted) and which is one compared to the known methods and devices Ensure significantly better security against interception and evaluation by third parties.
- the digitized speech signal is converted by a first complex input filter with a bandwidth that corresponds to the bandwidth of the transmission channel into a complex signal that is phase-modulated by means of a key signal controlled by pseudo random numbers,
- phase-modulated speech signal with a pilot signal likewise phase-modulated in pseudo-random distribution to form a veiled useful signal to be transmitted
- the useful signal in a sequential sequence together with a preamble serving for receiver-side synchronization and useful signal equalization, passes through a first complex output filter as a complex signal, which generates a real output signal, which after digital-to-analog conversion is output to a transmit signal processor, and that on the receiver side
- the digitized received signal is converted into a complex signal by a second complex input filter with a bandwidth that corresponds to the bandwidth of the transition channel,
- the veiled ' useful signal is separated from its phase-modulated pilot signal superimposed on the transmitter side by linking to the synchronized pilot signal generated on the receiver side, and
- phase-modulated, veiled digital voice signal obtained in this way is unveiled by inverse phase modulation by means of the key signal generated by the receiver and clock-controlled by the preamble, and as a complex signal passes through a second complex output filter, which generates a real output signal, which is digital Is given to a received signal processing.
- One aspect that is essential for the method according to the invention is that after the input-side digitization, both the transmission and the reception-side, complex filtering, preferably using a Hilbert filter, is carried out, which generates a complex signal from a real one, which is subject to a reduction in the sampling rate fen, the bandwidth of the respective complex filter corresponding to the reduced sampling rate. All operations essential for the further process then take place with the complex signals at a reduced clock frequency.
- the complex signal is preferably subjected to a sampling rate increase by inserting zeros into the data stream on both the transmitting and receiving sides.
- a subsequent complex filter preferably also a Hilbert filter, serves as an interpolation filter and generates a real signal with a sampling frequency corresponding to the channel bandwidth.
- a device for speech concealment and unveiling in devices for speech transmission, which, on the one hand, has a front-end unit for digitizing a speech signal and adapting a transmit signal to a predetermined transmission channel and / or for digitizing a received signal and for adapting the processed received signal to a speech ⁇ on the other hand, is characterized in that a key generator controlled by a (pseudo) random number generator acts on the transmission side of a digital phase modulator which phase modulates the digitized speech signal, - the phase-modulated speech signal with a one supplied by a pilot signal generator, also in Random distribution phase-modulated pilot signal is combined to form a useful signal, a preamble generator generates a preamble which serves for receiver-side synchronization and useful signal equalization and which via an actuated in a defined clock sequence switch is sequentially output signal processing together with the 'useful signal to the front-end unit for transmitting, and that emfangs furnish a digital equalizing filter for equalization of
- a device for detecting the preamble within the received useful signal, which, depending on a specified section of the preamble, triggers the calculation of the equalizer coefficients in a higher-level computing unit for the equalizer filter and then initializes the unveiling of the useful signal by activating a clock synchronization device which, on the one hand, from the received, demodulated pilot signal by complex multiplication with one generated on the receiving side
- Pilot signal a control signal for sampling clock correction and, on the other hand, under the control of a (pseudo) random number generator likewise initialized with the clock synchronization, delivers a phase-modulated pilot signal from the pilot signal supplied by the reception-side pilot signal generator via a modulator, which is linked to the equalized useful signal and then as a phase-modulated one Speech signal in a phase demodulator under the control of the synchronized, receiving-side random number generator is converted into the unmodulated, digital speech signal which is emitted to the front end unit for conversion into an audible signal.
- a modulator which is linked to the equalized useful signal and then as a phase-modulated one Speech signal in a phase demodulator under the control of the synchronized, receiving-side random number generator is converted into the unmodulated, digital speech signal which is emitted to the front end unit for conversion into an audible signal.
- FIG. 1 shows the block diagram of a speech concealment / unveiling module according to the invention, hereinafter referred to as "SV module";
- 2 shows the principle of obfuscation with an arbitrarily chosen course of time
- 3 shows the functional block diagram of the transmitting part of the SV module
- FIG. 5 shows the functional block diagram of the receiving section of the SV module
- FIG. 6 shows the block diagram of the signal processing on the transmission side of the SV module
- FIG. 7 shows the structure of an input-side (first) complex filter, preferably a Hilbert filter:
- FIG. 8 shows the frequency response of the input-side (first) complex filter according to FIG. 7;
- FIG. 9 shows the structure of a first complex output filter, preferably a Hilbert filter in the transmission part of the SV module;
- FIG. 10 shows the frequency response of the first complex output filter according to FIG. 9;
- FIG 11 shows the block diagram of the reception-side signal processing in the preamble detection phase (clear position);
- FIG. 13 shows an operating and functional flow diagram for the transmitter-side signal processing according to the block diagram of FIG. 6;
- FIG. 14 shows an operating and functional sequence program for the reception-side signal processing according to the block diagrams of FIGS. 12 and 12.
