WO2003075482A1 - Automatic equalizer and coefficient training method thereof - Google Patents

Automatic equalizer and coefficient training method thereof Download PDF

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Publication number
WO2003075482A1
WO2003075482A1 PCT/JP2002/001085 JP0201085W WO03075482A1 WO 2003075482 A1 WO2003075482 A1 WO 2003075482A1 JP 0201085 W JP0201085 W JP 0201085W WO 03075482 A1 WO03075482 A1 WO 03075482A1
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WIPO (PCT)
Prior art keywords
equalizer
coefficient
target
output
transfer function
Prior art date
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PCT/JP2002/001085
Other languages
French (fr)
Japanese (ja)
Inventor
Mitsuo Kakuishi
Nobukazu Koizumi
Hideyuki Araki
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Fujitsu Limited
Priority date (The priority date 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 date listed.)
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Publication date
Application filed by Fujitsu Limited filed Critical Fujitsu Limited
Priority to PCT/JP2002/001085 priority Critical patent/WO2003075482A1/en
Publication of WO2003075482A1 publication Critical patent/WO2003075482A1/en

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Classifications

    • HELECTRICITY
    • H04ELECTRIC COMMUNICATION TECHNIQUE
    • H04LTRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
    • H04L25/00Baseband systems
    • H04L25/02Details ; arrangements for supplying electrical power along data transmission lines
    • H04L25/03Shaping networks in transmitter or receiver, e.g. adaptive shaping networks
    • H04L25/03006Arrangements for removing intersymbol interference
    • H04L25/03012Arrangements for removing intersymbol interference operating in the time domain
    • H04L25/03019Arrangements for removing intersymbol interference operating in the time domain adaptive, i.e. capable of adjustment during data reception
    • H04L25/03057Arrangements for removing intersymbol interference operating in the time domain adaptive, i.e. capable of adjustment during data reception with a recursive structure
    • HELECTRICITY
    • H04ELECTRIC COMMUNICATION TECHNIQUE
    • H04BTRANSMISSION
    • H04B3/00Line transmission systems
    • H04B3/02Details
    • H04B3/04Control of transmission; Equalising
    • H04B3/14Control of transmission; Equalising characterised by the equalising network used
    • H04B3/142Control of transmission; Equalising characterised by the equalising network used using echo-equalisers, e.g. transversal
    • HELECTRICITY
    • H04ELECTRIC COMMUNICATION TECHNIQUE
    • H04LTRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
    • H04L25/00Baseband systems
    • H04L25/02Details ; arrangements for supplying electrical power along data transmission lines
    • H04L25/03Shaping networks in transmitter or receiver, e.g. adaptive shaping networks
    • H04L25/03006Arrangements for removing intersymbol interference
    • H04L2025/03433Arrangements for removing intersymbol interference characterised by equaliser structure
    • H04L2025/03439Fixed structures
    • H04L2025/03445Time domain
    • H04L2025/03471Tapped delay lines
    • H04L2025/03484Tapped delay lines time-recursive
    • H04L2025/0349Tapped delay lines time-recursive as a feedback filter

Definitions

  • the present invention relates to an automatic equalizer and a coefficient training method thereof, and is suitable for use, for example, in an apparatus for receiving a signal transmitted by a discrete multi-tone (Discrete Multi-Tone) modulation method using a subscriber line pair cable as a transmission path.
  • the present invention relates to an automatic equalizer and its coefficient training method.
  • XDSL Digital Subscriber Line
  • This xDSL is a transmission method using the existing subscriber line (pair cable) and is one of the modulation and demodulation technologies.
  • x DSL is broadly divided into the upstream transmission speed from the subscriber's home (hereafter, the subscriber side) to the accommodation station (hereafter, the office side) and the downstream transmission speed from the office side to the subscriber side.
  • the subscriber side home
  • accommodation station hereafter, the office side
  • the downstream transmission speed from the office side to the subscriber side are categorized as symmetric and asymmetric.
  • a symmetric type has a high-bit-rate DSL (HD SL) with an uplink and downlink transmission speed of about 1.5 to 2.0 Mbps (megabit-noise), and a 160 k to 2
  • HD SL high-bit-rate DSL
  • SDSL Single-line DSL
  • AD SL Asymmetric DSL
  • the AD SL further includes "G.dmt” having a downlink transmission speed of about 8 Mbps and "G.lite” (also called a simplified AD SL) having a downlink transmission rate of about 1.5 Mbps.
  • G.dmt having a downlink transmission speed of about 8 Mbps
  • G.lite also called a simplified AD SL
  • DMT Discrete Multi-Tone
  • This DMT modulation method is, in a nutshell, a subcarrier whose transmission frequency band is about every 4 kHz (in the case of “G.lite”, depending on the conditions, up to 128 carriers in the downlink direction). And modulates each carrier.
  • This DMT modulation method has a feature that it is resistant to noise of a specific frequency because communication with another subcarrier is possible even if a certain subcarrier becomes unusable due to the influence of noise having a specific frequency.
  • DMT modulation method The details of the principle of the DMT modulation method are described in, for example, the document "JACBingham, Multicarrier Modulation For Data Transmission An Idea Whose Time has Come", IEEE Co. n. Mag., Pp5-14, May, 1990J. I have.
  • DSL digital subscriber line
  • the ADSL transmission system focuses on its main components.
  • a device (AD SL modem) 210 is connected to each other via a transmission line (metallic line) 300 such as a subscriber line pair cable.
  • the subscriber-side modem 110 includes a DMT modulator 101, a digital transmission filter 102, a digital Z-analog (D / A) converter 103, an analog band-pass filter as a transmission system. (BPF: Band Pass Filter) 104, etc., and the station-side modem 210 has a transformer 201, analog low-pass filter (LPF) 202, AZD converter 203, digital B PF 204, Time-domain Equalizer (TEQ) 205, Fast Fourier Transformer (FFT) 206, Frequency-domain Equalizer (FEQ) 207, Classifier 208 Etc. are provided.
  • LPF analog low-pass filter
  • FFT Fast Fourier Transformer
  • FEQ Frequency-domain Equalizer
  • the ADSL modem 110 of the subscriber 100 receives the signal from the ADSL modem 210 of the office 200 (hereinafter referred to as the office modem 210).
  • a receiving system having the same function as the system is provided, and the station-side modem 210 is provided with a transmitting system having the same function as the transmitting system in the subscriber-side modem 110.
  • the communication of (2) is performed in the same manner as the uplink communication described below.
  • the DMT modulation section 101 is for performing DMT modulation on transmission data transmitted from a subscriber terminal such as a personal computer (PC).
  • a subscriber terminal such as a personal computer (PC).
  • PC personal computer
  • a serial-to-parallel buffer Serial to Parallel Buffer
  • Encoder encoder
  • IFFT inverse fast Fourier transformer
  • the serial-to-parallel buffer 111 stores transmission data, which is serial data, for one symbol time (approximately l / 4 kHz), converts the stored data into parallel data, and outputs the parallel data. According to a predetermined transmission bitmap (not shown), the number of transmission bits is assigned to each carrier (subcarrier) (frequency band division).
  • the entire M-bit data is simultaneously transmitted on Nc carrier waves as a whole, and the receiving side (station side 200) transforms Ns time data (symbols) into a fast Fourier transform (FFT: Fast).
  • FFT fast Fourier transform
  • M bits of information are extracted from the amplitude and phase of the Nc carrier waves.
  • each carrier is placed at a distance of Afc (about 4 kHz) in the used frequency band.
  • Afc about 4 kHz
  • the number of symbols Ns is a power of 2 It is common to use "64" or "256".
  • the encoder 112 performs quadrature amplitude modulation (for example, QAM (Quadrature Amplitude Modulation)) on the bit string output from the serial-parallel buffer 111 by changing the amplitude and phase for each carrier.
  • the IFFT 113 performs inverse fast Fourier transform on the signal point data (data in the frequency domain) output from the encoder 112 to obtain the signal point. It converts the data into Ns time-domain signal sequences. As a result, the M-bit data is converted into a time-domain signal (symbol) consisting of Ns sample values and transmitted.
  • the digital transmission filter 102 is for removing unnecessary components of the symbol sequence (digital data) obtained by the DMT modulator 101, and the 0/8 converter 103
  • the output of the digital transmission filter 102 is converted to analog data, and the analog BPF 104 is for removing unnecessary components of the output of the DZA converter 103.
  • the transformer 201 is for cutting off the direct current (dc) component of the signal received from the transmission line 300
  • the analog LPF 202 is for cutting off the high frequency component signal and the noise component.
  • the AZD converter 203 converts the received signal (analog signal) passed through the analog LPF 202 into a digital signal.
  • the digital BPF 204 removes noise components and the like outside the desired band from the output of the AZD converter 203, and the TEQ 205 generates inter-symbol interference (ISI: Inter Symbol Interference) with the input signal on the transmission side ( This is to perform predetermined processing so that it falls within the cyclic prefix added on the subscriber side 100).
  • ISI Inter Symbol Interference
  • one point of the DMT modulation method is that by providing a period called a guard time between symbol sequences, demodulation on the receiving side can be made less susceptible to transmission line distortion.
  • a part (L sample) corresponding to the guard time at the end of the symbol sent after that is sent in advance is a cycle
  • the TEQ 205 adaptively updates its own coefficient (tap coefficient), and determines the length of the impulse response from the output of the transmitting IFFT 113 to the receiving FFT 206 via the transmission path 300 by the guard time. It works as follows. However, in practice, it is difficult to completely reduce the portion exceeding the guard time to zero, and the size is reduced as much as possible.
  • the purpose of the TEQ205 applied to the ADSL transmission system is quite different from that of the equalizer used in systems other than the DMT modulation system.
  • the frequency characteristics of the amplitude and group delay time are flattened. Is the main purpose.
  • the TEQ 205 itself is a FIR (Finite Impulse Response) type transversal equalizer, and has the same configuration as the equalizer used in the QAM system and the like.
  • the coefficient is updated based on the difference between the equalized output of the TEQ 205 and the output of the channel target equalizer 205a, which is obtained by the adder 205b (the details will be described later).
  • the guard time is lengthened, the TEQ 205 is unnecessary or the required order can be reduced.However, if the guard time is lengthened, the transmission efficiency deteriorates, so the guard time is selected to be about 1Z16 of the symbol time. Is common.
