Classifications

• • H—ELECTRICITY
  • H04—ELECTRIC COMMUNICATION TECHNIQUE
  • H04Q—SELECTING
  • H04Q1/00—Details of selecting apparatus or arrangements
  • H04Q1/18—Electrical details
  • H04Q1/30—Signalling arrangements; Manipulation of signalling currents
  • H04Q1/44—Signalling arrangements; Manipulation of signalling currents using alternate current
  • H04Q1/444—Signalling arrangements; Manipulation of signalling currents using alternate current with voice-band signalling frequencies
  • H04Q1/45—Signalling arrangements; Manipulation of signalling currents using alternate current with voice-band signalling frequencies using multi-frequency signalling
  • H04Q1/457—Signalling arrangements; Manipulation of signalling currents using alternate current with voice-band signalling frequencies using multi-frequency signalling with conversion of multifrequency signals into digital signals
  • H04Q1/4575—Signalling arrangements; Manipulation of signalling currents using alternate current with voice-band signalling frequencies using multi-frequency signalling with conversion of multifrequency signals into digital signals which are transmitted in digital form
• • H—ELECTRICITY
  • H04—ELECTRIC COMMUNICATION TECHNIQUE
  • H04Q—SELECTING
  • H04Q1/00—Details of selecting apparatus or arrangements
  • H04Q1/18—Electrical details
  • H04Q1/30—Signalling arrangements; Manipulation of signalling currents
  • H04Q1/44—Signalling arrangements; Manipulation of signalling currents using alternate current
  • H04Q1/444—Signalling arrangements; Manipulation of signalling currents using alternate current with voice-band signalling frequencies
  • H04Q1/46—Signalling arrangements; Manipulation of signalling currents using alternate current with voice-band signalling frequencies comprising means for distinguishing between a signalling current of predetermined frequency and a complex current containing that frequency, e.g. speech current

Definitions

• the invention relates to a circuit arrangement for recognizing at least one sound signal in a signal mixture, with a number of bandpasses corresponding to the number of possible signal frequencies, the respective center frequency of which corresponds to one of the signal frequencies, each bandpass outputting a bandpass signal corresponding to its filter characteristic
• Sound signals are not only signals with a frequency in the sound range, but also signals outside this frequency range, for example in the high frequency range.
• the sound signals are generated as almost pure sine waves and can fluctuate within a narrow bandwidth.
• circuit arrangements of the type mentioned at the outset have a number of bandpasses which corresponds to the number of possible frequencies of the sound signals to be detected.
• the bandpass filters are tuned precisely to the respective frequency and, based on their filter characteristics, mainly let vibrations of this frequency pass. The presence or absence of a sound signal can thus be determined at the signal level of a bandpass output.
• Such circuit arrangements are used, for example, in telephone technology in the detection of DTMF signals (double-tone multi-frequency signals), in audio tone evaluation, charge pulse evaluation, but also in other areas of technology when control information is generally transmitted with the aid of audio signals .
• Cost of response speed of the circuit arrangement can be achieved.
• This object is achieved according to the invention for a circuit arrangement of the type mentioned at the outset by comparing the level of the respective bandpass signal with the levels of the output signals of two comparison bandpasses also receiving the signal mixture, the center frequencies of which are higher or lower than the center frequency of the respective bandpass , and that depending on the comparison, ID signal is generated which signals the occurrence of a sound signal at the bandpass input when the comparison level is less than the level of the respective bandpass signal.
• the invention is based on the knowledge that a signal mixture composed of useful signals and interference signals always contains several frequencies and is distributed over a frequency band. Sections of this frequency band lying next to one another, which lie in the pass band of the respective bandpass or a comparison bandpass, are thus supplied with approximately the same signal powers averaged over time. As a result, the levels of the output signals of the bandpass and the comparison bandpasses are approximately the same. If the signal mixture contains a sound signal whose frequency coincides with the center frequency of the bandpass, the level of the bandpass signal is higher than the level of the output signals of the comparison bandpass whose center frequencies have a fixed distance from the center frequency of the bandpass.
• a comparison of the levels of the bandpass signal with those of the output signals of the comparison bandpass thus leads directly to the result of whether or not there is an audio signal.
• the bandpasses can have a considerably wider bandwidth than the fluctuation range of the sound signal. This results in a short settling time of the bandpass filters and a high response speed of the circuit arrangement.
• Such bandpasses have a relatively simple structure and, when implemented as a digital filter, require only a small amount of computation.
