WO1982004488A1 - Techniques d'adaptation pour la mesure et la determination automatiques de la frequence - Google Patents

Techniques d'adaptation pour la mesure et la determination automatiques de la frequence Download PDF

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Publication number
WO1982004488A1
WO1982004488A1 PCT/US1982/000664 US8200664W WO8204488A1 WO 1982004488 A1 WO1982004488 A1 WO 1982004488A1 US 8200664 W US8200664 W US 8200664W WO 8204488 A1 WO8204488 A1 WO 8204488A1
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Prior art keywords
signal
frequency
variable
filter
value
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PCT/US1982/000664
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English (en)
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Electric Co Western
Carl Jerome May Jr
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Electric Co Western
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Publication of WO1982004488A1 publication Critical patent/WO1982004488A1/fr

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    • 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
    • H03—ELECTRONIC CIRCUITRY
    • H03H—IMPEDANCE NETWORKS, e.g. RESONANT CIRCUITS; RESONATORS
    • H03H21/00—Adaptive networks
    • H03H21/0012—Digital adaptive filters
    • H—ELECTRICITY
    • H03—ELECTRONIC CIRCUITRY
    • H03H—IMPEDANCE NETWORKS, e.g. RESONANT CIRCUITS; RESONATORS
    • H03H21/00—Adaptive networks
    • H03H21/0012—Digital adaptive filters
    • H03H21/0025—Particular filtering methods