- the SV module essentially consists of a powerful, digital signal processor system and the peripheral components required for operation, combined with modern signal processing algorithms.
- the block diagram shown in FIG. 1 shows the components and assemblies important for digital signal processing. Functions such as power supply, clock generation, discrete inputs and analog input and output stages are not shown for the sake of clarity.
- the structure of the SV module according to FIG. 1 corresponds to a realized and functional prototype, which is still used in part to test and further improve the algorithm development.
- a targeted version of the series can itself be found in the block diagram representation.
- the description of this embodiment is by no means to be understood as the only possible embodiment of the invention. Rather - as can be recognized by the person skilled in the art - many modifications and changes are possible in all subareas and assemblies, both on the transmitter and receiver side, without leaving the scope of the technical teaching conveyed here.
- the essential signal processing unit is a signal processor 1, in which the processor type ADSP 21msp55 from Analog Devices is used, at least in the prototype version.
- This signal processor 1 already contains an AD converter 2 and a DA converter 3 with a resolution of 16 bits, for example, at a sampling rate of 8 kHz.
- separate RAM areas 2, 3 for data on the one hand (lk x 16) and program (2k x 24) on the other hand are integrated.
- the internal memory organization corresponds to the Harvard architecture, so that data access is possible in every command cycle in addition to the op code fetch. Without exception, all processor operations require one cycle. This means that a processing capacity of 13 MIPS (integer) is available.
- a mask-programmed variant of this processor (ADSP21msp56) is provided for series production and will additionally have a 2k x 24 bit ROM 6 on the program memory side. - ⁇ . -
- a further pair of AD / DA converters 8.9 is required for duplex operation. This is implemented by a converter module 7 of the type AD28msp02, which contains the converters identical to the signal processor 1 in a separate housing. The data transmission between the converter module 7 and the signal processor 1 takes place via fast serial interfaces.
- An EEPROM 10 is present as external memory, which stores loadable program parts as well as variables that are seldom to be changed, such as the key (further explanation below).
- the memory size here is 8k x 8 (series) or 32k x 8 (prototype) as indicated in FIG. 1.
- the state of a speech button, a squelch logic of a radio 11 and a crypt ON / OFF switch can be queried by the signal processor 1 on discrete input signals (not shown).
- the external EEPROM 10 can also be addressed as a data memory in order to be able to read and change variable parameters, such as the key.
- the program flow is structured in time by interrupts of the analog interfaces, which run freely with their specified conversion rate of 8 kHz and which trigger an interrupt each time the conversion has taken place.
- All functions of the SV module are implemented by digital signal processing. - / o -
- 3 shows the functional block diagram of the transmission part of the SV module:
- a key signal is generated in a key signal generator 23, with the aid of which the input signal of the microphone, i.e. the speech signal is obscured.
- a PTT button (not shown) is pressed, a so-called preamble generated in a preamble generator 24 is transmitted immediately before the veiled speech signal, which is illustrated by the three time-related partial diagrams in FIG. 2.
- the preamble is required for the synchronization of a further key signal generator 43 (cf. FIG. 5) and the setting of an equalizer 40 on the receiving side.
- the preamble is sent out periodically in a fixed time grid, with the prototype currently being tested every 5 seconds.
- the shifted voice signal is hidden for the duration of the preamble (currently approx. 200 ms).
- a pilot signal generator 20 supplies a special pilot signal which is additively linked to the disguised speech signal and which is used on the receiving side for synchronizing the sampling clock, as explained in more detail below.
- the front-end unit 22a / 22b shown in two sub-blocks takes care of the pre-processing of the analog input signal and conversion into a digital signal or the final processing of the encrypted voice signal on the transmission side and adaptation to the respective transmission device or the transmission channel. Further details are explained below.
- the beginning of a veiled transmission signal is - as can be seen in FIG. 4 - characterized by the preamble. For this reason, the reception signal is always analyzed on the reception side when the receiver is not in the unveiling mode. During this phase, the received signal is looped through the SV module unchanged. If the end of a preamble is recognized, the unveiling process is started with this recognition, ie the key generator 43 on the receiving side is started and the incoming useful signal is unveiled ("speech signal" in Fig. 4).
- FIG. 5 shows the functional block diagram of the receiving part of the SV module.
- the received signal is fed to a function block 44, the function of which is to recognize and analyze the received signal. If a preamble is received, the properties of the transmission channel and, therefrom, filter coefficients for an equalizer 51 at the receiving end are determined on the basis of this.
- an equalizer adapted to the transmission channel is available.
- the reception-side key generator 43 is started to uncover the useful signal.
- the sampling synchronization 55 evaluates the Pilto signal superimposed on the useful signal and separates it from the useful signal. The unveiled useful signal is then output.
