RU2291405C2 - Method and digital device for identification of tone signals in communication lines - Google Patents

Abstract

FIELD: communication engineering.

SUBSTANCE: method and device can be used for identification of tone signals for automatic umber determination of phone set of calling user in commutated channels of data transmission networks. To identify tone signals in communication lines, tone frequency analog signals, measured in phone network, are amplified preliminary. Then signal are limited in amplitude and are subject to digitization to digital form by means of analog-to-digital conversion and subsequent digital processing, at which processing the digital flow of discrete signals in tone frequency range in parallel code is subject to filtering in real time scale. Continuously filtered multi-frequency tone discrete signals are modulo detected and stored within specified time range, while forming multi-frequency spectral components at any clock step of general synchronization. Continuous inspection is performed to monitor if amplitude of useful signal exceeds level of errors in telephone line together with noise component of analog-to-digital conversion. Moments when amplitude exceeds level of errors are registered at sliding mode, intermediate storage is performed, multi-frequency code combinations are identified and identified signals are registered. Device for realization of method of identification of tone signals in communication lines as analog-to-digital converter connected to phone line through amplifier, and signal digital processing system. Amplifier is connected to phone line through commutation and match unit. Input of analog-to-digital connecter is connected with information input of buffer register, which is used as multi-positional switch. Output of switch is connected with common bus of data of adaptive narrow-band digital infinite-impulse response filters (IRF), connected in parallel. Outputs of filters are connected with control unit through multiplexer, modulo digital detectors un it and unit of arithmetic adders having storage function at sliding mode of operation of levels multi-frequency spectral components of useful signal and fixing of level of exceed over error. Outputs of control unit are connected with inputs of current analysis, identification and tone signal registration code logic unit. Indication unit is connected with output of logic unit through microprocessor set. Preferable standard of manufacture of adaptive narrow-band digital infinite-impulse response filters is described.

EFFECT: improved speed of operation of identification of tone signals in real time scale.

5 cl, 2 dwg

Description

The invention relates to communication technology and technology, for example, identification of tones for automatically determining the number (AON) of a caller’s telephone in switched channels of information transmission networks.

A known method and device based on a modern digital signal processor (DSP), which are designed to send a request signal code AON (500 Hz) via a telephone line to the telephone station (GTS) and remote reception in the tonal range of frequencies of information in a two-frequency tonal alarm system DTMF (Dual-Tone Multifrequency), code "2 of 6", about the number and category of the caller’s phone using the "interval-free package" method [1].

The disadvantage of the known AONs, which appeared in the consumer market since 1990 as terminal subscriber equipment, is the relatively low indicators for determining the number of the calling subscriber with deteriorating parameters of the subscriber telephone lines (uneven attenuation and distortion of the frequency response, industrial noise). A multi-frequency broadband signal passing through switched analog communication channels is distorted, since different frequency components are delayed for different times, depending on the parameters of the switched communication channel and the frequency of the signal under study.

The problem of detecting a harmonic signal (or its segments) against a background of noise has been studied quite well [1, 2, 3]. Technical implementation is complicated by a high noise level with an undefined distribution law and non-stationary parameters of information transmission channels in communication technology and technology.

A close analogue of the proposed is a Japanese patent "Method and apparatus for the analysis of music sounds" [G 01 H 3/00, JP 2750332 B2, 2126123A]. A digital device that implements a method for analyzing music sounds based on the direct and inverse fast Fourier transform (FFT) contains a preliminary amplifier of the analyzed sounds coming from a source, such as a microphone. Through an analog-to-digital converter (ADC), the amplified signal is supplied to the FFT unit. The received signal enters the memory unit and then passes through a buffer in which the spectra of the allocated sound (tonal) components are stored. The signal passes from the buffer to the inverse FFT block (IFFT). From the OBPF block, the signal enters the computing blocks to obtain the amplitude and, accordingly, frequency characteristics according to the given formulas.

A common disadvantage of the known methods and devices for identifying tonal signals in AN tasks is the difficulty of real-time implementation of the FFT and IFFT algorithms. These difficulties are especially characteristic during the technical implementation of digital narrow-band FIR filters (for example, using the FFT algorithm), which increase very rapidly with increasing bitmap ADC and quantization frequency. Therefore, to identify tonal signals in real time, it is necessary to apply adaptive digital filters with infinite noise immunity with an infinite impulse response (IIR filters).

The present invention is directed to improving the speed of identification of tonal signals in real time. The technical result achieved in solving this problem is to completely compensate for the random phase error of the tonal frequency signals, as well as signals with other types of modulation, for example, PCM.