Definitions

  • This invention relates to the determination of selected characteristics of signals and, more particularly, to an adaptive technique for determining a characteristic, such as frequency, of a digitally encoded version of a signal.
  • a characteristic such as frequency
  • some type of frequency control whether as a variable or a parameter is probably the most often used.
  • MF multifrequency
  • TOUCH-TONE dialing numerous test functions
  • data transmission numerous test functions
  • Both analog filters and digital filters of numerous types have been devised for these purposes.
  • the cost and complexity of available arrangements have left something to be desired given the number and variety of applications in the telecommunications industry alone.
  • a further consideration is the presence of quantities of noise in various forms that accompany the signals in a transmission environment.
  • digital filtering techniques have evolved, by and large, as direct replacements of analog arrangements or by using a transformation process, such as the discrete Fourier transform (DFT) or the fast Fourier transform (FFT).
  • DFT discrete Fourier transform
  • FFT fast Fourier transform
  • the direct replacement process is an utter disregard of the inherent advantages of digital filtering processes.
  • the transformation process although theoretically appealing, is expensive since it generally requires complex and dedicated hardware to provide the speed necessary for real time applications.
  • Recent efforts utilized to provide the bandwidth saving of transmission media, such as digital speech interpolation systems, for example place further demands on the rapid and accurate classification of digital signals into speech, data, or tones each with as many other details as are appropriate for the purpose of the classification.
  • the invention is an adaptive technique that capitalizes on signal correlation in a manner convenient for providing usable frequency information.
  • a feature of the implementations of this technique is high computational efficiency to provide speed and economy for applications particularly those using integrated circuits.
  • An adaptive technique includes an arrangement and method capable of determining the frequency of a signal component present in digitally encoded samples of an input signal. The technique uses an adaptively weighted combination of derived values from at least two samples of the input signal to predict a new signal value.
  • the difference between the predicted value and the actual value defines an error.
  • a variable related to the frequency present is developed and retained using the error and the sign of one of the derived values. This variable serves as the coefficient used to weight the derived values in forming the prediction of the new value.
  • the arrangement serves as a notch filter and the variable may be used to indicate the frequency.
  • an illustrative embodiment falls into the class of a finite impulse response filter. Another embodiment falls into the class of an infinite impulse response filter. Still another illustrative embodiment has variable parameters, M and Q, whose selected values determine the filter class without rearranging the embodiment.
  • Still further aspects of the inventive principles are utilized to provide multiple frequency adaptive filtering.
  • Cascade connecting of two adaptive filters provides an arrangement for concurrent adaption of each filter.
  • Each filter adapts to one of the two frequencies which may be present in an input signal.
  • Three adaptive filters are cascaded together to provide the same capability for three different frequency components in an input signal. In these two arrangements, phase shifting is utilized so each adaptive filter is able to function on its particular signal component.
  • FIG. 1 is a map relating the error of a prediction to the magnitude of the input signal.
  • FIG. 2 is a basic illustration of an embodiment of the invention.
  • FIG. 3 is a feedforward or FIR illustrative embodiment of the inventive principles.
  • FIG. 4 is a recursive or IIR illustrative embodiment of the invention.
  • FIG. 5 is a particularly flexible illustrative embodiment of the invention.
  • FIG. 6 is a flow chart of an overall procedure using the described adaptive technique.
  • FIG. 7 is a flow chart of the basic adaptive filter technique.
  • FIG. 8 is a family of filter characteristics obtainable using the described adaptive filter techniques.
  • FIG. 9 illustrates an application of the inventive technique to dual frequency filtering.
  • FIG. 10 illustrates an application for adaptively filtering three different frequencies.
  • A, T and ⁇ are arbitrary, but fixed, and ⁇ (n) is the uncorrelated (noise) component of y(n) (including encoding noise). If it is assumed that ⁇ (n) is zero for the moment, a representation of a sampled sine wave is given by the unique minimum (second) order difference equation for y(n) which is
  • B may be determined with a single computation.
  • an error signal, ⁇ (n) may be defined as the difference between the actual incoming sample as defined by Equation (3) and the estimate given by Equation (5) that is
  • Equation (6) may be arranged to define the ERROR( ) and again, under ideal conditions, a single correction to can reduce ERROR( ) to zero. In an actual application with noisy input signals and finite precision the same computation gives an estimate of ERROR( ). This noisy estimate is still adequate to converge if the corrections are applied judiciously. In general, the correction to becomes
  • Equation (11) is the digital Exponentially Mapped Past, EMP, of (n).
  • EMP averaging is particularly useful in control or detection situations where interest is directed at the recent past of a process and is described in detail, for example in the IRE Transactions on Automatic Control, Vol. AC-5 (January 1960), pages 11-17. From Equation (11), (n) will approach B exponentially (without bias) with the time constant of
  • a correction algorithm can be as simple as: C B equals a single fixed correction magnitude with an appropriate sign. The magnitude of such a correction is bounded by the required accuracy in . A high accuracy requirement and large initial error would lead to long convergence times. If speed is required we need to make large corrections, at least initially.
  • the first way involves the recognition that large errors in can produce high magnitudes of ⁇ (n). Large errors are detected by comparing
  • a second way of avoiding the deficiency of a single fixed magnitude correction involves enlisting the aid of an external control. For example, when