- Fig. 6 shows a detailed block diagram of the signal processing on the transmission side in the case of obfuscation.
- the individual function blocks are described in more detail in the following subsections. All signal processing functions, which are illustrated verbally in the flowchart in FIG. 13, are implemented with the aid of one signal processor 1 (cf. FIG. 1).
- the double lines and arrows in FIG. 6 are intended to identify analytical signals. Real signals are represented by simple lines and arrows.
- Plain text operation is realized by a simple feedback on the digital side of the analog front end 22.
- the input-side analog front-end unit 22 has the task of level adjustment, sampling the analog input signal c (t), and converting it into a digital signal c (t>).
- the A / D converter part of the analog front end 22 (not shown in detail) consists of two analog input amplifiers and an A / D converter.
- the digitized input signal c (v) acts on a first complex input filter 30 to suppress the lower sideband.
- This filter 30 also ensures that the bandwidth of the input signal (digitized voice signal) is limited to a bandwidth that corresponds to that of the transmission channel, ie 2,667 kHz in the present exemplary embodiment.
- the complex first input filter 30 generates a complex output signal consisting of real and imaginary parts from a real input signal, with any real or imaginary part between any -ft-
- the first complex input filter (as well as the complex input filter on the receiving side; see below) is a higher-order Hilbert filter.
- This first Hilbert filter 30 on the input side is a recursive filter, the transfer function of which
- the input signal of this Hilbert filter 30 is the sampled, real received signal c (v).
- the recursive part of this filter has only real ones
- the transverse part has complex coefficients aj.
- This first Hilbert filter 30 is based on the design of an elliptical low-pass filter.
- the low pass is converted into a Hilbert band pass by a transformation in the frequency domain.
- the frequency response of the Hilbert filter 30 implemented in the prototype of the invention is shown in FIG. 8.
- the band-limited output signal d (v) of the first complex input filter acts on a function block designated as sampling rate reduction 31, in which the sampling clock by a certain, preferably integer factor, in the present exemplary embodiment by a factor of 3 to 2,667 kH ⁇ is reduced.
- sampling rate reduction 31 in which the sampling clock by a certain, preferably integer factor, in the present exemplary embodiment by a factor of 3 to 2,667 kH ⁇ is reduced.
- a suitable dimensioning of the first Hilbert filter 30 on the input side ensures that no aliasing effects occur.
- Hilbert filter 30 and sampling reduction 31 means that any frequency band with a bandwidth of 2,667 kHz can contain the complete useful information.
- the pilot signal generator 20 is used to generate a pilot signal q (n) which is used on the receiving side for clock tracking.
- the pilot signal arises from the phase modulation of the pilot signal described below.
- the (pseudo) random number generator 34 (cf. FIG. 6) as part of the key signal generator 23 has the task of generating equally distributed numbers in the range from 1 to 64, for example. These numbers are used to select random values from a field of 64 complex values (cf. block "data record” in FIG. 6).
- Two key signals z s (n), z p (n) are generated from the selected values, one of which (z s (n)) for
- Phase modulation of the useful signal and the second (Zp (n)) is used to generate the pilot signal q (n).
- the random number generator 34 implemented in the current embodiment of the invention is based on the linear congruence method.
- the random values r (n) are according to the regulation
- the starting value r (0) is generally unimportant, since with a suitable choice of the constant values a and c, all m possible values are generated before the random sequence is repeated.
- the random numbers generated are equally distributed in the range from 0 to (m-1).
- 6 bits of the respective random value r (n) are used further as a random number.
- 6 bits are used to generate random numbers for "scrambling" (the phase modulation) of the useful signal x (n) and 6 bits are used to generate random numbers for "scrambling” (the phase modulation) of the pilot tone p (n). used.
- the random number generator 34 thus delivers two random numbers r s (n) and r p (n) in each cycle.
- the random number generator 34 After each transmission of a preamble, the random number generator 34 is reinitialized with a fixed starting value x ⁇ 0).
- control values for the phase modulators 32 and 33 are represented by a data set of 64 complex values. Values are selected from this set by the random number generator 34, and a random signal for phase modulation is thus generated.
- the 64 complex values are used as the data record
- control or input values z s (n) and z p (n) ' all have them
- Amplitude "1" but have different phases.
- the random number-controlled phase modulators 32, 33 are explained in more detail below.
- phase modulator units 32 and 33 are required in the transmission part of the SV module (FIG. 6).
- a phase modulator 33 is required for concealing the useful signal x (n) by a key signal z s (n) supplied by the random number generator 34.
- the other phase modulator 32 is used to generate the pilot signal q (n) from the pilot tone p (n) supplied by the pilot tone generator using the other key signal z p (n). Since the
- each phase modulator 32, 33 performs a complex multiplication of the respective input signal -fe-
- the phase-modulated useful signal y (n) has a noise signal-like character.
- the information contained in the useful signal is completely distributed over a frequency band of 2,667 kHz width.