  • the FFT 206 is for converting the output after equalization by the TEQ 205 into frequency-domain data by the fast Fourier transform
  • the discriminator 208 includes a real part and an imaginary part of the output of the FEQ 207. Each of them is converted to digital data (digital I and Q values) (hereinafter referred to as identification processing).
  • the obtained digital data is output to a subsequent DMT demodulation section (not shown) as received data, and the DMT demodulation section modulates the data at a DMT modulation section 101 on the transmission side (subscriber side 100).
  • the reverse processing decoding, parallel-to-serial conversion, etc.
  • the signal is orthogonally demodulated.
  • the reception level greatly varies depending on the length of the transmission path 300, and therefore, an AGC converter for making the input level to the A / D converter 203 almost constant before the AZD converter 203 is used. (Automatic Gain Controlled)
  • a digital AGC is inserted after the AZD converter 203 in order to reduce the operation word length in digital signal processing. Illustration is omitted.
  • the operation of the ADSL transmission system configured as described above will be described.
  • the transmission data is held in the serial-parallel buffer 111 for one symbol time (1/4 kHz).
  • the stored data is divided for each predetermined number of transmission bits per carrier and output to the encoder 112.
  • the encoder 112 converts each of the input bit strings into signal points for quadrature amplitude modulation, and outputs the signal points to the IFFT 113.
  • IFFT 113 quadrature amplitude modulation is performed on each signal point by performing an inverse fast Fourier transform on the output of encoder 112.
  • a predetermined sample of the output of IFFT 113 is added to the head of the symbol as a cyclic prefix.
  • the IF FT output to which the cyclic prefix has been added as described above is input to a DZA converter 103 after unnecessary components have been removed by a digital transmission filter 102, where a predetermined sampling frequency (for example, 1.104 MHz ), Is converted into an analog signal, and is output to the transmission path 300.
  • a predetermined sampling frequency for example, 1.104 MHz
  • the analog signal received through the transmission path 300 is input to the analog LPF 202 via the transformer 201, and after the unnecessary components are removed by the analog LPF 202, the AZD converter 203 At digital Converted to a signal.
  • the obtained digital signal is input to a TEQ 205 after further removing unnecessary components by a digital BPF 204, and the length of the impulse response of the reception path is determined by the TEQ 205 to the guard time (size). (The click prefix).
  • the equalized signal is input to the FFT 206, where it is subjected to a fast Fourier transform to be converted into frequency-domain signal point data.
  • the obtained signal point data is subjected to the FEQ 207 to compensate for the influence on the amplitude and phase caused by passing through the transmission path 300 for each carrier having a different frequency.
  • the data is converted to digital data (digital I and Q values) and output to the subsequent processing unit (DMT demodulation unit) as received data.
  • a method of obtaining the tap coefficient of TEQ 205 described above (coefficient update) will be described below.
  • the method of obtaining such a coefficient is described in, for example, US Patent 5,285,474, 'Method for Equalizing A Multi-carrier Communication System'.
  • a circuit block (equalizer) called “channel target”.
  • channel target is a circuit with a transversal filter configuration having transfer characteristics equivalent to the transmission system from the IFFT output of the transmitting side (subscriber side 100) to the FFT input of the receiving side (station side 200).
  • the block length is set to L or less, which is the number of taps equivalent to the guard time.
  • the same signal (training signal) is sent to this “channel target” and the transmission system (hereinafter referred to as “real route”) from the transmitting IFFT output to the receiving FFT input, and the difference between their outputs Adaptively change the coefficients of the TEQ 205 included in the transversal filter of the channel target 205a (hereafter referred to as the target equalizer 205a) and the actual route so that is always zero.
  • the target equalizer 205a Adaptively change the coefficients of the TEQ 205 included in the transversal filter of the channel target 205a
  • the actual route so that is always zero.
  • a training time is provided, during which the same data sequence as the input data sequence of the transmission side (subscriber side 100) generated by the transmission side random bit sequencer 121 is sent to the pseudo random bit sequencer 205.
  • Target equalization by c The output of the TEQ 205 and the output of the evening get equalizer 205a are compared (added) by the adder 205b, and the evening get equalizer is set so that the difference becomes small.
  • the tap coefficients of 205a and TEQ205 are changed adaptively.
  • the impulse response between the transmitting-side IFFT output and the receiving-side FFT input is almost the same as that of the target equalizer 205a.
  • the impulse response of the evening gate is the tap coefficient itself of the target equalizer 205a, and its time width is shorter than the guard time, so that the impulse response between the transmitter I FFT output and the receiver FFT input is The length is also equal to or less than the guard time, and the effect of transmission line distortion on DMT modulation can be suppressed.
  • the method of obtaining the values on the time axis uses a conventionally well-known transversal filter coefficient updating method. That is, the tap coefficient is updated using the difference between the two outputs and the output value of the delay element in the transversal filter. It should be noted that when a solution is obtained in this time range, an extremely long transmission path 300 involves an extremely large time delay in an actual route. Therefore, such a delay is dealt with by correcting a delay of an integral multiple of the sampling period using a delay unit 205d shown in FIG.
  • the characteristics of both the TEQ 205 and the target equalizer 205a are matched as much as possible regardless of the time axis and the frequency axis.
  • the target characteristic simulates the characteristic from the transmitter IFFT output to the receiver FFT input, so that the gain is sufficient in the frequency band where the transmission signal passes, compared to the other bands. High, that is, in a band through which the transmission signal passes, the difference in signal delay due to frequency in the pass band is small, and it may be incomplete, but it is desirable that delay equalization be performed.
  • a range from about 26 kHz corresponding to sub-channel “6” to about 138 kHz corresponding to sub-channel “32” is assigned as a frequency band from the subscriber to the central office, and the maximum passband frequency is 138 kHz.
  • the target equalizer 205a of the DMT modulation method processing at 552 kHz, which is four times the frequency of the bandpass filter, should have the initial value of the transfer function of the band-pass filter with a pass band of approximately 26 to 138 kHz. is there.
  • the coefficient of the channel target 205 is fixed, the coefficient of the TEQ 205 is made variable, and when the TEQ coefficient stops moving, the coefficient of the target equalizer is made variable to further reduce the difference. By adopting it, the characteristics of TEQ205 can be obtained efficiently.
  • the TEQ205 makes the amplitude of the part of the impulse response of the actual route that is longer than the guard time width extremely small, and each subchannel obtained as the output of the FFT206 The received data is not interfered with by other subchannel signals of the current symbol, nor by the signal of the previous symbol (that is, ISI is as small as possible).
  • the S / N deteriorates, and the amount of information (number of bits) transmitted through the sub-channels decreases.
  • the number of bits that can be arranged for each sub-channel is reduced by the noise of the transmission path 300 in addition to the above-mentioned intra-channel interference and inter-channel interference.
  • the total number of bits that can be arranged for each subchannel is the total number of arranged bits per symbol, and the total number of arranged bits multiplied by the symbol repetition frequency is the throughput.
  • the in-band characteristics of the TEQ205 are gently high-pass filter characteristics that equalize the line characteristics of the line, and when the line length is zero, The in-band characteristics of the TEQ 205 are flat loss characteristics.
  • a “bridge tap” is an open-ended branch line of an unspecified length for a branch in the middle of a track.
  • the transfer characteristic often has a large loss in a specific frequency band of the passband, and cannot be handled by the conventional TEQ 205, and there is a large loss between the actual route and the target route. The difference remains, and as a result, the length of the impulse response of the actual route cannot be shortened, and the total number of arranged bits often decreases.
  • TEQ 205 is an FIR-type adaptive equalizer, and therefore is not suitable for equalizing a peak of a loss centered on a specific frequency. That is, in general, in an FIR type adaptive equalizer such as a transpersal type equalizer, the transfer function is only a numerator function and no denominator function is present, so a steep loss peak can be realized, but a steep gain It is not suitable for realizing the characteristics of the peak (that is, compensating for the steep valley of the gain).
  • IIR type equalizers cannot be used as TEQ 205.Therefore, when a “bridge tap” is present, there is a problem that sufficient good throughput may not be obtained. Was.
  • the method of adaptively finding the coefficient is not clear, for example, one having a simple F (frequency) characteristic such as an emphasis circuit that changes the coefficient stepwise in association with the AGC gain At present, it was not used for any other purpose.
  • the present invention has been made in view of such a problem, and has found a method for stably and adaptively obtaining coefficients of an IIR type equalizer, thereby sufficiently compensating for a steep gain valley of a received signal.
  • the purpose is to be able to use IIR type equalizers in addition to FIR type equalizers.
  • an automatic equalizer of the present invention adaptively performs equalization processing on a signal received from a predetermined transmission path, and has the following units. It is characterized by.
  • An equalizer that shortens the length of the impulse response characteristics of the reception transmission line (from the transmission IFFT output to the FFT input) (hereinafter referred to as a real route equalizer)
  • the target equalizer that simulates the impulse response characteristics of the route up to the output of the real route equalizer including the transmission path
  • the tap coefficients of the target equalizer are updated based on the output of the real root equalizer and the output of the evening get equalizer, and a part of the updated tap coefficients ( ) Is used to set some tap coefficients of the real root equalizer.
  • the tap coefficient of the evening equalizer is calculated based on the output of the real root equalizer and the output of the target equalizer during the coefficient training period. Is updated and the transfer function of the target equalizer.
  • the tap coefficients of the IIR type equalizer can be set so that some reciprocals of the factorized factor are used as transfer functions.
  • an IIR type equalizer having a denominator function part as a transfer function and suitable for compensating for a steep valley in the gain of a received signal, and thereby, regardless of the conditions of the transmission path, It is possible to always obtain a good transmission amount, that is, a throughput.
  • FIG. 1 and 2 are block diagrams illustrating the principle of the present invention.
  • FIG. 3 is a block diagram showing a configuration of the automatic equalizer according to the first embodiment of the present invention.
  • 4 to 7 are block diagrams for explaining the function of the coefficient update block shown in FIG.
  • FIG. 8 is a block diagram showing a configuration example of the first-channel first-get equalizer shown in FIGS.
  • FIG. 9 is a block diagram showing a configuration example of the second channel sunset-equalizer shown in FIGS.