  • Equation (13) All the convergence algorithms determine the direction for a correction in the same manner, i.e., Equation (13). Note that this sign is dependent on the sign of the error signal and the sign of a delayed version of the input signal. For sources having a good signal to noise ratio, the sign of the input samples is rarely affected by noise, when ERROR( ) is large, the error signal is also large and its sign is dominated by the correlated component. In this case, relatively large corrections can be made confidently to . Unfortunately as approaches the correct value, the correlated component of the error signal will become smaller than the noise component. Then the sign of the error signal becomes very unreliable for determining the sign of the correction to . In this case corrections must be averaged over many samples.
  • FIG. 2 illustrates a particular form of the adaptive filter comprising a notch filter with an appropriate adaptation algorithm.
  • Overall notch filter 10 has a transfer function, H N , and includes estimation filter 11 with a transfer function, H E , and means to produce an output error signal, ⁇ , the difference between the estimate and the incoming sample.
  • the adaptation algorithm is used to form the frequency estimate in box 13.
  • the correction to the estimate may be considered as two items, direction and magnitude.
  • the direction or sign of the correction may be obtained by exclusively ORing the sign of x(n-1) and the sign of the error signal.
  • the magnitude may be selected from a short list of preselected fixed values with the larger values used initially in adaptation and smaller values used later to obtain an accurate convergence.
  • the estimation filter also can provide x(n-1), a signal in phase with the partial of with respect to as required by the convergence algorithm.
  • the estimation filter requires a single input variable, , to define the frequency component being estimated.
  • the adaptation algorithm is driven by the notch filter output signal, ⁇ , and x (n-1).
  • FIG. 3 is a basic feedforward implementation of the principles of the invention.
  • the input signal y(n) is applied to signal adder 21 and a 2-tap delay line formed by delays 22 and 23.
  • the output of delay 22 provides the immediate previous code group y(n-1) while delay 23 provides the code group y(n-2) immediately previous to that of delay 22.
  • multiplier 24 which weights the output of delay 22 in accordance with a variable parameter to produce an input for adder 26.
  • the components of FIG. 3 described in the foregoing paragraph form an adaptive filter which, when appropriately controlled, serves to attenuate a frequency component of significant signal energy in the input signal.
  • adder 26 provides an estimate of the current code group based upon a weighted combination of the two previous code groups.
  • Adder 21 combines the predicted code group with the actual code group to provide an error signal ⁇ (n) which is the difference or the error in the prediction.
  • the error signal and the output of delay 22 are the inputs to correction generator 27 which serves in providing a variable parameter.
  • This parameter has a prescribed relationship to the frequency of the signal component.
  • the arrangement of FIG. 3 automatically functions to decrease the error signal by providing maximum attenuation.
  • the output of generator 27 is applied to accumulator 28 fortned by interconnected store 31 and adder 32.
  • the output of store 31 provides the variable parameter for multiplier 24.
  • the output of adder 21 provides an indication of the accuracy of the prediction of the current sample while the output of delay 22 provides an "inphase" signal indicative of the direction, i. e., its sign, of the correction of the variable parameter.
  • Generator 27, whose output is designated C B (n) may be determined, for example, by the fractional relationship of as long as the denominator is not zero. Through successive corrections this arrangement will reduce ⁇ (n) to essentially zero until the input signal changes in frequency level or phase.
  • frequency indicator 36 provides an output signal accurately representative of frequency.
  • Indicator 36 may take the form of a look-up table., or readonly-memory programmed to provide ⁇ where
  • FIG. 4 is a recursive arrangement which utilizes the inventive technique.
  • the input signal is initially compared to its predicted, or estimated value, by adder 31.
  • the estimate is based upon derived values produced from two previous code groups of the input signal.
  • the error now is weighted by a factor of 1/Q in sealer 32.
  • the output of sealer 32 is combined with a signal of magnitude equal to the prediction but opposite in sign which is provided by multiplier 33 to adder 34.
  • Adder 34 produces an output that is applied to 2-tap delay line comprising delays 36 and 37 in filter 38.
  • the other input to filter 38 is generated by adapter 39.
  • Adapter 39 generates a variable parameter by using the outputs of adder 31 and delay 36. This variable parameter weights the derived value from delay 36 so that adder 42 yields the predicted value.
  • the output of adapter 39 applied to frequency indicator 43 generates a variable value accurately related to the frequency of significant energy in the input signal. Due to the memory of filter 38 provided by the feedback signal component and the output of adapter 34 this arrangement may be typed as a second order recursive adaptive filter.
  • the performance characteristic of the arrangement of FIG. 4 is affected by the value of Q in sealer 32.
  • speed of convergence is high for low values of Q , e . g . , Q ⁇ 2, while high values of Q , e . g . , Q>6 , provide precision for the frequency output of indicator 43.
  • Both performance characteristics are appropriately desirable and may be provided by using a low value of Q for an initial response until it is determined that a tone or frequency component is present and then switched to a relatively high value of Q for a more precise determination of the frequency.
  • sealer 32 may be readily implemented by a select gate or a binary shift.
  • the arrangement of FIG. 4 in actual implementation represents a modest increase in complexity over the implementation of the arrangement of FIG. 3. This is more than outweighted by a higher degree of noise immunity for practical applications.
  • FIG. 5 is a generalized form of circuit which is capable of representing the circuitry of either FIG. 3 or FIG. 4.
  • the analytical basis of this circuit may be represented by various analytical relationships which follow.
  • z -1 and z -2 are the intervals of the delays used respectively to provide x 1 and x 2 in FIG. 5.