To do this, in the known method, which consists in pre-amplifying analog signals of tonal frequencies taken from the telephone network, limiting the amplitude, digitizing these signals into digital form using analog-to-digital conversion and subsequent digital processing, with this digital processing, the digital stream of discrete signals in the tonal frequency range in the parallel code is filtered in real time, continuously filtered multifrequency discrete tones are detected modulo, accumulate during a fixed time interval, forming multi-frequency spectral components at each clock step of the overall synchronization, they continuously monitor the excess in amplitude of the useful signal over the noise level of the telephone network along with the noise component of the analog-to-digital conversion, fix the moments of these excesses in the sliding mode, perform an intermediate accumulation, identify multi-frequency code combinations and register the identified signals.

In this case, real-time digital filtering is preferably carried out by the conveyor processing method in digital IIR filters, the natural frequency, quality factor and transmission coefficient of each filter are adaptively retuned, and the filtered signal is orthogonal to the signal in the previous step.

A digital device for implementing the proposed method for identifying tonal signals in communication networks contains, in addition to the known analog-to-digital converter connected via an amplifier to a telephone line, a digital signal processing circuit in which the amplifier is connected to a telephone line through a switching and matching unit, the analog-to-digital converter is connected by its output to the information input of the buffer register used as a multi-position key, the output of which is connected to a common data bus for parallel-connected adaptive narrow-band digital IIR filters, the outputs of which through a multiplexer, a digital block of detectors modulo and arithmetic adders with accumulation are connected to the control unit in the sliding mode of the levels of the multi-frequency spectral components of the useful signal and recording events when these levels are exceeded by interference, outputs which are connected respectively to the inputs of the logical unit of the current analysis, identification and registration of tone codes, and the display unit p It is connected to the output of the logic unit through a microprocessor kit.

Moreover, each adaptive narrow-band digital IIR filter is preferably made in the form of a series-connected first arithmetic multiplication unit, an arithmetic adder-subtractor and two digital integrators, the first of which consists of a second block of arithmetic multiplication and a first arithmetic adder with accumulation connected in series with each other and the second digital integrator consists of the third arithmetic unit of multiplication and the second arithmetic sum ora with accumulation, and the output of the second digital integrator is connected to the second input of the arithmetic adder-subtractor and is simultaneously the first output of the IIR filter, the output of the first digital integrator is simultaneously the second output of the IIR filter, and the second input of the first arithmetic multiplication unit and the combined second inputs of the second and the third arithmetic multiplication blocks are connected to the corresponding control outputs of the microprocessor kit.

Figure 1 shows the functional diagram of the proposed device;

figure 2 is a structural diagram of an adaptive narrowband digital IIR filter of the second order.

The identification of tones in communication networks, for example, a public switched telephone network, is carried out as follows (Fig. 1). A call signal of the caller ID code (500 Hz) is sent via a connecting telephone line to the city telephone exchange (GTS). Information in the DTMF dual-frequency tonal alarm system, code “2 of 6”, about the number and category of the caller’s phone using the “interval-free packet” method is received in analogue form through the switching and coordination unit with telephone line 1. These signals are amplified by amplifier 2 and are limited in amplitude to prevent the failure of the analog-to-digital converter (ADC) 3, which is connected to the output of amplifier 2 with its information input. The output of the ADC 3 in parallel code is through buffer register 4, used as nogopozitsionnogo key connected to a common data bus to which are connected in parallel to the data inputs of the code in the general case N narrowband adaptive IIR digital filters 5-10. These filters simultaneously parallel real-time digital filtering of the multi-frequency signal captured as a digital stream from the output of the ADC 4. Moreover, the discrete signals from the first and second outputs of each digital IIR filter 5-10 are mutually orthogonal on a fixed observation interval, which corresponds to the imaginary and the material component of the current spectrum of tonal signals after the FFT in the prototype. The outputs of digital IIR filters 5-10, performing continuous digital filtering through a multiplexer 11, are connected to the corresponding inputs of a block 12 of series-connected detectors modulo filtered digital multi-frequency tonal signals and arithmetic accumulators, the outputs of which, in turn, are connected to the corresponding inputs unit 13 for continuous monitoring of the excess in amplitude of the current spectral components of the multi-frequency useful signal at each step of the overall synchronization of all previous blocks pipelined digital stream identifiable signals. Unit 13 controls the excess of the signal over the noise level of the telephone network together with the noise component of the ADC records the moments of these excesses in the sliding mode. The corresponding outputs of this block 13 are connected to the inputs of the logical current analysis block 14, the accumulation of events of the previous block, the identification and subsequent registration of the corresponding tone codes, for example, "2 of 6". This information is output through the microprocessor kit 15 to the display unit 16. The overall synchronization of all digital signal processing and control units is carried out by the microprocessor kit 15, for example, from an internal precision thermally stabilized high-frequency generator through a frequency divider. The first synchronization and control output of the microprocessor set 15 is connected to the synchronization input of the ADC 3 and the buffer register 4. The second synchronization and control output of the microprocessor set is connected (not shown conventionally in Fig. 1) with the corresponding synchronization inputs of adaptive narrow-band digital IIR filters 5-10. The third synchronization and control output of microprocessor kit 15 is connected to the corresponding inputs of multiplexer 11, module 12 of digital detectors modulo and arithmetic accumulators, block 13 of current level control and logic block 14 of current analysis of events of code combinations. The fourth synchronization and control output of the microprocessor set 15 is connected to the control input of the display unit 16.