• bandpasses with fixed center frequencies as well as tunable bandpasses with variable pass ranges whose center frequencies are variable can be used. If only bandpasses with a fixed center frequency are to be used, then three bandpasses are required for each sound signal to be detected, i.e. in addition to the band's own bandpass, two further comparison bandpasses, which each receive the incoming signal mix, are provided.
• the signal mixture can be analyzed at the same time, whereby one sound signal or several sound signals occurring simultaneously in the signal mixture can be recognized very quickly
• bandpass filter with a variable passband is first set to the center frequency of the audio signal and the level of its output signal is determined.
• the center frequencies of the adjacent passbands, which correspond to the passbands of the comparison bandpass filters, are then set and the corresponding levels of the output signals are also determined.
• the levels of the signals determined in this way are compared with one another in the manner already described, and the detection signal for sound signals is generated. These processes have to be repeated for each sound signal.
• the bandpass and the comparison bandpasses work digitally, i.e. the structure customary for digital filters is used, in which memory and arithmetic units are used as components. Instead of continuous signals, discrete sequences of numbers are processed. For this purpose, the incoming analog signals are sampled at short, equidistant time intervals and the sampled values are converted into digital values. By simply changing arithmetic operations or the sampling intervals, any filter characteristics as well as different center frequencies of the bandpass filters can be set. This further development makes it possible to implement bandpasses with different frequency behavior in a single signal processor. The number of these is only limited by the memory volume and the computing speed of the signal processor.
• the arithmetic operations that simulate a bandpass are defined in a arithmetic program. If the arithmetic program is processed several times, the frequency-determining parameters of the arithmetic algorithm being changed, this corresponds to the simulation of a bandpass with a variable passband. If several computer programs with the same structure are provided, each with permanently assigned parameters, bandpasses with unchangeable passbands are simulated.
• second-order digital filters can be used as bandpasses, which have sufficient frequency selectivity.
• the computational effort per sample value and the settling time of such filters are still within the permissible limits for simple requirements such as those that occur, for example, in the audio frequency range.
• B (f) the level of the output signal of the respective comparison bandpass or the bandpass signal with the center frequency f
• Ta the sampling period
• x (n) the nth sample value of the input signal of the bandpass
• n the run variable of the sample value
• N the Is the number of sampled values evaluated.
• This type of evaluation of the spectral distribution within the specified time window corresponds to the filter characteristic of a bandpass filter.
• the spectral component B (f) of the frequency spectrum which indicates the level of the output signal of a bandpass filter, can be determined very quickly by means of conventional signal processors, since arithmetic operations in the arithmetic unit, which only include the summation of products, are the simplest and fastest Routines of a signal processor belong.
• sample values x (n) can be increased with an evaluation function
• Double-tone multi-frequency signals are used in telephony to transmit the dialing information, control of voice memories (voice boxes), among others.
• the control information is transmitted simultaneously in an upper and in a lower frequency band with sound signals as carriers. Since a valid DTMF signal is only available if there is a sound signal per frequency band, the total evaluation time can be considerably reduced by determining the band pass at the maximum level within the respective frequency band. In the subsequent step, only the output signals of the bandpasses determined in this way are used for comparison with the comparison bandpasses corresponding to them.
• FIG. 2 shows a schematic block diagram of a DTMF receiver
• FIG. 3 shows a logical block diagram for evaluating DTMF signals
• FIG. 8 shows a circuit arrangement for determining the level of output signals of a digital bandpass with a fast digital algorithm
• FIG. 9 frequency curves of a digital bandpass filter according to FIG. 8 with and without evaluation of the scanning signals.
• 1 shows parts of a digitally operating telecommunications system in a schematic block diagram.
• This system uses an ISDN switching system 8818, designated by 10, from Nixdorf Computer AG to set up the subscriber connections.
• Two telephone subscribers 12, 14 and, as a further subscriber, a local exchange 16, which can communicate with one another in both directions, are connected to the switching system 10 by way of example.
• the subscribers 12, 14, 16 are connected via subscriber connection circuits 18, 20, 22 to a central PCM bus 24, which transmits pulse code-modulated signals.
• the subscriber line circuits 18, 20, 22 convert the incoming analog signals of the subscribers 12, 14, 16 on the communication network into digital signals and transmit them to the PCM bus 24 using a time division multiplex method.
• the subscribers 12, 14, 16 can also exchange control information which uses DTMF signals (double-tone multi-frequency signals) as information carriers.
• DTMF signals double-tone multi-frequency signals
• These signals are sinusoidal electrical signals whose respective frequencies can fluctuate within narrow limits, for example +/- 1.8% of the center frequency.