  • filter 51 additionally includes multipliers 52 and 53, and adder 54.
  • the additional components allow filter 51 to provide different, but related, signals to each of adders 56 and 57.
  • the signal for adder 56 is a prediction or estimate of the current signal from adapter 58 based upon signal history
  • the other output of filter 51 is the related signal which is applied to adder 57 serving to weight the past history.
  • the duration andthe relative weighting and time of the history is fed back into the filter to provide the overall circuit with a measure of noise immunity.
  • the output of frequency indicator 59 represents the frequency of the signal component in the input signal.
  • M and Q vary the operational characteristics of the circuit such that a family of estimation filters is represented.
  • M and Q respectively are the factor of multiplier 53 and the reciprocal of the factor of multiplier 61.
  • the following table lists the various types of filter responses via the circuit of FIG. 5 obtained using specific values or ranges of values for M and Q.
  • M and Q are not restricted to integer values since optimum performance in a specific application may require H and Q to have other values, such as intermediate values. However, in some specific implementation, powers of two values are less expensive to implement and sufficient.
  • FIG . 3 FIR 2 1
  • FIG . 4 I IR 2 > 1
  • FIGS . 7 & 8 IIR 1 >1
  • FIG. 6 is a flow chart of an overall adaptive filter procedure suitable for implementation on a general purpose computer capable of doing digital signal processing.
  • Digital signal processing microprocessors may be programmed to perform this procedure.
  • the compact integrated form of these devices is very desirable, but such a device should be capable of providing sufficient signal processing between successive signal values.
  • Three different symbols are utilized: oval symbols denote program labels and are used as descriptive entry points into the program; rectangular symbols, commonly referred to as operational blocks, require the performance of a particular operational step which may involve a number of subsidiary steps; and diamond shaped symbols, commonly referred to as either conditional branch points or decision blocks, require a test to be performed to determine the course of the following operation.
  • registers in the programmed machine designated IOC and AUC for input output control and arithmetic unit control are respectively set for the expected input and output data formats and the type of rounding or truncation to be used in performing the adaption.
  • Blocks 71 and 72 provide these operations.
  • all RAM locations are called to be cleared so that the read/write memory is available as a scratch pad.
  • Prime Bstart is called for in block 74 to determine the adaption starting frequency in accordance with the range of expected input frequencies.
  • the output register is set to zero to indicate that the threshold level has not been exceeded.
  • conditional branch point 77 the input register is examined to check for the arrival of a code word representing a new sample. If not, the program returns to WAIT1 to provide a refresh of the Prime Bstart in the RAM. (The type of RAM used required this refresh which may not be necessary in other implementations.) The looping back will continue until the test in conditional branch point 77 is met indicating that a new sample has arrived.
  • the block labeled filter 78 calls for an unconditional adaption and as a result changes the value of B. These two operations occur respectively in internal blocks 79 and 81 which are described in connection with FIG. 7.
  • the next step is conditional branch point 82 in filter 78 which determines if the exponentially mapped past value of the input signal (EMPS) has exceeded a threshold.
  • EMPS exponentially mapped past value of the input signal
  • This test is a level check equivalent to the full wave rectified value of the input signal stored in a resistive-capacitive integrator.
  • a typical threshold value for telecommunications applications may be on the order of 40 dBmO. Once this threshold is exceeded, it indicates that a true input signal is considered present so the procedure may be advanced to provide meaningful adaption. In addition it allows the filter transient to commence as adaption is held to the starting value.
  • the output register has its least significant bit set to one to indicate that a meaningful input signal has been recognized.
  • a loop counter is set to 63 before proceeding to block 88. This is used such that the adaption process will take place unconditionally for 64 successive input values. The duration of unconditional adaption in terms of input values is an engineering choice that may be readily changed to suit each application.
  • PRIME BTEST is set to &Btable (&BTBL). This provides an address for a location of a test value of B to be used later in the program. The procedure is now at a point where a location in ROM is pointed to by a value in RAM for access at a later step in the procedure.
  • conditional branch point 93 tests the threshold of the EMPS signal which may typically have a decay time constant of 4 ms. This also provides a bridging capability should the signal disappear for a short period of time before returning. The entire process is thus prevented from restarting upon the occurrence of a short dropout. If the dropout is not short so that the threshold is not met, then branch point 93 recycles the program back to operational block 72.
  • conditional branch point 96 is exited, the loop counter is decremented so block 103 is required to be reset to zero to eliminate the occurrence of a minus one.
  • the test of branch point 102 determines the ratio of the integrated value of the full wave rectified input signal to that of the integrated value of the full wave rectified error signal. The value of two or greater provides a signal-to- noise or error ratio of at least 6 dB. However, this is merely an engineering choice which may be readily changed with due consideration to the adaption performance for a specific application.
  • the values of BIGB, MIDB, and SMALLB respectively are 0.00821, 0.00410, and 0.00041. These values produce the largest frequency change at 700 Hz. Frequency changes of 20 Hz correspond to each BIGB change, 10 Hz for each MIDB change, and finally 1 Hz for each change in SMALLB. These are the frequency changes at 700 Hz. If this test is not satisfied then further correction is required of B for convergence and the procedure advances to block 113 wherein middle B (MIDB) is used to provide moderate steps in the adjustment of the value of B when the flow loops back to REGPRIME which will again utilize the operations of filter 91. The flow during loop back passes through operational block 105 which calls for an output word to indicate that B is not suitable to obtain frequency information.