A more detailed diagram of a preferred embodiment of an adaptive narrow-band digital IIR filter is shown in FIG. Each adaptive narrow-band digital IIR filter 5-10 is preferably made in the form of a series-connected first arithmetic multiplier 17, an arithmetic adder-subtractor 18 and two digital integrators 19, 20, the first of which consists of a second arithmetic multiplier 21 and a first arithmetic adder with by accumulation 22 connected in series with each other, and the second digital integrator 20 consists of series-connected third arithmetic multiplication unit 23 and the second arithmetic an adder with accumulation 24, the output of the second digital integrator 20 is connected to the second input of the arithmetic adder-subtractor 18 and at the same time is the first output of the IIR filter, the output of the first digital integrator 19 is simultaneously the second output of the IIR filter, and the second input of the first arithmetic multiplication unit 17 and the combined second inputs of the second and third arithmetic multiplication units 21, 23 are connected to the corresponding control outputs of the microprocessor set 15.

The information input of each adaptive narrow-band digital IIR filter of the second order 5-10 from the common data bus from the output of the buffer register 4, used as a multi-position key and controlled from the corresponding output of the microprocessor set 15, continuously receives a stream of discrete signals:

Z (kT) = X (kT) + N (kT),

where X (kT) is a useful multi-frequency discrete tone signal;

N (kT) is the generalized interference;

T is the ADC time quantization step;

k = 0, 1, 2, 3, ... M.

An additional disadvantage of the method and device for identifying tonal signals based on the FFT and IFFT methods is the need to calculate and store in the buffer memory DSPs in digital form the discrete ordinates sin (ωt) and cos (ωt). The angular frequency ω, measured in radians / s, is related to the frequency of the individual components of the multi-frequency tonal signaling, as ω = 2πf, where f is the frequency in Hz.

Multi-frequency tonal signaling, for example "2 of 6", characteristic of Russia and a number of other countries, strictly regulates individual tonal frequencies, for example 500 Hz, 700 Hz, 900 Hz, 1100 Hz, 1300 Hz, 1500 Hz, 1700 Hz, ...

The presented scheme of a digital IIR filter corresponds to a second-order filter with a maximum Q factor for a correctly selected step of quantization of the ADC 3 in time and sufficient bit depth in amplitude. The optimal time quantization step with a one percent permissible error:

as indicated in the book: V.S. Gutnikov. Filtering measuring signals. L .: Energoatomizdat, 1990, p. 50. In this case, at least 22 samples are required per sinusoidal period.

This disadvantage is deprived of the adaptive narrow-band digital IIR filter circuit of FIG. 2. The discrete signal Z (kT) is subjected to conveyor processing of the first arithmetic multiplication unit 17, the arithmetic adder-subtractor 18 and two digital integrators 19, 20. The first digital integrator 19 consists of the second arithmetic multiplier 21 and the first arithmetic adder with accumulation 22. Second the digital integrator 20 comprises respectively a third arithmetic multiplication unit 23 and a second arithmetic adder with accumulation 24.

The first output of the digital IIR filter Y1 (kT) is connected to the second input of the arithmetic adder-subtractor 18. The second input of the first block of the arithmetic multiplication 17, corresponding to the transmission coefficient of the digital IIR filter, is digitally reconfigured by one of the control digital outputs of the microprocessor set 15. Second the combined second inputs respectively of the arithmetic unit 21 and multiplication by the multiplication of the third arithmetic unit 23 determines the natural frequency f _r a separate component of the multicarrier signal ation. This parameter is digitally configured by the second separate control output of the microprocessor set 15. Moreover, the accuracy of these two independently configurable parameters is determined only by the bit grid of the control outputs of the microprocessor set 15 and arithmetic multiplication units 17, 21, 23.

The recurrence equations corresponding to this embodiment of the technical implementation of the adaptive narrow-band digital IIR filter are as follows:

Y3 (kT) = Z (kT) K _F -Y1 (kT),

where Y3 (kT) is the digital signal at the output of the arithmetic adder-subtractor 18;

Y1 (kT) - digitally filtered signal from the first output of the IIR filter;

To _F is the transmission coefficient of the digital IIR filter, which can be retuned to compensate for the attenuation in the communication line of a particular component of the multi-frequency signaling f _t .