- phase modulation according to the invention bears a certain similarity to a 64-stage PSK modulation as used in digital transmission technology.
- the purpose is completely different:
- the phase of a carrier signal is switched in the sampling cycle (phase shift keying).
- the phase of the carrier signal thus contains the digital information to be transmitted.
- the phase of the carrier is determined on the receiving side at defined sampling times.
- a decision maker assigns the corresponding digital information to each determined phase and thus wins the transmitted message.
- phase modulation it is not the modulation signal but the signal to be modulated that carries the information to be transmitted.
- This information is predetermined by its quasi-continuous signal course.
- the phase modulation is only used to change the signal to be transmitted in such a way that the original signal curve can no longer be inferred. This makes a speech signal completely incomprehensible.
- the useful information is obscured by the phase modulation. On the receiving side, the useful information can be obtained by the operation inverse to equation (4)
- the received signal y (n) must match the (phase-modulated) transmit signal y (n).
- the modulation signal ie the key signal z s (n) must be known on the receiving side.
- the first requirement requires equalization of the transmission channel on the receiving side.
- the second requirement requires knowledge of the key signal and exact synchronization on the receiving side.
- the number of values of the key signal z s (n) is determined by the number of stages of the modulation (here 64), the number of possible values for x (n) and y (n) is determined by the word length in the signal processing. 15 agrees.
- the signal values of the generated pilot tone are designated p (n) and the signal values of the associated key signal are designated Z p (n), the signal values of the pilot signal result from the relationship
- the transmission signal In order to be able to transmit the analytical signal generated with the clock frequency of 2,667 kHz, the transmission signal must be adapted to the transmission channel. Since in the example shown the analog
- 30 front end 22 predetermined sampling frequency is 8 kHz, the sampling rate must first be increased to 8 kHz.
- the increase in the sampling rate by a factor of 3. i.e. from 2,667 kHZ to 8 "_ kHz, is achieved by inserting two signal values with the value 0 between two existing signal values, i.e.
- d s (v) ..., w (n - l), 0, 0, w (n), 0, 0, w ⁇ n + l), ... (7)
- the sampling rate is increased in connection with a first complex output filter 35 for adapting the analytical transmission signal to the transmission channel.
- the real part of the analytical output signal of this complex output filter 35 is fed to the analog front end 22.
- the first complex output filter 35 first generates an analytical signal from a complex input signal d s (v), the real and imaginary part of which is phase-shifted by 90 ° for any frequency, and from this a real output signal c s (v). At the same time spectral components outside the usable bandwidth of the transmission channel are suppressed.
- the output-side first complex filter 35 is preferably a (second) Hilbert filter, i.e. a recursive filter, the structure of which is shown in FIG. 9.
- the input signal d s (v) of this second Hilbert filter 35 is an analytical signal; the output signal c s (v), however, is a real signal.
- the design of the filter is based on the design of an elliptical low-pass filter.
- the low pass is then converted into a Hilbert band pass by a transformation in the frequency range.
- the preamble generator 24 is used to generate a preamble at the start of a transmission via radio or telephone channels. In order to enable switching to an ongoing transmission on the reception side, the generation of a preamble is triggered at fixed time intervals.
- the preamble used consists of two successive signal sections.
- the first signal section is a so-called CPFSK signal (Continuous Phase Frequency Shift Keying).
- the second section is a noise-like signal.
- the first part is used in the receiver to detect the preamble and to synchronize the receiver.
- the second signal part serves to equalize the transmission channel.
- the CPFSK signal is generated by the CPFSK modulation of a special data frequency.
- the length of this sequence is, for example, 240 bits.
- the transmission rate is 1,778 kbit / s.
- the structure of the data sequence is chosen so that a very reliable detection of the preamble is possible with a special method on the receiving side. For further details, reference is again made to the publication DE-Cl 41 08 806 (ref. [4]) and to ref. (5).
- the total length of the preamble in this example is approximately 230 ms.
- two different operating modes of the SV module can be distinguished. On the one hand, this is the phase of preamble detection, during which the SV module is in the clear position, and on the other hand, this is the unveiling phase.
- three types of signal processing can be distinguished, namely analog signal processing, digital signal processing in the 8 kHz cycle and digital signal processing in the cycle of 2,667 kHz.
- the calculation of the equalizer coefficients runs in the background without being connected to a specific sampling clock.
- Figure 11 illustrates the functional block diagram of signal processing.
- the received signal only passes through the analog front end 52 with its filter.
- the received signal remains essentially unaffected by the SV module.
- the sampled received signal (8 kHz sampling frequency, 16 bit word width) is fed to the preamble detection block 55 after filtering with a second complex input filter 40 on the reception side, in particular a third Hilbert filter (bandpass) and a sampling rate reduction 43 to 2,667 kHz. At the same time, the samples of the received signal are buffered in the buffer 41.