  • FIG. 10 is a block diagram showing a configuration example of the transversal equalizer (T EQ) shown in FIG.
  • FIG. 11 is a block diagram showing a configuration example of the recursive equalizer shown in FIG.
  • FIG. 12 is a diagram for explaining pole determination during coefficient training according to the present embodiment.
  • FIG. 13 is a block diagram showing a configuration in a case where the recursive equalizer shown in FIGS. 4 to 7 is arranged after TEQU.
  • FIG. 14 is a block diagram for explaining a function of a coefficient update block of the automatic equalizer according to the second embodiment of the present invention.
  • FIG. 15 is a diagram for explaining the operation of the coefficient update block according to the second embodiment.
  • FIG. 16 is a block diagram for explaining the function of the coefficient update block according to the second embodiment after the completion of coefficient training.
  • Figure 17 is a block diagram showing the configuration of a conventional digital subscriber line (AD SL) transmission system. It is a lock figure.
  • AD SL digital subscriber line
  • FIG. 18 is a block diagram showing a configuration of the DMT modulator shown in FIG.
  • FIG. 19 is a block diagram for explaining the method of updating the TEQ coefficient shown in FIG. BEST MODE FOR CARRYING OUT THE INVENTION
  • the point of the present invention is to adaptively obtain the coefficients of the IIR type equalizer on the receiving system (real route) side input to the FFT using the target equalizer.
  • a pseudo transmission path (target route) is provided. Therefore, a system as shown in Fig. 1 can be considered because it can be compared with the signal from the actual transmission path (real route).
  • the transfer function of the transmission path is HI (z) (z—, the transfer function of TEQ is H
  • Figure 3 shows an automatic equalizer configuration that achieves this.
  • the configuration shown in FIG. 3 is also applied to, for example, the office-side modem 210 shown in FIG. 17, where 1 is a recursive equalizer, 2 is a TEQ equivalent to the TEQ 205 shown in FIG. 17, 3 is a window block, and 4 is a diagram.
  • FFT equivalent to FFT 206 shown in 17 and 5 show a coefficient update block, respectively.
  • the recoverable equalizer 1 converts the input signal (the output of the digital BP F 204 in the case of FIG.
  • H teq d This is an IIR (Infinite Impulse Response) type equalizer using z as the denominator of the transfer function.
  • IIR Infinite Impulse Response
  • z the denominator of the transfer function.
  • it is configured as a 3-tap IIR type equalizer as shown in FIG. 11.
  • predetermined coefficients are respectively multiplied by the multipliers 15 and 16.
  • the sum is obtained by the adder 12, and the result is added to the current input signal by the adder 11, so that an equalized output for the input signal can be obtained. ing.
  • FIR type equalizer transversal type equalizer
  • the sum by the adder 23 is obtained as an equalized output with respect to the input signal (the output of the reciprocating equalizer 1).
  • the number of taps in TEQ2 is based on the number of samples L corresponding to the guard time length, but there is no problem if it is larger than that. Also, their initial values may all be "0.0".
  • the window block 4 is for delaying the output of TEQ2 by a predetermined time and inputting it to the FFT 4
  • the coefficient update block 5 is for recursive equalizer 1
  • the coefficient of TEQ 2 are adaptively updated based on the output of TEQ 2.
  • the recursive switching signal and the TEQ switching signal pass the recursive equalizer 1 through the input signal (through). )
  • the state can be changed, and the updated coefficient value can be reflected (set) in the reactive equalizer 1.
  • the functional units shown in FIGS. 4 and 6 are realized by software processing. That is, as shown in FIGS.
  • the coefficient update block 5 includes a pseudo-random bit sequencer 51, a delay block 52, a first channel target equalizer 53, and a second channel target equalizer. It has an adder 54, an adder 55 and an inverse second channel evening get function block 56.
  • the pseudo-random bit sequencer 51 generates the same pseudo-random signal (training signal) as the pseudo-random bit sequencer 12 1 on the transmitting side described above with reference to FIG. 19, and the delay block 52
  • the delay time of the pseudo-random signal on the target route side input from the pseudo-random pit sequencer 51 is adjusted, and the pseudo-random bit sequencer 121 on the transmitting side is input to the TEQ 2 via the transmission path 300 by the pseudo-random bit sequencer 121. This is to match the delay time of the pseudo-random signal on the real route side with the delay time of the pseudo-random signal on the target route side.
  • the delay block 52 may be provided on the real route side (the output side of TEQ 2) (the delay time of the pseudo random signal on the real route side may be adjusted).
  • the first channel evening get equalizer (hereinafter referred to as the first target equalizer) 53 acts on the output (pseudo-random signal) of the delay block 52, for example, as shown in FIG.
  • the first target equalizer acts on the output (pseudo-random signal) of the delay block 52, for example, as shown in FIG.
  • FIR type transversal type
  • the first target equalizer 53 the number of registers (delay elements) 531 corresponding to the number of taps, the time axis obtained by holding the input signals one by one in time series, is obtained.
  • the sample data at the above plural points are multiplied by a multiplier coefficient in a multiplier 532, respectively, and the sum of those multipliers is added to the input signal (output of the delay block 52). It can be obtained as output.
  • the number of taps in the first evening get equalizer 53 is equal to the number of taps corresponding to the guard time length described above. A value close to the number L of pulls, and the initial value uses a value preset by the above-described method.
  • the second channel target equalizer (subordinate equalizer) 54 acts on the post-equalization output of the first evening equalizer 53.
  • the first target equalizer 54 is used. It is configured as an FIR type equalizer in the same way as 53.
  • the second channel target equalizer (hereinafter, referred to as the second target equalizer) 54 has a lower order (2 to 4 order; 3 to 4) than the first target equalizer 53 described above.
  • the tap coefficients of the recursive equalizer 1 that will be the denominator function part on the real route side are finally set using the tap coefficients, the tap coefficient of the first tap is considered.
  • the numerical value is fixed to "1.0", and the other tap coefficients are initially set to "0.0".
  • FIG. 9 shows the configuration of the second target equalizer 54 in the case of three taps.
  • the time axis obtained by holding the input signals one by one in time series in the two resistors (delay elements) 54 1 is obtained.
  • Multipliers 542, 543, and 544 multiply the sample data of the above three points by tap coefficients, and the sum of those multipliers is added to the input signal (first target equalizer). (The output of the device 53).
  • the adder 55 obtains a difference between the output of the TEQ 2 and the output of the second target equalizer 54 (an equalization error signal e; hereinafter, simply abbreviated as “error e”).
  • error e an equalization error signal
  • the coefficients of the TEQ 2, the first target equalizer 53, and the second target equalizer 54 are updated so that the error e is minimized.
  • the error e 'taking into account the deformation in the second target equalizer must be used instead of the error e obtained by the adder 55. No. This is because the error used for updating the coefficients of the transversal equalizer must be the error of the (immediate) output terminal.
  • an inverting amplifier gain is “1.0”, but the phase is rotated by 180 degrees
  • the error e ′ is obtained by the inverse second channel target function block (Yuichi get output point error calculation block) 5 6, and the inverse of the transfer function of the equalizer of the second target equalizer 54 is It has a transfer function, and the error obtained by the adder 55 is equalized in the time domain by the equalizer function to obtain an error e 'at the output point of the first target equalizer 53. It is used for updating the coefficient of the first target equalizer 53.
  • the coefficient update block 5 of the present embodiment includes the adder 55, the inverse second channel target function block 56, the first evening get equalizer 53, and the second evening get equalizer 5. It is composed of a first difference calculation updating unit that updates each coefficient of 4 and TEQ 2.
  • the pseudo-random bit sequencers 12 1 and 51 add the same pattern of pseudo-random signals to the real route (TEQ 2) and the target route (delay block 52). .
  • the recursive equalizer 1 is set to the through setting by the recursive switching signal, the coefficient of the first target equalizer 53 is fixed, and the TEQ 2 and the second evening are set.
  • the coefficients of the one-get equalizer 54 are made variable, and the coefficients are updated adaptively (first acquisition process).
  • the coefficient updating method (automatic equalization algorithm) is that the target route output (the output of the second target equalizer 54) is subtracted from the actual route output (the output of TEQ 2) by the adder 55.
  • the error e obtained by the above, a well-known method of adaptively approaching the coefficient value in the multiplier in the transversal type equalizers 2, 54 to the best value is used. is there.
  • the second correction algorithm for the coefficient C (j) is expressed by the following equation (3).
  • Xj M represents the input signal sequence of TEQ2
  • e (v) represents the error.
  • the details of the automatic equalization algorithm are described in, for example, the document “Adaptive Signal Processing” (edited by Shokodo Shigeo Tsujii, first published on May 15, 1995, p. 24), and described in other documents other than the above. It is also possible to apply various algorithms.
  • the coefficient update block 5 next performs a second pull-in process in which the coefficient of the first target equalizer 53 is also made variable and the error e is further reduced. .
  • the recursive equalizer 1 and TEQ 2 are maintained in the through setting by the recursive switching signal.
  • the coefficient update block 5 performs the recursive equalizer 1 so that the transfer function of the recursive equalizer 1 becomes the reciprocal of the transfer function of the second target equalizer 54.
  • Set the tap coefficient of That is, the tap coefficients to be multiplied by multipliers 543 and 544 in FIG. 9 are set to the tap coefficients to be multiplied by multipliers 15 and 16 in FIG. 11, respectively.
  • the second target equalizer 54 on the target route side performs a process of setting to through, and the recursive equalizer 1 is set to non-through by a recursive switching signal.
  • the coefficient update block 5 makes the coefficient of the TEQ 2 and the coefficient of the first target equalizer 53 variable, and updates the coefficient so that the error e is further reduced.
  • the coefficient of the Jamaica equalizer 1 is fixed.
  • the tap coefficient is later set as the tap coefficient of the recursive equalizer 1, and the pole of the transfer function of the recursive equalizer 1 is set on the complex plane. It is necessary to check the updated value so that it is not placed outside the unit circle (hereinafter simply referred to as “unit circle”).
  • b O. 9, b ⁇ a—0.9, b ⁇ —a—0.9 is derived (actually, it is 1.0 instead of 0.9, In the case of actual operation, the margin is set to 0.9 in consideration of the effects of coefficient rounding, etc.). This indicates that it suffices that a and b exist within the triangular area shown in FIG.