Y2 (kT) = Y2 [(k-1) T] + Y3 (kT) (T / T _F ),

where Y2 (kT) is the digital signal from the output of the first digital integrator 19, which serves as the second output of the IIR filter.

Y1 (kT) = Y1 [(k-1) T] + Y2 (kT) (T / T _F ),

where the coefficient (T / T _F ) corresponds to the natural frequency of the digital IIR filter and can quickly be reconfigured with high accuracy in digital form by microprocessor kit 15. So, for example, if f _T = 1700 Hz, then the half-period T _F / 2 = 1/34 10 ^-2 s. To fulfill the requirement of one percent error, it is necessary to reduce the sampling step to T = 10 ^-5 s (10 μs). General synchronization of all blocks of the digital device shown in Fig. 1 is carried out from a microprocessor kit 15, which generates synchronization signals, for example, through a frequency divider (1:10) from a precision thermally stabilized crystal oscillator with f _Г = 10 MHz.

Discrete signals from the first and second outputs of the digital IIR filter are mutually orthogonal, which corresponds to the prototype of the material and imaginary components of the FFT and IFFT. In the digital quadrature demodulator circuit, these signals correspond to the in-phase and orthogonal components. The advantage of this technical implementation option is not only in speed, which is inversely proportional to the sum of elementary arithmetic operations at each step of the optimal moving spectral analysis, but also in accuracy, since it is not necessary to calculate and store in the DSP buffer memory individual discrete sin (ω _T t) and cos (ω _T t).

Literature

1. S.L. Koryakin-Chernyak. Caller ID in telephones. Telephone sets from A to Z, book 2. St. Petersburg, Science and Technology, 2003 (330 p.)

2. M.S. Kupriyanov, B. D. Matyushkin, D. B. Golovkin. Implementation of a multifunctional harmonic signal detector on the DSP, "Digital Signal Processing" No. 1, 2001, p. 28-31.

3. Subscriber device of the UON-51 phone number identifier. Technical description and instruction manual. Moscow, State Institution NPO "STiS" of the Ministry of Internal Affairs of Russia, 2001 (20 p.)

4. G 01 H 3/00, Method and apparatus for analyzing music sounds, Arai Kiyotsugu, JP 2750332 B2 2126123 A, (10) 0213. 08/08/99.

Claims (4)

1. A method for identifying tonal signals in communication networks, which consists in pre-amplifying analog signals of tonal frequencies taken from a public telephone network, limiting the amplitude, digitizing these signals into digital form using analog-to-digital conversion and subsequent digital processing, characterized in that the digital stream of discrete signals in the tonal frequency range in the parallel code is filtered in real time, the continuously filtered discrete signals are detected modulo accumulate them over a fixed time interval, forming multi-frequency spectral components at each clock step of the overall synchronization, perform continuous monitoring of the excess in the amplitude of the useful signal over the noise level of the telephone network together with the noise component of the analog-to-digital conversion, record the moments of these excesses in the sliding mode, perform intermediate accumulation, identify multi-frequency code combinations and register the identified signals. 2. The method according to claim 1, characterized in that the real-time digital filtering is carried out by the conveyor processing method in digital IIR filters, the natural frequency, quality factor and transmission coefficient of each filter are adaptively reconfigured, and the filtered signal is orthogonal to the signal in the previous step. 3. A device for identifying tonal signals in communication networks, containing an analog-to-digital converter connected through an amplifier to a telephone line, and a circuit for digital processing of these signals, characterized in that the amplifier is connected to a telephone line through a switching unit and matching with a telephone line, analog -digital converter is connected by its output to the information input of the buffer register, the output of which is connected to a common data bus of parallel-connected adaptive narrow-band digital IIR filters the outputs of which through the multiplexer, the digital block of modulo detectors and arithmetic adders with accumulation are connected to the block for monitoring the level of signals and recording events of excess of these levels over the noise, the outputs of which are connected respectively to the inputs of the logical unit for the current analysis, identification and registration of tone codes, and the display unit is connected to the output of the logical unit through a microprocessor kit. 4. The device according to claim 3, characterized in that each adaptive narrow-band digital IIR filter is made in the form of series-connected first arithmetic multiplication unit, arithmetic adder-subtractor and two digital integrators, the first of which consists of the second block of arithmetic multiplication and the first arithmetic accumulator adders connected in series with each other, and the second digital integrator consists of a third arithmetic multiplication unit and a second arithmetic connected in series accumulator, the output of the second digital integrator connected to the second input of the arithmetic adder-subtractor and at the same time is the first output of the IIR filter, the output of the first digital integrator is simultaneously the second output of the IIR filter, and the second input of the first arithmetic multiplication unit and the combined second inputs the second and third arithmetic multiplication units are connected to the corresponding control outputs of the microprocessor set.

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