• a DTMF receiver 26 in which the invention is implemented, is connected to the PCM bus 24 for evaluating the DTMF signals.
• the DTMF receiver 26 has access to a microcomputer 28, which processes the recognition signals provided by the DTMF receiver 26 and processes them e.g. passes on to a control of the switching system.
• An interface module 32 establishes the connection to the PCM bus 24 and, according to the time-division multiplex method, intervenes at predetermined time intervals. cut to its signals.
• the structure and structure of the data of the communication system described here can correspond to the CCITT-Nor, for example.
• the digital words provided by the Interfa module 32 are so-called A-law coded signals, which have a logarithmic digital coding with a word length of 8 bits. This 8-bit word is converted in a code converter 34 into a digital word with a length of 12 bits with linear coding.
• the digital word which contains voice signals and DTMF signals as information is continuously stored in a memory 36.
• the output digital words are processed by a signal processor 30.
• This is particularly suitable for the fast processing of digital data, as is required for digital filters.
• a TMS 320 from Texas Instruments can be used as signal processor 30.
• the signal processor 30 has access via bus 42 to a fast RAM memory 38 with random access, which serves as a working memory for intermediate results, and to a ROM read-only memory 40, in which constants or coefficients are stored.
• the signal processor 30 controls the memory 36 in order to ensure a synchronous workflow.
• the result of the signal evaluation by the signal processor 30 is output via the bus 42 to a downstream controller (not shown).
• the memory 36 has two switchable memory areas SP1 and SP2 controlled by the signal processor 30.
• the memory area SP1 is loaded with incoming digital words; at the same time, the memory area SP2 is read out
• the switches 50, 52 are switched over, and the incoming data words are now read into the memory area SP2 and data is read out from the memory area SP1.
• the memory 36 thus continuously receives data at a constant data rate, and it can be read out for evaluating the signals at a higher data rate.
• the data words output by the memory 36 are evaluated in the downstream signal processor 30 using digital filter algorithms. These filter algorithms emulate the bandpasses, which are designated B1 to B8 in FIG. 3.
• the center frequencies of the band passes B1 to B8 are matched to the frequencies of the DTMF signals. According to the CCITT standard, the center frequencies lie in two frequency ranges, bandpasses B1 to B4 being assigned to a lower frequency range and bandpasses B5 to B8 being assigned to an upper frequency range.
• the levels of the bandpass signals of the bandpasses B1 to B4 of the lower frequency range are determined and the bandpass is determined at the maximum level.
• the bandpass B2 has the largest signal level.
• the filter calculations for the comparison bandpasses V2u, V2o belonging to the bandpass B2 are carried out with a lower center frequency or a higher center frequency than the center frequency of the bandpass B2. If the levels of the output signals of the comparison bandpasses V2u, V2o are lower than the signal level of the bandpass B2, the presence of a tone signal in the lower frequency range is signaled.
• the bandpass B5 is painter signal level of the upper frequency range determined and compared with corresponding signal levels of the comparison bandpass filters V5o, V5u.
• a logic device 54 preferably in the form of digital
• Algorithms that are processed in the signal processor 30 are implemented and perform the level comparison. If there is a sound signal in the upper and in the lower frequency range at the same time, a detection signal 56 for DTMF signals is generated. In accordance with the CCITT standards, the time conditions for valid DTMF signals are also checked in logic module 54, the DTMF signals are decoded and the detection signal 56 is passed on to the downstream controller.
• the individual evaluation steps for recognizing a DTMF signal are shown in a more general form in a logical flowchart.
• the signal levels S1 to S8 of the bandpass filters B1 to B8 are calculated both in the lower and in the upper frequency range.
• the bandpass Bn with maximum signal level Sn in the lower frequency range is determined in step 62, the bandpass Bm with maximum signal level Sm in the upper frequency range in step 64.
• the levels of the output signals Ano, Anu of the comparison bandpass filters Vno belonging to the bandpass Bn are calculated with a higher or Vnu with a lower center frequency.
• step 68 A similar evaluation takes place in step 68, but with reference to the comparison bandpasses Vmo, Vmu of the upper frequency range belonging to the bandpass Bm.