  • middle B MIDB
  • Block 104 calls for B to be changed in fine steps. Then the flow proceeds to block 106 calling for the value of BLOW to be obtained from STABLE.
  • conditional branch point 107 the value of the incremented B from filter 91 is compared to BLOW. If this test is not satisfied it indicates that the actual frequency is below the lower tolerance of the frequency tested and the sequence proceeds to block 101 to indicate that a suitable test result is not to be indicated. If the test of branch point 107 is positive, then branch point 108 is next. At branch point 108, the value of B is checked or compared to BHIGH which is also obtained from STABLE at an address an increment of one away from BLOW. If this condition is satisfied, it indicates that the value of B is between the upper and the lower frequency limits and, hence, the frequency is within tolerance. Accordingly, block 109 may now indicate an output for that particular frequency value.
  • This output for frequency, or test result, actually is called to be indicated for the first time by block 105. If this jtest is not satisfied, it indicates that the frequency actually adapted is not within an acceptable frequency range. Then block 111 points to the next higher frequency range in SBTBL as the process then recycles back to PREADAPT for further adjustment of the value of B using the next input value.
  • Operational block 111 functions as a moving pointer for successive frequency ranges in ascending value corresponding to ascending address locations in the SBTBL.
  • the value of B is not within the limits of the range bounded by BLOW and BHIGH and the next higher frequency range is used when the tests of conditional branch points 107 and 108 are subsequently applied.
  • a value of BTEST greater than any possible value for B is used to recycle the sequence back to point 89. Then as the sequence subsequently enters point 107, it loops back to REGPRIME reinitializing the table process.
  • FIG. 7 is a diagram of the basic adaptive filter subroutine called at two points in FIG. 6. In block 120 a new sample of the input signal is obtained from ibf.
  • the estimate of the new sample is generated using x 1 and x 2 , which are previously derived values from the two preceding samples, and B the frequency related variable.
  • the value of B is usually the result of a previous iteration of this subroutine.
  • the estimated value is stored in a RAM.
  • block 122 calls for determining the error ⁇ of the estimate by subtracting it from the new, or current, sample. The value of the error is also stored.
  • the value of g is calculated using the same quantities as that of block 121.
  • Block 124 serves to form x o using the error ⁇ divided by Q added to the value of g.
  • the new values of x 1 and x 2 are obtained in block 126 by respectively using the previous values of x 1 and x o . These new values are stored to be used in the next cycle through the subroutine of FIG. 7.
  • the previous value of B is corrected for use in determining the next estimate for the next successive value of the input signal which is obtained in accordance with block 120 when the process repeats itself as required according to FIG. 6.
  • FIG. 8 illustrates a typical filter frequency response characteristic. The filter output is normalized to represent the gain of the overall arrangement.
  • FIG. 9 depicts an arrangement for utilizing the inventive principles for adaptively filtering multiple frequencies in a signal.
  • This arrangement provides concurrent multiple frequency convergence.
  • the input signal is applied to adaptive filter 131 which will serve to attenuate one frequency signal component to suppress it while passing on the remaining frequency signal component.
  • the output of filter 131 is applied as the input of adaptive filter 132 which will be left with the after frequency of the two frequencies at the input of filter 131.
  • Filter 132 operates in the same manner as described for the single frequency adaptive filters. Accordingly, the error signal and the inphase signal are applied to adaptive generator 133 which provides the variable 2 related to the input frequency of the component applied to filter 132.
  • the inphase signal from filter 131 and the error signal from filter 132 are applied to adaptive generator 134.
  • phase shifter 136 has the same internal circuitry as filter 132 so that its phase shift may be adaptively varied as directed by the variable 2.
  • filter 132 attenuates one frequency component of the input signal while filter 131 attenuates the other frequency of the input signal.
  • frequency indicator 137 provides an output representative of the two frequencies of convergence of filter 131 and 132.
  • the filtered output signal ⁇ 2 o is also available.
  • FIG. 10 extends the inventive concept for simultaneously filtering three frequencies.
  • adaptive filters 141, 142, and 143 respectively, filter frequencies F1, F2 and F3.
  • Filter 143 adapts to the frequency component which is left remaining after the input signal passes through filters 141 and 142.
  • the operation of filter 143 and adaptive generator 144 is similar to filter 132 and generator 133 in FIG. 9.
  • the error signal from filter 143 is used in the adaption process by all three adaptive generators 144, 147, and 151.
  • the inphase signal from filter 142 is phase adjusted by phase shifter 146 before application to adaptive generator 147 which not only adjusts filter 142 but also phase shifter 148.
  • the inphase signal from filter 141 is phase adjusted twice, first by phase shifter 148 and then by phase shifter 149.
  • Each phase shifter has the same type of internal circuitry as the adaptive filter driven by the same variable value from the respective one of the adaptive generators. This enables the inphase signal to have the appropriate phase relationship with the error signal from filter 143 for application to adaptive generator 151 to adjust filter 141.
  • Frequency indicator 152 uses the three variable values to provide an output representative of the three frequency components of the input signal.
  • each of filters 142 and 143 are adapted to an input signal which respectively has one or both of the remaining interfering frequencies attenuated by the other filters in the cascade arrangement.
  • this arrangement allows each filter to operate as though the other frequencies and filters are not present.
  • the filtered output ⁇ 3 o is available in the arrangement of FIG. 10.