- the preamble detection block 55 automatically and very reliably detects the reception of the preamble. Notes are Ref. [4] (DE-Cl-41 08 806) and Ref. [5] refer to.
- the function and structure of the second complex input filter 40 essentially corresponds to that of the first complex input filter 30 described on the transmission side.
- the preamble detection has two functions: On the one hand, this is the detection of the reception of the preamble and the switchover to unveiling. On the other hand, the preamble provides an exact time reference. This is necessary for the initialization and synchronization of the unveiling process.
- the second section of the preamble ie the noise signal
- the impulse response or the coefficient set for the equalizer filter 51 is then calculated with the aid of an FFT (Fast Fourier Transformation) and a target spectrum present in the receiver and stored in the program RAM 5 (FIG. 1).
- FFT Fast Fourier Transformation
- FIG. 12 shows the signal processing in this phase.
- the flowchart for the functional subsequent steps of signal processing on the receiving side is illustrated in FIG. 14.
- the received signal is converted by the analog front end 52 into a digital signal with, for example, 8 kHz sampling frequency and 16 bit word width.
- This signal passes through the equalizer 51, whose task is to equalize the transmission channel, which will be explained in more detail below.
- an analytical signal with the sampling frequency 2,667 kHz is present.
- This signal s (n) consists of the veiled useful signal and the superimposed pilot signal.
- the pilot signal is a phase modulated signal as described above.
- the pilot signal is evaluated in the clock synchronization block 45 and separated from the useful signal.
- the useful signal is then decoupled by a phase demodulator (descrambler) 59.
- the function and structure of the second complex output filter 62 essentially corresponds to that of the first complex output filter 35.
- the evaluation of the pilot signal in the clock synchronization block 55 also provides a manipulated variable for the regulation of fluctuations in the sampling clock (clock correction).
- the regulation of the sampling clock is necessary due to the high demands on the synchronicity during the unveiling. lent. Fluctuations in the sampling clock are caused by sample scatter and drifts in the crystal oscillators used.
- the received-rate signal s (n) reduced in sampling rate passes through a phase demodulator (descrambler) 58.
- the output signal q (n) of this phase modulator 58 consists of a carrier signal component and a superimposed noise signal-like signal component which is generated by the useful signal.
- the carrier signal is converted into the DC signal position.
- an analytical DC signal is available, the real part of which is a measure of the level of the pilot signal and the imaginary part of which is used as a manipulated variable for controlling the sampling clock.
- a pilot signal q (n) is generated at the receiving end and subtracted from the receiving terminal (s).
- the generated pilot signal q (n) corresponds exactly to the received pilot signal, so that the useful signal is completely separated from the pilot signal by the subtraction.
- the signal y (n) obtained from the subtraction process is correct. except for a possibly superimposed interference signal, with the signal y (n) at the output of the phase modulator 33 on the transmission side (cf. FIG. 6).
- the phase modulator 57 and the two phase demodulators 58, 59 are controlled by two (pseudo) random number generators 54.
- a random number generator controls the phase modulator 57 and the phase demodulator 58 of the clock synchronization block 55, the other controls the phase modulator 59 for the unveiling of the useful signal y (n).
- the random number generators correspond to those on the transmission side; like the pilot signal generator 50, they are synchronized with the detection of a preamble to the received signal.
- the input section of the analog front end 52 is responsible for the level adjustment, the sampling of the analog received signal and the conversion into a digital signal.
- the AD28msp02 module from Analog Devices is again used as the analog front end 52 in the prototype implementation (cf. Ref. [3]). This module corresponds exactly to the analog front end used in the signal processor ADSP-21msp55.
- the analog front end 52 in turn consists of two analog input amplifiers, a switchable 20 dB preamplifier and an A / D converter.
- the equalizer 51 is used to equalize the frequency response of the transmission channel in the range of the transmission bandwidth of z. B. 300 Hz to 3 kHz.
- the transmission channel includes all modules from the first complex output filter 35 of the transmitting part to the second complex input filter 40 of the receiving part (both inclusive).
- the equalizer 51 is by eir. transversal digital filter realized with 128 steps.
- the transfer function is:
- the coefficients & are determined during the reception of a preamble.
- the second complex input filter 40 serves to suppress the lower sideband of the input signal and to limit the bandwidth of the input signal (received speech signal) to a bandwidth of approximately 2.66 kHz.
- the second complex input filter 40 (Hilbert filter) is a recursive filter, the structure of which corresponds to that of the first complex filter 30 on the input side, so that reference can be made to FIG. 7 in this respect.
- the input signal of the second complex input filter 40 is the real output signal c (v) of the equalizer 51.
- the design of this filter is based on the design of an elliptical low-pass filter.
- the low pass was converted into a Hilbert band pass by a transformation in the frequency domain.
- a sampling rate reduction 43 is carried out in the receiving part to reduce the sampling clock in the example shown by a factor of "3" to 2,667 kHz.