• the levels of the different signals are compared with one another in the evaluation steps 70, 72. If the signal level Sn in the lower frequency range is greater than the levels Ano, Anu of the comparison bandpass filters Vno, Vnu, a branch is made to step 72. Here it is checked whether the signal level Sm is greater than the level of the output signals Arno, Amu. If this is the case, a DTMF signal is present, which is signaled in evaluation step 74. If the comparison conditions in steps 70, 72 are not met, a branch is made to step 76, which indicates the absence of a DTMF signal. Further evaluation steps, which concern, for example, the maximum permissible deviation of the signal levels in the upper and lower frequency band and the monitoring of time criteria when evaluating the signal levels, are not dealt with here, since these are part of the usual evaluation of DTMF
• FIG. 5 shows the filter characteristics of the bandpasses B1 to B4 of the lower frequency band with the center frequencies f1, f2, f3, f4.
• the attenuation curve a of the bandpass filters is plotted in dB over the frequency f. Filters of the second order are used, which achieve a high frequency selection with reasonable effort and ensure high interference signal suppression.
• the band passes B5 to B8 of the upper frequency band likewise have the attenuation curves shown in FIG. 5 at the center frequencies f5, f6, f7, f8.
• FIG. 6 shows the attenuation curve a over frequency f for a bandpass filter Bn with center frequency fn with the associated comparison bandpass filters Vno and Vnu.
• the latter have an upper center frequency fno or a lower center frequency fnu.
• the level comparison at the intersections 77, 79 achieves a high degree of selectivity between the passband and the stopband of the entire circuit arrangement.
• the pass band is defined by the intersections 77, 79 of the bandpasses Vnu, Bn, Vno. This passband can be set by the filter characteristics of the bandpass filters and their center frequency and preferably corresponds to the maximum fluctuation range b of the on-signal.
• FIG. 7 shows the circuit diagram of a circuit arrangement for determining the level of the output signal of a recursive digital bandpass filter of the second order.
• the sampled values x (n) form the input signals of the digital filter, to which a digital value 82 is added at the summation point 80 in order to obtain the output signal y (n) of the digital bandpass.
• the value 82 is obtained by adding two values 84, 86 at the summing point 88, which result from the time-delayed output signal y (n) to which factors are applied.
• the simply delayed output signal y (n-1) is multiplied by a filter coefficient -b1; the twice delayed output signal y (n-2) with the filter coefficient -b2.
• the Filter ⁇ coefficients b1, b2 are the familiar tables for digitalized ⁇ e filter removed.
• the absolute value jy (n) j of the output signal y (n) is to be determined in an absolute value image 94.
• y (n) ⁇ is added in the summing element 96 to a total amount value Y (n) delayed by the delay element 98 by a time pulse.
• the total value Y (N), determined digitally after N samples, corresponds to the rectifying value of an alternating variable and is a measure of the level of the output signal of a bandpass.
• other quantities indicating the level can also be formed, for example the effective value, the root mean square, the peak value, etc. However, the arithmetic operations are then more time-consuming.
• the sample values x (n) which can be taken from the memory 36 are divided into two arithmetic branches A, B.
• the sampled values x (n) are multiplied by a coefficient al (n) which is dependent on the run variable n and added at the summation point 100 to a value Re which has been delayed by a clock pulse via a delay element 102.
• a corresponding multiplication by the coefficient a2 (n) and likewise an addition at the summation point 104 with a value Im delayed by the delay element 106 take place in the second arithmetic branch B.
• the level Y (N) can be determined directly by pulling the root of the equation given above; However, for the sake of simplicity, the quadratic level Y 2 (N) can also be processed without disadvantages.
• the coefficients a1 (n), a2 (n) and c (n) or products of these coefficients are stored in the read-only memory 40 (see FIG. 2) and are called up each time the computing operations are carried out.
• FIG. 10 shows the frequency response of the digital bandpass shown in FIG. 9.
• the curve 110 is shown without the evaluation function c (n) and the curve 112 with the evaluation function.
• the attenuation a in dB shows a minimum at the frequency f1 for both curves.
• a higher damping is achieved with an increasing frequency distance from the center frequency f1.
• the flat course of curve 110 near the center frequency has no effect on the frequency selectivity of the circuit arrangement, since, as was explained in the description of FIG. 6, this only depends on the intersection of the attenuation course of the bandpasses involved in the sound signal evaluation

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• Engineering & Computer Science (AREA)
• Computer Networks & Wireless Communication (AREA)
• Stereo-Broadcasting Methods (AREA)
• Measurement Of Mechanical Vibrations Or Ultrasonic Waves (AREA)
• Digital Transmission Methods That Use Modulated Carrier Waves (AREA)
• Telephone Function (AREA)

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• 1988
  • 1988-09-12 DE DE19883831047 patent/DE3831047A1/de active Granted
• 1989
  • 1989-09-06 WO PCT/EP1989/001039 patent/WO1990003087A2/de unknown

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