  • the coefficient update block 5 controls the coefficient update by software so as to satisfy this condition. That is, the coefficient update block 5 sets the tap coefficients in the multipliers 542 and 543 to the multipliers 15 and 16 as they are when the pole exists outside the triangular area shown in FIG. If they exist outside the triangle area shown in Fig. 12, the vertices of the triangle shown in Fig. 12 are (-1.8,0.9), (1.8,0.9), and (0.0, -0.9), respectively. Depending on which side 6 to 8 of the triangle the pole (a, b) is near, the following three processes (1) to (3) are performed, and then a 'and b' are multiplied by multipliers 15, 16 Set to each.
  • the introduction of the IIR type recursive equalizer 1 having the denominator function as the transfer function (equalizer function) reduces the number of orders, With a small amount of hardware, it is possible to effectively compensate for a steep gain valley that TEQ 2 is not good at. Therefore, a good transmission amount, that is, a throughput can always be obtained regardless of the condition of the transmission path 300.
  • the coefficient of the recursive equalizer 1 the tap coefficient of the equalizer on the evening gate side is used at the time of training, so that the parameter can be obtained quickly and stably.
  • the F (frequency) characteristic of the first target equalizer 53 mainly depends on the characteristics of an analog filter (for example, filters 104 and 202 shown in FIG. 17) applied to the transmission side and the reception side.
  • an analog filter for example, filters 104 and 202 shown in FIG. 17
  • the second pull-in process may be omitted in some cases because an evening get equalizer with a fixed coefficient can be used substantially.
  • the arrangement position of the above-described recursive equalizer 1 can be considered to be after the TEQ 2 as shown in FIG. 13, for example.
  • the error e ⁇ for updating the tap coefficient of the TEQ 2 can be considered.
  • the same theory as the configuration shown in Fig. 4 (Fig. 6) ("The error used for updating the coefficients of the trans-persal equalizer is the output immediately after it)
  • the inverse recovery equalizer 1 ' is required.
  • the target equalizer is not separated from the first target equalizer 53 and the second target equalizer 54 as in the first embodiment, and the recursive equalization is performed as shown in FIG.
  • One (L + L '— first-order) channel target equalizer 57 whose order is increased by the order L', which is the denominator function of unit 1, performs the pull-in process, and at the stage when the coefficient training progresses, this target Factor the transfer function of equalizer 57 (Step S 1 in Figure 15), convert it to the product of a first-order or second-order z- 1 function, and select an appropriate one (L '— Extraction of linear function: As shown in step S 2) of FIG. 15 and FIG.
  • the coefficient updating block 5 performs processing to make the denominator function of the transfer function of the recursive equalizer 1 with the fixed coefficients on the real root side. For the remaining terms that are not set for relocation, function expansion is performed (step S4 in Fig. 15), and the tap coefficients are set so as to be the transfer function of the evening-equalizer 57 'with a predetermined order L or less. Try again.
  • the coefficient update block 5 in the second embodiment updates the tap coefficients of the evening equalizer 57 so that the difference between the output of the TEQ 2 and the output of the target equalizer 57 is minimized. It has a function as a second difference calculation update unit, and after the training period ends, the transfer function of the target equalizer 57 is factorized and the zero point is desirably present in the unit circle.
  • the tap coefficient of the recursive equalizer 1 is set so that the inverse function of the z function becomes the transfer function of the recursive equalizer 1 with fixed coefficients.
  • the transfer function has no denominator function and is only a numerator function, so its zero is allowed inside and outside the unit circle.
  • the denominator function of the transfer function of recursive equalizer 1 Since is not allowed outside the unit circle, we must check the zeros of the factorized function and select from the terms whose zeros are inside the unit circle. The check of the zero point is as described above in the first embodiment.
  • the frequency characteristics of the gain of the final target route should be as flat as possible in the transmission frequency band or as close to single-line characteristics as possible, and the gain should decrease monotonically with the frequency gap. Choose so that no large gain dips or peaks remain, or small gain dips or peaks.
  • the initial target equalizer 57 also includes a term corresponding to the recursive equalizer 1 as in this example, it is difficult to specify the initial value of the target equalizer 57. Therefore, in this case, it is desirable to provide an initial value to TEQ 2 and to initially fix the coefficient of TEQ 2 and make the coefficient of the target equalizer 57 variable.
  • the coefficient update block 5 updates the coefficient of the TEQ 2 and the target equalizer 5 7 ′ while fixing the coefficient of the recursive equalizer 1, and further compresses the error e. .
  • a fixed-coefficient recursive equalizer 1 updated at the evening-get-point side during training is introduced, thereby reducing the amount of hardware. Therefore, it is possible to sufficiently compensate for the steep gain valley of the received signal that TEQ 2 is not good at. Therefore, irrespective of the conditions of the transmission path 300, a good transmission amount, that is, a throughput can always be obtained, and all parameters of the equalizer system can be obtained at high speed and in a stable manner.
  • either the time domain or the frequency domain calculation method can be applied to update the coefficients of the equalizer system.
  • the FIR type is realized by introducing a fixed coefficient IIR type equalizer that uses the tap coefficients of the Eich device that is updated stably on the target route side during training.
  • a fixed coefficient IIR type equalizer that uses the tap coefficients of the Eich device that is updated stably on the target route side during training.
  • High throughput can be secured. Therefore, high-speed, high-quality services can be provided to subscribers in ADSL services and the like, and their usefulness is considered to be extremely high.

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Abstract

A coefficient update is performed by assuming a denominator function portion of a transfer function of an IIR (Infinite Impulse Response) type equalizer as a part of a numerator function portion of a transfer function of a FIR (Finite Impulse Response) type target equalizer (53, 54), so that a coefficient of the IIR type equalizer can be calculated stably and adaptively. In order to sufficiently compensate a steep gain valley, in addition to the FIR type equalizer (2), it is possible to use the IIR type equalizer which has not been able to be used conventionally.

Description

明 細 書 自動等化器及びその係数トレーニング方法 技術分野  Description Automatic equalizer and its coefficient training method
本発明は、 自動等化器及びその係数トレーニング方法に関し、 例えば、 加入者 線ペアケーブルを伝送路としてディスクリートマルチトーン (Discrete Multi-Tone) 変調方式により伝送される信号を受信する装置に用いて好適な、 自 動等化器及びその係数トレーニング方法に関する。 背景技術  The present invention relates to an automatic equalizer and a coefficient training method thereof, and is suitable for use, for example, in an apparatus for receiving a signal transmitted by a discrete multi-tone (Discrete Multi-Tone) modulation method using a subscriber line pair cable as a transmission path. The present invention relates to an automatic equalizer and its coefficient training method. Background art
近年、 インターネット等のマルチメディア型サービスが一般家庭を含めた社会 全体へと広く普及してきており、 このようなサービスを利用するための経済的で 信頼性の高いディジタル加入者線伝送技術の早期提供が強く求められている。 こ こで、 通信回線を新たに敷設するためには、 膨大なコストと時間が必要となるの で、 既存の通信回線を利用して高速にデータ通信を行なう方法が種々提案されて いる。  In recent years, multimedia services such as the Internet have become widespread throughout society, including ordinary households, and the early provision of economical and highly reliable digital subscriber line transmission technology for using such services. Is strongly required. Here, enormous costs and time are required to lay a new communication line, and various methods for performing high-speed data communication using existing communication lines have been proposed.
例えば、 既設の電話回線を高速データ通信回線として利用するディジタル加入 者線伝送技術として、近年、 X D S L (Digital Subscriber Line)が普及しつつある。 この x D S Lは、 既存の加入者線 (ペアケーブル) を利用した伝送方式であり、 また、変復調技術の一つでもある。 x D S Lには、大きく分けて、加入者宅 (以下、 加入者側という)から収容局 (以下、 局側という)への上り伝送速度と、 局側から加 入者側への下り伝送速度とが、 対称のものと非対称のものとに分類される。  For example, in recent years, XDSL (Digital Subscriber Line) has become widespread as a digital subscriber line transmission technology using an existing telephone line as a high-speed data communication line. This xDSL is a transmission method using the existing subscriber line (pair cable) and is one of the modulation and demodulation technologies. x DSL is broadly divided into the upstream transmission speed from the subscriber's home (hereafter, the subscriber side) to the accommodation station (hereafter, the office side) and the downstream transmission speed from the office side to the subscriber side. Are categorized as symmetric and asymmetric.
例えば、 対称型のものには、 上り 下りの伝送速度がともに 1 . 5〜2 . 0 M b p s (メガビットノ秒)程度の HD S L (High-bit-rate DSL)や、 1 6 0 k〜 2 . 0 M b p s程度の S D S L (Single-line DSL)等があり、 非対称型のものには、 最 近、 日本国内で急速に普及しつつある AD S L (Asymmetric DSL)がある。  For example, a symmetric type has a high-bit-rate DSL (HD SL) with an uplink and downlink transmission speed of about 1.5 to 2.0 Mbps (megabit-noise), and a 160 k to 2 There are SDSL (Single-line DSL) at about 0 Mbps, and the asymmetric type is AD SL (Asymmetric DSL), which has recently been rapidly spreading in Japan.
ここで、この AD S Lには、さらに、下り伝送速度が 8 M b p s程度の「G.dmt」 と 1 . 5 M b p s程度の「G.lite」 (簡易版 AD S Lとも呼ばれる) とがあるが、 どちらも D M T (Discrete Multi-Tone)変調と睜ばれる特有の変調方式が採用され ている。 Here, the AD SL further includes "G.dmt" having a downlink transmission speed of about 8 Mbps and "G.lite" (also called a simplified AD SL) having a downlink transmission rate of about 1.5 Mbps. , In both cases, a specific modulation scheme called DMT (Discrete Multi-Tone) modulation is adopted.
この DMT変調方式とは、 簡単にいうと、 伝送周波数帯域を約 4 kHz毎のサ ブキャリア ( 「G.lite」 の場合、 条件にもよるが、 下り方向は最大で 128本近 くのキャリア) に分割して、 それぞれのキャリアに変調を加える方式である。 こ の DMT変調方式では、 特定の周波数をもつノイズの影響により或るサブキヤリ ァが使用不能になっても、 他のサブキャリアでの通信が可能なので、 特定周波数 のノイズに強いという特徴がある。  This DMT modulation method is, in a nutshell, a subcarrier whose transmission frequency band is about every 4 kHz (in the case of “G.lite”, depending on the conditions, up to 128 carriers in the downlink direction). And modulates each carrier. This DMT modulation method has a feature that it is resistant to noise of a specific frequency because communication with another subcarrier is possible even if a certain subcarrier becomes unusable due to the influence of noise having a specific frequency.