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  • Computer Networks & Wireless Communication (AREA)
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Abstract

Plusieurs modes de realisation d'une technique sont presentes pour tirer profit de la correlation entre des signaux de maniere a fournir des informations utilisables concernant la frequence. Dans un mode specifique de realisation, un filtre (51) est adapte a une composante de signal d'un signal d'entree utilisant une variable generee par un adaptateur (58). La variable possede une relation determinee avec la frequence de la composante du signal. La variable est utilisee pour ponderer une combinaison de valeurs de signaux derivees d'un historique de signaux pour former une prediction d'un signal futur. Un additionneur (56) combine la nouvelle valeur du signal avec la prediction pour former un signal d'erreur qui est utilise par le filtre (51) et l'adaptateur (58). Lorsque le filtre converge, un indicateur de frequence (59) utilisant la variable produit un signal de sortie indiquant la frequence de la composante du signal.
PCT/US1982/000664 1981-06-15 1982-05-17 Techniques d'adaptation pour la mesure et la determination automatiques de la frequence WO1982004488A1 (fr)

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US06/273,611 US4403298A (en) 1981-06-15 1981-06-15 Adaptive techniques for automatic frequency determination and measurement
US273611810615 1981-06-15

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FR2622021A1 (fr) * 1987-10-16 1989-04-21 Trt Telecom Radio Electr Dispositif pour mesurer la distance " h " qui le separe d'un objet

Also Published As

Publication number Publication date
EP0081556A4 (fr) 1984-08-20
GB2100537A (en) 1982-12-22
US4403298A (en) 1983-09-06
CA1186750A (fr) 1985-05-07
JPS58500925A (ja) 1983-06-02
GB2100537B (en) 1985-03-06
EP0081556A1 (fr) 1983-06-22

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