- a suitable dimensioning of the second complex input filter 40 ensures that no aliasing effects occur.
- the combination of complex input filter 40 and sampling rate reduction 43 means that any frequency band with a bandwidth of 2,667 kHz can contain the complete useful information.
- every third output value of the second complex input filter 40 is realized in that the transverse part of this filter is operated at 8/3 kHz. This means that the filter output values are only calculated and processed in every third cycle of the 8 kHz sampling cycle.
- the pilot tone generator 50 delivers an identical signal to the pilot signal generator 37 on the transmission side. This signal is required in the clock synchronization block 55 for converting the received and demodulated pilot signal q (n) into the DC signal position and for generating a phase-modulated pilot signal p (n) at the receiving end.
- the averaging 56 serves to average the analytical signal q (n) transformed into the DC signal position, so that the real part is the level of the received pilot tone and the imaginary part is a manipulated variable for the sampling clock tracking (clock correction).
- the averaging is implemented in such a way that the average over the last 128 input signal values q (n) transformed into the DC signal position is formed every 128 sampling cycles.
- the random number generator 54 has the task of generating evenly distributed numbers in the range from 1 to 64, quite analogously to the random number generator 34 on the transmission side. These numbers are used to select random values from a field of 64 complex values.
- two key signals z p (n) and z s (n) are generated from the selected values, one of which (z s (n)) for phase demodulation, ie for uncovering the useful signal y (n) and the second (z p ( n)) is used in the clock synchronization block 55 on the one hand for uncovering the received pilot signal and on the other hand for generating the pilot signal on the receiver side. Due to the clock synchronization, the key signals are of course identical to the key signals z p (n) and z s (n) on the transmission side.
- the implementation of the random number generator 54 is otherwise identical to the implementation in the transmission part, so that reference can be made to the above statements.
- the random numbers supplied to the phase modulator 57 and the phase demodulators 58 and 59 consist of a set of 64 complex values from which discrete values are selected by the random number generator 54.
- the same 64 complex values are used as data records, analogous to the transmission side
- phase modulators 58, 59 already mentioned are required in the receiving station of the SV module.
- One phase demodulator 59 is used to unveil the useful signal y (n) by means of the one key signal z s (n).
- the other phase demodulator 58 is used to recover the Pilots used from the received pilot signal. As already mentioned, these key signals must be identical to the key signals on the transmission side.
- the phase demodulator 57 is used to generate the pilot signal from the pilot tone supplied by the pilot tone generator 50.
- the sampling rate is increased by a factor of 3 - in the example shown from 2,667 kHz to 8 kHz - by inserting two signal values with the value "0" between two signal values according to the following relationship:
- Another (second) complex output filter 62 preferably a (fourth) Hilbert filter, is used to convert the analytical output signal into a real output signal. This serves to limit the bandwidth of the output signal (speech signal) to approx. 2,667 kHz.
- the second complex output filter 62 is again a recursive filter, the structure of which corresponds to that of the first complex output filter 35 on the transmission side and is illustrated in FIG. 9.
- the input signal of the second complex output filter 62 (fourth Hilbert filter) is again an analytical signal.
- the output signal is a real signal.
- the design of the filter in the tested embodiment of the invention is based on the design of an elliptical low-pass filter.
- the low pass is converted into a Hilbert band pass by a transformation in the frequency domain.
- the task of the analog front end 52 on the output side is to convert the digital output signal into an analog output signal (audio signal). This also includes level adjustment.
- the D / A converter part of the analog front end 52 (output) (not shown in detail) consists of a D / A converter, an analog smoothing filter, a programmable amplifier and a differential amplifier.
- the following specifications apply to the output of the analog front end 52:
- Word width 16 bit gain: adjustable in the range of
- the random number generator 54 used begins with each resynchronization at the same starting point.
- the security of obfuscation can be increased if the starting point is changed with every resynchronization. This can be achieved by transmitting the starting point of the random number generator 54 in the preamble.
- Analog Devices ADSP-21msp50 / 55/56 Datasheet, Mixed Signal Processor.