なお、 DMT変調方式の原理の詳細については、例えば、文献「J. A. C.Bingham, Multicarrier Modulation For Data Transmission An Idea Whose Time has Come", IEEE Comm n. Mag., pp5-14, May, 1990J に記載されている。 以下、 このような DMT変調方式を採用するディジタル加入者線 (ADSL) 伝送システムの詳細について説明する。  The details of the principle of the DMT modulation method are described in, for example, the document "JACBingham, Multicarrier Modulation For Data Transmission An Idea Whose Time has Come", IEEE Co. n. Mag., Pp5-14, May, 1990J. I have. The details of the digital subscriber line (ADSL) transmission system employing such a DMT modulation method are described below.
(1) ADSL伝送システムの説明  (1) Description of ADSL transmission system
図 17に示すように、 ADSL伝送システムは、 その要部に着目すると、 例え ば、 加入者側 100に設置された AD S L装置 (AD S Lモデム) 1 10と局側 200に設置された AD S L装置 (AD S Lモデム) 210とが加入者線ペアケ 一ブル等の伝送路 (メタリック回線) 300を介して相互に接続されて構成され る。  As shown in Fig. 17, the ADSL transmission system focuses on its main components. For example, an AD SL device (AD SL modem) 110 installed on the subscriber side 100 and an AD SL device installed on the office 200 A device (AD SL modem) 210 is connected to each other via a transmission line (metallic line) 300 such as a subscriber line pair cable.
そして、この図 17に示すように、加入者側モデム 1 10には、送信系として、 DMT変調部 101,ディジタル送信フィルタ 102,ディジタル Zアナログ(D /A) 変換器 1 03, アナログ帯域通過フィルタ (B P F: Band Pass Filter) 104等がそなえられ、局側モデム 2 10には、受信系として、 トランス 201, アナログ低域通過フィルタ (LPF: Low Pass Filter) 202, AZD変換器 2 03,ディジタル B PF 204,時間域等化器(T E Q: Time-domain Equalizer) 205, 高速フーリェ変換器 (FFT: Fast Fourier Transformer) 206, 周 波数域等化器 (FEQ: Frequency-domain Equalizer) 207, 識別器 208等 がそなえられている。  As shown in FIG. 17, the subscriber-side modem 110 includes a DMT modulator 101, a digital transmission filter 102, a digital Z-analog (D / A) converter 103, an analog band-pass filter as a transmission system. (BPF: Band Pass Filter) 104, etc., and the station-side modem 210 has a transformer 201, analog low-pass filter (LPF) 202, AZD converter 203, digital B PF 204, Time-domain Equalizer (TEQ) 205, Fast Fourier Transformer (FFT) 206, Frequency-domain Equalizer (FEQ) 207, Classifier 208 Etc. are provided.
なお、 図 1 7では、 加入者側 100から局側 200への上り方向についての構 成しか示していないが、実際は、加入者側 100の AD S Lモデム 1 10 (以下、 加入者側モデム 110という)には、局側 200の ADSLモデム 210 (以下、 局側モデム 210という) における受信系と同等の機能をもった受信系が設けら れ、 局側モデム 210には、 加入者側モデム 110における送信系と同等の機能 をもった送信系が設けられており、 逆方向 (下り方向) の通信についても、 原理 的には、 以下に説明する上り方向の通信と同様にして行なわれるようになつてい る。 Note that in Fig. 17, the configuration for the upstream direction from the subscriber side 100 to the office side 200 is shown. However, in practice, the ADSL modem 110 of the subscriber 100 (hereinafter referred to as the subscriber modem 110) receives the signal from the ADSL modem 210 of the office 200 (hereinafter referred to as the office modem 210). A receiving system having the same function as the system is provided, and the station-side modem 210 is provided with a transmitting system having the same function as the transmitting system in the subscriber-side modem 110. In principle, the communication of (2) is performed in the same manner as the uplink communication described below.
ここで、 上記の加入者側モデム 110において、 DMT変調部 101は、 パー ソナルコンピュータ (PC) 等の加入者端末から送られた送信データについて D MT変調を施すためのもので、 このために、 例えば、 図 18に示すように、 シリ アル—パラレルバッファ (Serial to Parallel Buffer) 11 1, エンコーダ (Encoder) 112, 逆高速フ一リェ変換器 ( I F F T: Inverse Fast Fourier Transformer) 113をそなえて構成される。  Here, in the subscriber side modem 110 described above, the DMT modulation section 101 is for performing DMT modulation on transmission data transmitted from a subscriber terminal such as a personal computer (PC). For example, as shown in Fig. 18, it is composed of a serial-to-parallel buffer (Serial to Parallel Buffer) 111, an encoder (Encoder) 112, and an inverse fast Fourier transformer (IFFT) 113. You.
シリアル一パラレルバッファ 1 11は、 シリアルデータである送信データを 1 シンポル時間 (約 l/4kHz )分だけ格納するとともに、 格納したデ一夕をパラ レルデータに変換して出力するもので、 このとき、 所定の送信ビットマップ (図 示省略) に従って、 各搬送波 (サブキャリア) に対する伝送ビット数の割り当て (周波数帯の分割) が行なわれるようになつている。  The serial-to-parallel buffer 111 stores transmission data, which is serial data, for one symbol time (approximately l / 4 kHz), converts the stored data into parallel data, and outputs the parallel data. According to a predetermined transmission bitmap (not shown), the number of transmission bits is assigned to each carrier (subcarrier) (frequency band division).
例えば、 搬送波数を Nc (=1〜! 1; nは 2以上の整数) とすると、 図 18で は、 M 'fsbit/sの送信デ一タ (シリアル信号) が、 シリアル—パラレルバッフ ァ 111にて、 シンポルレ一ト fsの Mピットのパラレルデータに変換され、その うちの miビットが i番目 ( i = l〜n) の同じ搬送波で伝送されるべきビット グループとして割り当てられることを表わしている。  For example, assuming that the number of carriers is Nc (= 1 to! 1; n is an integer of 2 or more), in FIG. 18, the M'fsbit / s transmission data (serial signal) is represented by a serial-parallel buffer. Is converted into parallel data of M pits of the symbol fs, and the mi bits of the bits are assigned as the i-th (i = l to n) bit groups to be transmitted on the same carrier. .
これにより、 Mビットのデ一夕は全体としては Nc個の搬送波で同時に伝送さ れ、 受信側 (局側 200) では、 Ns個の時間データ (シンポル) を高速フーリ ェ変換(F FT: Fast Fourier mnsform) を用いて周波数域に変換し、 Nc個の 搬送波の振幅、 位相から合計 Mビットの情報を取り出すことになる。 なお、 各搬 送波は、 使用周波数帯域内で Afc (約 4 kHz) ずつ離れて配置される。 また、 I F FTと F FTの演算効率を考慮して、 上記のシンポル数 Nsは 2のべき乗値 である "64" とか "256"が用いられるのが一般的である。 As a result, the entire M-bit data is simultaneously transmitted on Nc carrier waves as a whole, and the receiving side (station side 200) transforms Ns time data (symbols) into a fast Fourier transform (FFT: Fast). Fourier mnsform) is used to convert to the frequency domain, and M bits of information are extracted from the amplitude and phase of the Nc carrier waves. Note that each carrier is placed at a distance of Afc (about 4 kHz) in the used frequency band. Considering the computation efficiency of IF FT and F FT, the number of symbols Ns is a power of 2 It is common to use "64" or "256".
次に、 上記のエンコーダ 112は、 このシリアル—パラレルバッファ 111か ら出力されるビット列を、 搬送波毎にその振幅, 位相を変えることで、 それぞれ 直交振幅変調 〔例えば、 QAM(Quadrature Amplitude Modulation)〕 するため の信号点デ一夕に変換するものであり、 I FFT113は、 このエンコーダ 11 2から出力される信号点データ (周波数領域のデータ) に対して逆高速フーリエ 変換を施すことにより、 その信号点データを Ns個の時間域信号列に変換するも のである。 これにより、 Mビットのデータは、 Ns個の標本値からなる時間域信 号 (シンポル) に変換されて送信されることになる。  Next, the encoder 112 performs quadrature amplitude modulation (for example, QAM (Quadrature Amplitude Modulation)) on the bit string output from the serial-parallel buffer 111 by changing the amplitude and phase for each carrier. The IFFT 113 performs inverse fast Fourier transform on the signal point data (data in the frequency domain) output from the encoder 112 to obtain the signal point. It converts the data into Ns time-domain signal sequences. As a result, the M-bit data is converted into a time-domain signal (symbol) consisting of Ns sample values and transmitted.
次に、 図 17において、 ディジタル送信フィルタ 102は、 上記 DMT変調部 101により得られたシンポル列 (ディジタルデータ) の不用成分を除去するた めのものであり、 0/八変換器103は、 このディジタル送信フィルタ 102の 出力をアナログデータに変換するためのものであり、 アナログ BP F 104は、 この DZA変換器 103の出力の不用成分を除去するためのものである。  Next, in FIG. 17, the digital transmission filter 102 is for removing unnecessary components of the symbol sequence (digital data) obtained by the DMT modulator 101, and the 0/8 converter 103 The output of the digital transmission filter 102 is converted to analog data, and the analog BPF 104 is for removing unnecessary components of the output of the DZA converter 103.
一方、 局側モデム 210において、 トランス 201は、 伝送路 300から受信 される信号の直流 (dc) 成分を遮断するためのものであり、 アナログ LPF2 02は、 高周波成分の信号及び雑音成分を遮断するものであり、 AZD変換器 2 03は、 このアナログ LPF 202を通過してきた受信信号 (アナログ信号) を ディジタル信号に変換するものである。  On the other hand, in the station-side modem 210, the transformer 201 is for cutting off the direct current (dc) component of the signal received from the transmission line 300, and the analog LPF 202 is for cutting off the high frequency component signal and the noise component. The AZD converter 203 converts the received signal (analog signal) passed through the analog LPF 202 into a digital signal.
さらに、 ディジタル BPF 204は、 上記 AZD変換器 203の出力について 所望帯域外の雑音成分などを除去するものであり、 TEQ205は、 入力信号に 対するシンポル間干渉(I S I : Inter Symbol Interference)が送信側 (加入者側 100) において付加されたサイクリックプリフィックス内に収まるように所定 の処理を施すためのものである。  Further, the digital BPF 204 removes noise components and the like outside the desired band from the output of the AZD converter 203, and the TEQ 205 generates inter-symbol interference (ISI: Inter Symbol Interference) with the input signal on the transmission side ( This is to perform predetermined processing so that it falls within the cyclic prefix added on the subscriber side 100).
ここで、 DMT変調方式の 1つのポイントは、 シンボル列の間にガード時間と いう期間を設けることにより、 受信側での復調が伝送路歪みの影響を受けにくい ようにすることができることである。 ガ一ド時間には、 その後に送るシンポルの 最後部のガード時間に相当する部分 (Lサンプル) を予め送る。 これがサイクリ  Here, one point of the DMT modulation method is that by providing a period called a guard time between symbol sequences, demodulation on the receiving side can be made less susceptible to transmission line distortion. At the guard time, a part (L sample) corresponding to the guard time at the end of the symbol sent after that is sent in advance. This is a cycle
(Cyclic Prefix) と呼ばれる。 一般に、 送信側から発せられた時間幅零のインパルスは、 伝送路の歪みのため に時間方向に有限の幅をもつレスポンスになり受信側に到達する。 それゆえ、 D MT変調方式においても、受信シンポルは送信シンポルに比べて時間幅が広がり、 振幅方向も送信されたものとは異なる形になる。 しかし、 時間方向の広がりがガ ード時間以下であって、 F FT区間の選択が正しければ、 F FT出力は送信側 I F FTの入力に対して複素数の固定係数を乗じたものと同一になり、 伝送路歪み の影響を受けないことになる。 (Cyclic Prefix). In general, an impulse with a time width of zero emitted from the transmission side becomes a response having a finite width in the time direction due to distortion of the transmission path and reaches the reception side. Therefore, even in the DMT modulation method, the reception symbol has a wider time width than the transmission symbol, and the amplitude direction is different from that of the transmitted symbol. However, if the spread in the time direction is less than the guard time and the selection of the FFT section is correct, the output of the FFT will be the same as the input of the transmitting IFFT multiplied by a complex fixed coefficient. However, transmission line distortion is not affected.
そこで、 上記 TEQ205は、 自己の係数 (タップ係数) が適応的に更新され ることにより、 送信側 I FFT113の出力から伝送路 300を経由して受信側 FFT206までのインパルスレスポンスの長さをガード時間以下にするように 動作するのである。 ただし、 実際には、 ガード時間を超える部分を完全に零にす ることは困難であり、 できるだけ小さくすることが行なわれる。  Therefore, the TEQ 205 adaptively updates its own coefficient (tap coefficient), and determines the length of the impulse response from the output of the transmitting IFFT 113 to the receiving FFT 206 via the transmission path 300 by the guard time. It works as follows. However, in practice, it is difficult to completely reduce the portion exceeding the guard time to zero, and the size is reduced as much as possible.
このように、 ADSL伝送システムに適用される TEQ205は、 DMT変調 方式以外の方式で用いられる等化器とはかなり目的を異にする (一般に、 振幅や 群遅延時間の周波数特性を平坦にすることを主目的にする)。なお、 TEQ205 自体は、 F I R (Finite Impulse Response)型のトランスバーサル等化器であり、 Q AM方式等で用いられる等化器と同様の構成である。 その係数更新は、 加算器 205 bにて得られる、 TEQ205の等化後出力とチャンネルターゲット等化 器 205 aの出力との差分に基づいて行なわれる (詳細については、 後述する)。 また、 ガード時間を長くすれば、 TEQ 205は不要又はその必要次数を減ら すことができるが、 ガード時間を長くすると、 伝送効率が悪くなるから、 ガード 時間はシンポル時間の 1Z16程度に選ばれるのが一般的である。  Thus, the purpose of the TEQ205 applied to the ADSL transmission system is quite different from that of the equalizer used in systems other than the DMT modulation system. (Generally, the frequency characteristics of the amplitude and group delay time are flattened. Is the main purpose). Note that the TEQ 205 itself is a FIR (Finite Impulse Response) type transversal equalizer, and has the same configuration as the equalizer used in the QAM system and the like. The coefficient is updated based on the difference between the equalized output of the TEQ 205 and the output of the channel target equalizer 205a, which is obtained by the adder 205b (the details will be described later). In addition, if the guard time is lengthened, the TEQ 205 is unnecessary or the required order can be reduced.However, if the guard time is lengthened, the transmission efficiency deteriorates, so the guard time is selected to be about 1Z16 of the symbol time. Is common.
次に、 FFT206は、 上記の TEQ205による等化後出力を高速フーリエ 変換によって周波数領域のデータに変換するためのものであり、 FEQ 207は、 この F FT 206の出力について、サブチャンネル(サブキャリアと等価である) 毎に複素定数 a+j b ( j = -D を掛け算することにより、サブチャンネル毎に F FT出力を周波数領域で等化して、 伝送路 300を通ることによって受信信号 が受けた振幅および位相への影響を周波数の異なるキヤリァ毎に補償するための ものであり、 識別器 208は、 この FEQ207の出力の実数部及び虚数部のそ れぞれをディジタルデ一夕 (ディジタル I, Q値) に変換 (以下、 識別処理とい う) するものである。 Next, the FFT 206 is for converting the output after equalization by the TEQ 205 into frequency-domain data by the fast Fourier transform, and the FEQ 207 is a sub-channel (sub-carrier and By multiplying by a complex constant a + jb (j = -D), the FFT output is equalized in the frequency domain for each subchannel, and the amplitude of the received signal received by passing through the transmission path 300 And the effect on the phase is compensated for each carrier having a different frequency. The discriminator 208 includes a real part and an imaginary part of the output of the FEQ 207. Each of them is converted to digital data (digital I and Q values) (hereinafter referred to as identification processing).
なお、得られたディジタルデータは、受信データとして後段の DMT復調部(図 示省略) へ出力され、 DMT復調部にて、 送信側 (加入者側 100) の DMT変 調部 101での変調処理とは逆の処理 (デコード, パラレル一シリアル変換等) を施されたのち、 直交復調される。  The obtained digital data is output to a subsequent DMT demodulation section (not shown) as received data, and the DMT demodulation section modulates the data at a DMT modulation section 101 on the transmission side (subscriber side 100). After performing the reverse processing (decoding, parallel-to-serial conversion, etc.), the signal is orthogonally demodulated.
また、 実際の装置では、 伝送路 300の長さによって受信レベルが大きく変動 するため、 AZD変換器 203の前に、 A/D変換器 203への入力レベルをほ ぼ一定にするための AG C (Automatic Gain Controlled)増幅器を揷入し、 AZ D変換器 203の後にも、 ディジタル信号処理での演算語長を減らす目的で、 デ イジタル AGCを挿入する楊合が多いが、 図 17においてはその図示を省略して いる。  Further, in an actual device, the reception level greatly varies depending on the length of the transmission path 300, and therefore, an AGC converter for making the input level to the A / D converter 203 almost constant before the AZD converter 203 is used. (Automatic Gain Controlled) In many cases, a digital AGC is inserted after the AZD converter 203 in order to reduce the operation word length in digital signal processing. Illustration is omitted.
以下、上述のごとく構成された ADSL伝送システムの動作について説明する。 まず、 加入者側モデム 110において、 DMT変調部 101に送信デ一夕が入 力されると、 シリアル—パラレルバッファ 111にその送信データが 1シンボル 時間(1/4 kHz)分だけ保持される。 保持されたデ一夕は予め決められたキヤ リア当たりの伝送ビット数毎に分割されて、 エンコーダ 112に出力される。 エンコーダ 112では、 入力されたビット列をそれぞれ直交振幅変調するため の信号点に変換して I F FT 113に出力する。 I F FT 113では、 このェン コーダ 112の出力に対して逆高速フーリエ変換を施すことで、 それぞれの信号 点についての直交振幅変調が施される。なお、この I FFT113の出力のうち、 所定サンプルがサイクリックプリフィックスとしてシンポルの先頭に付加される。  Hereinafter, the operation of the ADSL transmission system configured as described above will be described. First, in the subscriber-side modem 110, when transmission data is input to the DMT modulator 101, the transmission data is held in the serial-parallel buffer 111 for one symbol time (1/4 kHz). The stored data is divided for each predetermined number of transmission bits per carrier and output to the encoder 112. The encoder 112 converts each of the input bit strings into signal points for quadrature amplitude modulation, and outputs the signal points to the IFFT 113. In IFFT 113, quadrature amplitude modulation is performed on each signal point by performing an inverse fast Fourier transform on the output of encoder 112. A predetermined sample of the output of IFFT 113 is added to the head of the symbol as a cyclic prefix.
このようにサイクリックプリフィックスが付加された I F FT出力は、 デイジ タル送信フィルタ 102にて、 不要成分が除去されたのち、 DZA変換器 103 に入力され、 そこで、 所定サンプリング周波数 (例えば、 1. 104MHz) に よりサンプリングされてアナログ信号に変換され、 伝送路 300へ出力される。 一方、 局側モデム 210では、 伝送路 300を通じて受信されたアナログ信号 が、 トランス 201を経由してアナログ LPF 202に入力され、 アナログ LP F 202にて不要成分が除去されたのち、 AZD変換器 203にて、 ディジタル 信号に変換される。 The IF FT output to which the cyclic prefix has been added as described above is input to a DZA converter 103 after unnecessary components have been removed by a digital transmission filter 102, where a predetermined sampling frequency (for example, 1.104 MHz ), Is converted into an analog signal, and is output to the transmission path 300. On the other hand, in the station-side modem 210, the analog signal received through the transmission path 300 is input to the analog LPF 202 via the transformer 201, and after the unnecessary components are removed by the analog LPF 202, the AZD converter 203 At digital Converted to a signal.