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- Signal Processing (AREA)
- Digital Transmission Methods That Use Modulated Carrier Waves (AREA)
- Radio Relay Systems (AREA)
- Facsimile Transmission Control (AREA)
- Cable Transmission Systems, Equalization Of Radio And Reduction Of Echo (AREA)
- Compression, Expansion, Code Conversion, And Decoders (AREA)
- Mobile Radio Communication Systems (AREA)
- Reduction Or Emphasis Of Bandwidth Of Signals (AREA)
- Interconnected Communication Systems, Intercoms, And Interphones (AREA)
- Transmitters (AREA)
- Fittings On The Vehicle Exterior For Carrying Loads, And Devices For Holding Or Mounting Articles (AREA)
- Alarm Systems (AREA)
- Input Circuits Of Receivers And Coupling Of Receivers And Audio Equipment (AREA)
Abstract
Description
Claims
Applications Claiming Priority (3)
Application Number | Priority Date | Filing Date | Title |
---|---|---|---|
DE4339464 | 1993-11-19 | ||
DE4339464A DE4339464C2 (en) | 1993-11-19 | 1993-11-19 | Method for disguising and unveiling speech during voice transmission and device for carrying out the method |
PCT/EP1994/003693 WO1995015627A1 (en) | 1993-11-19 | 1994-11-09 | Process and device for speech scrambling and unscrambling in speech transmission |
Publications (2)
Publication Number | Publication Date |
---|---|
EP0729678A1 true EP0729678A1 (en) | 1996-09-04 |
EP0729678B1 EP0729678B1 (en) | 1998-08-12 |
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ID=6502948
Family Applications (1)
Application Number | Title | Priority Date | Filing Date |
---|---|---|---|
EP95900687A Expired - Lifetime EP0729678B1 (en) | 1993-11-19 | 1994-11-09 | Process and device for speech scrambling and unscrambling in speech transmission |
Country Status (17)
Country | Link |
---|---|
US (1) | US5778073A (en) |
EP (1) | EP0729678B1 (en) |
JP (1) | JPH09501291A (en) |
KR (1) | KR960706244A (en) |
AT (1) | ATE169787T1 (en) |
AU (1) | AU8141394A (en) |
CZ (1) | CZ143896A3 (en) |
DE (2) | DE4339464C2 (en) |
FI (1) | FI962106A (en) |
HU (1) | HUT74262A (en) |
PL (1) | PL174895B1 (en) |
RU (1) | RU2118059C1 (en) |
SG (1) | SG54159A1 (en) |
SK (1) | SK63096A3 (en) |
TW (1) | TW252241B (en) |
WO (1) | WO1995015627A1 (en) |
ZA (1) | ZA949167B (en) |
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FI101670B (en) * | 1995-12-15 | 1998-07-31 | Nokia Mobile Phones Ltd | A method for announcing the hiding of data transfer between a mobile station network and a mobile station |
US5949878A (en) * | 1996-06-28 | 1999-09-07 | Transcrypt International, Inc. | Method and apparatus for providing voice privacy in electronic communication systems |
JPH10290215A (en) * | 1997-04-15 | 1998-10-27 | Sony Corp | Data transmission reception method and data transmitter-receiver |
DE19746652A1 (en) * | 1997-10-22 | 1999-04-29 | Heinz Brych | Data transmitting and receiving and operating circuit |
US6266412B1 (en) * | 1998-06-15 | 2001-07-24 | Lucent Technologies Inc. | Encrypting speech coder |
US6937977B2 (en) * | 1999-10-05 | 2005-08-30 | Fastmobile, Inc. | Method and apparatus for processing an input speech signal during presentation of an output audio signal |
CA2329889A1 (en) * | 2000-12-29 | 2002-06-29 | Barbir Abdulkader | Encryption during modulation of signals |
US20020173333A1 (en) * | 2001-05-18 | 2002-11-21 | Buchholz Dale R. | Method and apparatus for processing barge-in requests |
DE10215019B4 (en) * | 2002-04-05 | 2007-05-16 | Doepke Schaltgeraete Gmbh & Co | Device for detecting electrical differential currents |
KR100428786B1 (en) * | 2001-08-30 | 2004-04-30 | 삼성전자주식회사 | Integrated circuit capable of protecting input/output data over internal bus |
KR100417125B1 (en) * | 2002-08-07 | 2004-02-05 | 주식회사 팬택앤큐리텔 | Method for Automatically Entering Secured Voice Communication Mode of Wireless Communication Terminal |
KR100483462B1 (en) * | 2002-11-25 | 2005-04-14 | 삼성전자주식회사 | Apparatus for Fast Fourier Transmitting, Method for Fast Fourier Transmitting, and Orthogonal Frequency Division Multiplexing receiving device having the same |
US7460624B2 (en) * | 2004-03-18 | 2008-12-02 | Motorola, Inc. | Method and system of reducing collisions in an asynchronous communication system |
EP1619793B1 (en) * | 2004-07-20 | 2015-06-17 | Harman Becker Automotive Systems GmbH | Audio enhancement system and method |
US7804912B2 (en) * | 2004-09-23 | 2010-09-28 | Motorola, Inc. | Method and apparatus for encryption of over-the-air communications in a wireless communication system |