得られたディジタル信号は、 次に、 ディジタル B P F 204にて、 さらに不要 成分が除去されたのち、 TEQ 205に入力され、 TEQ205にて、 受信路の ィンパルスレスポンスの長さが前記ガード時間 (サイクリックプリフィックス) 内に収まるように、 等化される。 等化後信号は、 FFT 206に入力され、 そこ で高速フーリエ変換が施されて、 周波数領域の信号点データに変換される。  The obtained digital signal is input to a TEQ 205 after further removing unnecessary components by a digital BPF 204, and the length of the impulse response of the reception path is determined by the TEQ 205 to the guard time (size). (The click prefix). The equalized signal is input to the FFT 206, where it is subjected to a fast Fourier transform to be converted into frequency-domain signal point data.
得られた信号点データは、 FEQ 207において、 伝送路 300を通ることに よって受けた振幅および位相への影響が周波数の異なるキヤリァ毎に補償された 後、 識別器 208にて、 識別処理が施されてディジタルデータ (ディジタル I, Q値) に変換されて受信データとして後段の処理部 (DMT復調部) へ出力され る。  The obtained signal point data is subjected to the FEQ 207 to compensate for the influence on the amplitude and phase caused by passing through the transmission path 300 for each carrier having a different frequency. The data is converted to digital data (digital I and Q values) and output to the subsequent processing unit (DMT demodulation unit) as received data.
(2) TEQ205のタツプ係数の求め方  (2) How to calculate tap coefficient of TEQ205
次に、 以下では、 上述した TEQ 205のタップ係数の求め方 (係数更新) に ついて説明する。 かかる係数の求め方については、 例えば、 米国特許 5,285,474 、 'Method for Equalizing A Multi-carrier Communication Systemノに 5ΐしレ。 即ち、 基本は、 「チャンネルターゲット」 と呼ばれる回路ブロック (等化器) を 想定することである。 かかる 「チャンネルターゲット」 は、 送信側 (加入者側 1 00) の I F FT出力から受信側 (局側 200) の F FT入力までの伝送系と等 価な伝達特性をもつトランスバーサルフィルタ構成の回路プロックであり、 その 夕ップ長をガード時間相当タツプ数である L以下に設定する。  Next, a method of obtaining the tap coefficient of TEQ 205 described above (coefficient update) will be described below. The method of obtaining such a coefficient is described in, for example, US Patent 5,285,474, 'Method for Equalizing A Multi-carrier Communication System'. In other words, the basic idea is to assume a circuit block (equalizer) called “channel target”. Such a “channel target” is a circuit with a transversal filter configuration having transfer characteristics equivalent to the transmission system from the IFFT output of the transmitting side (subscriber side 100) to the FFT input of the receiving side (station side 200). The block length is set to L or less, which is the number of taps equivalent to the guard time.
そして、 この 「チャンネルターゲット」 と、 送信側 I F FT出力から受信側 F FT入力までの伝送系 (以後、 実ルートと呼ぶ) とに同じ信号 (トレーニング信 号) を流して、 それらの出力の差がいつも零になるように、 チャンネルターゲッ ト 205 a (以下、 ターゲット等化器 205 aという) のトランスバーサルフィ ル夕と実ル一トに含まれる T E Q 205の係数を適応的に変えていくことが、 従 来行なわれている (図 19参照)。  Then, the same signal (training signal) is sent to this “channel target” and the transmission system (hereinafter referred to as “real route”) from the transmitting IFFT output to the receiving FFT input, and the difference between their outputs Adaptively change the coefficients of the TEQ 205 included in the transversal filter of the channel target 205a (hereafter referred to as the target equalizer 205a) and the actual route so that is always zero. Has been implemented (see Fig. 19).
具体的には、 トレーニング時間を設けて、 その間は送信側ランダムビットシー ケンサ 121により生成される送信側 (加入者側 100) の入力データ列と同一 デ一夕列信号を、 擬似ランダムビットシーケンサ 205 cによりターゲット等化 器 205 aの入力に印加して、 TEQ205の出力と夕一ゲット等化器 205 a の出力とを加算器 205 bにて比較 (加算) し、 その差分が小さくなるように夕 ーゲット等化器 205 aおよび TEQ205のタップ係数を適応的に変えてゆく のである。 More specifically, a training time is provided, during which the same data sequence as the input data sequence of the transmission side (subscriber side 100) generated by the transmission side random bit sequencer 121 is sent to the pseudo random bit sequencer 205. Target equalization by c The output of the TEQ 205 and the output of the evening get equalizer 205a are compared (added) by the adder 205b, and the evening get equalizer is set so that the difference becomes small. The tap coefficients of 205a and TEQ205 are changed adaptively.
この結果、 送信側 I FFT出力—受信側 FFT入力間のインパルスレスポンス は、 タ一ゲッ卜等化器 205 aのそれとほぼ同一となる。 夕ーゲットル一卜のィ ンパルスレスポンスは、ターゲット等化器 205 aのタップ係数そのものであり、 その時間幅はガード時間以下であるから、 送信側 I FFT出力一受信側 FFT入 力間のインパルスレスポンスの長さもガード時間以下となり、 DMT変調におけ る伝送路歪みの影響を抑圧することができる。  As a result, the impulse response between the transmitting-side IFFT output and the receiving-side FFT input is almost the same as that of the target equalizer 205a. The impulse response of the evening gate is the tap coefficient itself of the target equalizer 205a, and its time width is shorter than the guard time, so that the impulse response between the transmitter I FFT output and the receiver FFT input is The length is also equal to or less than the guard time, and the effect of transmission line distortion on DMT modulation can be suppressed.
ここで、 ターゲット等化器 205 aと TEQ 205の係数を求める場合に時間 軸で行なう方法と、周波数軸で行なう方法とがある。時間軸で求めていく方法は、 従来からよく知られているトランスバーサルフィルタの係数更新法を使用する。 即ち、 両出力の差とトランスバーサルフィルタ内の遅延素子の出力値を用いて夕 ップ係数を更新していく。 この時間域で解を求める場合に注意すべきは、 実ルー トでは伝送路 300が長くなると極めて大きな時間遅延を伴うことである。 そこ で、 かかる遅延に対しては、 図 19に示す遅延器 205 dを用いてサンプリング 周期の整数倍の遅延を補正することで対応する。  Here, when calculating the coefficients of the target equalizer 205a and the TEQ 205, there are a method that is performed on the time axis and a method that is performed on the frequency axis. The method of obtaining the values on the time axis uses a conventionally well-known transversal filter coefficient updating method. That is, the tap coefficient is updated using the difference between the two outputs and the output value of the delay element in the transversal filter. It should be noted that when a solution is obtained in this time range, an extremely long transmission path 300 involves an extremely large time delay in an actual route. Therefore, such a delay is dealt with by correcting a delay of an integral multiple of the sampling period using a delay unit 205d shown in FIG.
これに対し、 周波数軸で行なう方法は、 前記の米国特許 5,285,474に記載され ている。 図 19を流用して説明すると、 夕一ゲット等化器 205 aの後段に FF Tを接続しその出力と TEQ 205後段の F FT 206の出力とを比較してサブ チャンネル毎の差を計算し、 TEQ, ターゲット等化器 205 aそれぞれの係数 列を F FTで周波数軸に変換したものに周波数軸で修正を加える。かかる修正後、 I F FTで再び時間域軸に戻して時間窓でタップ係数数の調整を行なうという作 業を繰り返すことにより、 ターゲット等化器 205 aおよび TEQ 205の係数 を修正してゆく。  On the other hand, a method performed on the frequency axis is described in the aforementioned US Pat. No. 5,285,474. Explaining with reference to Fig. 19, an FFT is connected downstream of the evening get equalizer 205a, and its output is compared with the output of the FFT 206 downstream of the TEQ 205 to calculate the difference for each sub-channel. , TEQ, and target equalizer 205a. Correction is made on the frequency axis by transforming the coefficient sequence of each coefficient into the frequency axis by FFT. After such correction, the operation of returning to the time domain axis again by IFFT and adjusting the number of tap coefficients in the time window is repeated, thereby correcting the coefficients of the target equalizer 205a and the TEQ 205.
このように、 従来の DMT変調方式では、 時間軸、 周波数軸を問わず、 TEQ 205とターゲット等化器 205 aの両方の特性をできるだけ合わせるというも のである。 しかし、 実際には、 両者初期値零から引き込みを開始するのでは良い 解が得られない。 ターゲット特性は、 図 19から分かるように、 送信側 I FFT 出力から受信 F FT入力までの特性を擬似するものであるから、 伝送信号が通る 周波数帯域では、 それ以外の帯域に比べてゲインが十分高い、 即ち、 伝送信号が 通る帯域では通過域で周波数による信号の遅れの差が少なく、 不完全かもしれな いが遅延等化もなされていることが望ましい。 As described above, in the conventional DMT modulation method, the characteristics of both the TEQ 205 and the target equalizer 205a are matched as much as possible regardless of the time axis and the frequency axis. However, in practice, it is good to start pulling from both initial values of zero No solution is obtained. As can be seen from Fig. 19, the target characteristic simulates the characteristic from the transmitter IFFT output to the receiver FFT input, so that the gain is sufficient in the frequency band where the transmission signal passes, compared to the other bands. High, that is, in a band through which the transmission signal passes, the difference in signal delay due to frequency in the pass band is small, and it may be incomplete, but it is desirable that delay equalization be performed.
例えば、サブチヤンネル " 6 "に対応する約 26 k H zからサブチヤンネル " 3 2" に対応する約 138 kHzまでを加入者から電話局への周波数帯域として割 り当て、 最高通過域周波数 138 kHzの 4倍である 552 kHzで処理してい る DMT変調方式のターゲット等化器 205 aは、 概略 26〜 138 kH zを通 過域とする帯域通過フィル夕の伝達関数を初期値とすべきである。  For example, a range from about 26 kHz corresponding to sub-channel “6” to about 138 kHz corresponding to sub-channel “32” is assigned as a frequency band from the subscriber to the central office, and the maximum passband frequency is 138 kHz. The target equalizer 205a of the DMT modulation method processing at 552 kHz, which is four times the frequency of the bandpass filter, should have the initial value of the transfer function of the band-pass filter with a pass band of approximately 26 to 138 kHz. is there.