US8170221B2 (en) | 2005-03-21 | 2012-05-01 | Harman Becker Automotive Systems Gmbh | Audio enhancement system and method |
DE602005015426D1 (en) | 2005-05-04 | 2009-08-27 | Harman Becker Automotive Sys | System and method for intensifying audio signals |
KR100902112B1 (en) * | 2006-11-13 | 2009-06-09 | 한국전자통신연구원 | Insertion method and transmission method of vector information for voice data estimating in key re-synchronization, and voice data estimating method in key re-synchronization using vector information |
KR100906766B1 (en) * | 2007-06-18 | 2009-07-09 | 한국전자통신연구원 | Apparatus and method for transmitting/receiving voice capable of estimating voice data of re-synchronization section |
US20140047497A1 (en) * | 2008-03-12 | 2014-02-13 | Iberium Communications, Inc. | Method and system for symbol-rate-independent adaptive equalizer initialization |
US20090268910A1 (en) * | 2008-04-28 | 2009-10-29 | Samsung Electronics Co., Ltd. | Apparatus and method for initialization of a scrambling sequence for a downlink reference signal in a wireless network |
JP5212208B2 (en) * | 2009-03-23 | 2013-06-19 | 沖電気工業株式会社 | Receiving apparatus, method and program |
RU2546614C1 (en) * | 2013-09-26 | 2015-04-10 | Федеральное государственное бюджетное образовательное учреждение высшего профессионального образования "Пензенский государственный университет" (ФГБОУ ВПО "Пензенский государственный университет") | Method of masking analogue speech signals |
TWI631980B (en) * | 2017-07-24 | 2018-08-11 | 羽昌國際股份有限公司 | Vibration control system for oscillating solid medium |
CN110581743B (en) * | 2018-06-11 | 2021-01-22 | 京东方科技集团股份有限公司 | Electronic device, time synchronization system and time synchronization method |
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US5245660A (en) * | 1991-02-19 | 1993-09-14 | The United States Of America As Represented By The Secretary Of The Navy | System for producing synchronized signals |
US5291555A (en) * | 1992-12-14 | 1994-03-01 | Massachusetts Institute Of Technology | Communication using synchronized chaotic systems |
US5379346A (en) * | 1993-09-30 | 1995-01-03 | The United States Of America As Represented By The Secretary Of The Navy | Cascading synchronized chaotic systems |
-
1993
- 1993-11-19 DE DE4339464A patent/DE4339464C2/en not_active Expired - Fee Related
-
1994
- 1994-10-27 TW TW083109931A patent/TW252241B/zh active
- 1994-11-09 CZ CZ961438A patent/CZ143896A3/en unknown
- 1994-11-09 US US08/648,084 patent/US5778073A/en not_active Expired - Fee Related
- 1994-11-09 AT AT95900687T patent/ATE169787T1/en active
- 1994-11-09 EP EP95900687A patent/EP0729678B1/en not_active Expired - Lifetime
- 1994-11-09 SK SK630-96A patent/SK63096A3/en unknown
- 1994-11-09 DE DE59406692T patent/DE59406692D1/en not_active Expired - Fee Related
- 1994-11-09 KR KR1019960702086A patent/KR960706244A/en not_active Application Discontinuation
- 1994-11-09 HU HU9601333A patent/HUT74262A/en unknown
- 1994-11-09 WO PCT/EP1994/003693 patent/WO1995015627A1/en not_active Application Discontinuation
- 1994-11-09 JP JP7515354A patent/JPH09501291A/en active Pending
- 1994-11-09 AU AU81413/94A patent/AU8141394A/en not_active Abandoned
- 1994-11-09 PL PL94314289A patent/PL174895B1/en unknown
- 1994-11-09 SG SG1996002874A patent/SG54159A1/en unknown
- 1994-11-09 RU RU96105712A patent/RU2118059C1/en active
- 1994-11-18 ZA ZA949167A patent/ZA949167B/en unknown
-
1996
- 1996-05-17 FI FI962106A patent/FI962106A/en unknown
Non-Patent Citations (1)
Title |
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See references of WO9515627A1 * |
Also Published As
Publication number | Publication date |
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KR960706244A (en) | 1996-11-08 |
DE59406692D1 (en) | 1998-09-17 |
ATE169787T1 (en) | 1998-08-15 |
US5778073A (en) | 1998-07-07 |
RU2118059C1 (en) | 1998-08-20 |
PL314289A1 (en) | 1996-09-02 |
WO1995015627A1 (en) | 1995-06-08 |
HUT74262A (en) | 1996-11-28 |
PL174895B1 (en) | 1998-09-30 |
SG54159A1 (en) | 1998-11-16 |
AU8141394A (en) | 1995-06-19 |
DE4339464C2 (en) | 1995-11-16 |
TW252241B (en) | 1995-07-21 |
HU9601333D0 (en) | 1996-07-29 |
FI962106A0 (en) | 1996-05-17 |
FI962106A (en) | 1996-05-17 |
EP0729678B1 (en) | 1998-08-12 |
CZ143896A3 (en) | 1996-11-13 |
JPH09501291A (en) | 1997-02-04 |
DE4339464A1 (en) | 1995-05-24 |
ZA949167B (en) | 1995-07-25 |
SK63096A3 (en) | 1996-11-06 |
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