そして、 引き込み開始後、 しばらくは、 チャンネルターゲット 205の係数は 固定とし、 TEQ205の係数を可変とし、 TEQ係数が動かなくなったところ で、ターゲット等化器係数を可変にしてさらに差を小さくする手法を採ることで、 効率良く、 TEQ205の特性を得ることができる。  For a while after the start of the acquisition, the coefficient of the channel target 205 is fixed, the coefficient of the TEQ 205 is made variable, and when the TEQ coefficient stops moving, the coefficient of the target equalizer is made variable to further reduce the difference. By adopting it, the characteristics of TEQ205 can be obtained efficiently.
ここで、 DMT変調方式での良い特性とはどういう特性をいうのかというと、 TEQ205により実ルートのインパルスレスポンスのガード時間幅以上の部分 の振幅が極めて小さくなり、 FFT206の出力として得られる各サブチャンネ ル受信データが現シンポルの他サブチヤンネル信号から干渉を受けず、 前シンポ ルからの信号からも干渉を受けないことである (つまり、 I S Iができるだけ小 さいことである)。  Here, what is the good characteristic of the DMT modulation method? The TEQ205 makes the amplitude of the part of the impulse response of the actual route that is longer than the guard time width extremely small, and each subchannel obtained as the output of the FFT206 The received data is not interfered with by other subchannel signals of the current symbol, nor by the signal of the previous symbol (that is, ISI is as small as possible).
干渉 (I S I) のあるサブチャンネルでは、 S/Nが悪くなり、 そのサブチヤ ンネル経由での情報伝達量 (ビット数) が少なくなる。 サブチャンネル毎に配置 しうるビット数は上記のチャンネル間内干渉、 チャンネル間干渉以外に伝送路 3 00の雑音でも小さくなることは勿論である。 サブチャンネル毎に配置しうるビ ット数の合計がシンポル当りの総配置ビット数であり、 総配置ビット数にシンポ ル繰り返し周波数を乗じたものがスループットである。  In sub-channels with interference (ISI), the S / N deteriorates, and the amount of information (number of bits) transmitted through the sub-channels decreases. It goes without saying that the number of bits that can be arranged for each sub-channel is reduced by the noise of the transmission path 300 in addition to the above-mentioned intra-channel interference and inter-channel interference. The total number of bits that can be arranged for each subchannel is the total number of arranged bits per symbol, and the total number of arranged bits multiplied by the symbol repetition frequency is the throughput.
伝送路 300が単調な線路の場合は、 上記の方法で良い結果が得られる場合が 多い。 例えば、 線路長が 4 kmの場合には TEQ205の帯域内特性は線路の口 ス特性を等化する緩やかな高域通過フィルタ特性となり、 線路長が零のときには T E Q 2 0 5の帯域内特性は平坦なロス特性になる。 When the transmission line 300 is a monotonous line, good results are often obtained by the above method. For example, when the line length is 4 km, the in-band characteristics of the TEQ205 are gently high-pass filter characteristics that equalize the line characteristics of the line, and when the line length is zero, The in-band characteristics of the TEQ 205 are flat loss characteristics.
しかし、 伝送路 3 0 0に 「ブリッジタップ」 が存在する場合には、 良い結果が 得られない場合が多くなる。 「プリッジタップ」というのは線路の途中にある分岐 のための長さが不特定の終端開放の枝線のことである。 この 「ブリッジタップ」 が存在すると、 伝達特性は通過帯域の特定周波数帯でロスが大きくなることが多 く、 従来の T E Q 2 0 5では対応できず、 実ルートとターゲットルートの間に大 きな差が残り、その結果、実ルートのィンパルスレスポンスの長さを短くできず、 総配置ビット数が小さくなる場合がしばしば発生する。  However, when a “bridge tap” exists in the transmission path 300, good results are often not obtained. A "bridge tap" is an open-ended branch line of an unspecified length for a branch in the middle of a track. When this “bridge tap” is present, the transfer characteristic often has a large loss in a specific frequency band of the passband, and cannot be handled by the conventional TEQ 205, and there is a large loss between the actual route and the target route. The difference remains, and as a result, the length of the impulse response of the actual route cannot be shortened, and the total number of arranged bits often decreases.
これは、 T E Q 2 0 5が F I R型の適応等化器であるが故、 特定周波数を中心 とするロスの山を等化することに不向きであるからである。 即ち、 一般に、 トラ ンスパーサル型等化器等の F I R型の適応等化器では、 その伝達関数は分子関数 のみで分母関数が存在しないため、 急峻なロスの山は実現できるが、 急峻なゲイ ンの山の特性を実現する (つまり、 急峻なゲインの谷を補償する) のには不向き なのである。  This is because TEQ 205 is an FIR-type adaptive equalizer, and therefore is not suitable for equalizing a peak of a loss centered on a specific frequency. That is, in general, in an FIR type adaptive equalizer such as a transpersal type equalizer, the transfer function is only a numerator function and no denominator function is present, so a steep loss peak can be realized, but a steep gain It is not suitable for realizing the characteristics of the peak (that is, compensating for the steep valley of the gain).
このため、 ロスの山を等化 (補償) するには T E Q 2 0 5の特性に同じ形のゲ インの山をもたせる必要がある。 そこで、 T E Q 2 0 5には、 急峻なゲインの谷 を補償するのが得意である、 I I R (Infinite Impulse Response) 型の適応等化 器を適用するのが効率的である。 なお、 効率的とは、 少ない演算量で所要の特性 が得られるということである。 しかし、 I I R型の T E Qにすると、 分母閧数の 係数を伝送路 3 0 0、 「ブリッジタップ」に合わせて適応的に求めていく方法が無 い。  Therefore, in order to equalize (compensate) the loss peak, it is necessary to provide the same peak shape to the characteristic of TEQ205. Therefore, it is efficient to apply an IIR (Infinite Impulse Response) type adaptive equalizer, which is good at compensating for steep gain valleys, to TEQ205. Efficiency means that the required characteristics can be obtained with a small amount of computation. However, if the IIR type T EQ is used, there is no method for adaptively finding the coefficient of the denominator according to the transmission path 300 and the “bridge tap”.
例えば、 "ディジタルフィルタの設計" (武部幹著 東海出版 pp247〜pp249) に、 I I R型適応等化器について記載されているが、 「逐次近似の収束が一般に保 証されず、 また、 最小自乗平均誤差に必ずしも収束しない」 旨の記載があり、 I I R型の適応等化器は実際には使えないのである。  For example, “Design of Digital Filters” (Miki Takebe, Tokai Shuppan, pp247-pp249) describes an IIR adaptive equalizer, but “convergence of successive approximation is not generally guaranteed. It does not always converge to the error ", and the IIR type adaptive equalizer cannot be used in practice.
以上のように、 従来は、 T E Q 2 0 5として I I R型の等化器が使えない故、 「ブリッジタップ」 が存在する場合には、 十分に良いスループットが得られない 場合があるという課題があつた。  As described above, conventionally, IIR type equalizers cannot be used as TEQ 205.Therefore, when a “bridge tap” is present, there is a problem that sufficient good throughput may not be obtained. Was.
例えば、 加入者端に、 1 . 5 km離れて 5 0 0 mの 「ブリッジタップ」 が 2本 あるような線路において、 T E Q 2 0 5として従来のように 1 6〜 2 4タップの トランスバーサル型等化器を使用した場合の総配置ビッ卜数は、 8 3〜8 8ビッ トと少ない。 For example, two “bridge taps” 1.5 km apart and 500 m apart When a transversal equalizer with 16 to 24 taps is used as the TEQ 205 as in the past for a certain line, the total number of arranged bits is as small as 83 to 88 bits.
これに対し、 2次の分母関数を追加し、 その係数を最適化すると、 理論上は、 総配置ビット数を 1 3 4〜1 9 1ビットと 1 . 6〜2 . 3倍にできる。このため、 従来から 2次、 即ち、 3タップのリカーシブフィルタ (I I R型の等化器) を追 加すれば、 かなり特性を向上できるのではないかと考えられてきた。  On the other hand, if a second-order denominator function is added and its coefficient is optimized, the total number of arranged bits can theoretically be increased to 134-19.1 bits, or 1.6-2.3 times. For this reason, it has been thought that adding a second-order, or three-tap, recursive filter (IIR type equalizer) could significantly improve the characteristics.
しかし、前記のように、その係数を適応的に求める方法が明らかではないため、 例えば、 AG Cゲインに関連付けて段階的に係数を変えるエンファシス回路のよ うな単純な F (周波数)特性をもつもの以外には使われなかったのが現状である。 本発明は、 このような課題に鑑み創案されたもので、 I I R型の等化器の係数 を安定して適応的に求める手法を見出すことにより、 受信信号の急峻なゲインの 谷を十分に補償するのに F I R型の等化器に加えて、 I I R型の等化器を使用で きるようにすることを目的とする。  However, as described above, since the method of adaptively finding the coefficient is not clear, for example, one having a simple F (frequency) characteristic such as an emphasis circuit that changes the coefficient stepwise in association with the AGC gain At present, it was not used for any other purpose. The present invention has been made in view of such a problem, and has found a method for stably and adaptively obtaining coefficients of an IIR type equalizer, thereby sufficiently compensating for a steep gain valley of a received signal. The purpose is to be able to use IIR type equalizers in addition to FIR type equalizers.
発明の開示 Disclosure of the invention
上記の目的を達成するために、 本発明の自動等化器は、 所定の伝送路からの受 信信号について適応的に等化処理を施すものであって、 次の各部を有しているこ とを特徴としている。  In order to achieve the above object, an automatic equalizer of the present invention adaptively performs equalization processing on a signal received from a predetermined transmission path, and has the following units. It is characterized by.
(1)受信伝送路(送信 I F F T出力から F F T入力まで) のインパルス応答特性 の長さを短縮する等化器 (以下、 実ルート等化器という)  (1) An equalizer that shortens the length of the impulse response characteristics of the reception transmission line (from the transmission IFFT output to the FFT input) (hereinafter referred to as a real route equalizer)
(2)この実ル一ト等化器の係数トレーニング期間において、伝送路を含む実ルー ト等化器の出力までのルートのインパルス応答特性を擬似するターゲット等化器 (2) During the coefficient training period of this real route equalizer, the target equalizer that simulates the impulse response characteristics of the route up to the output of the real route equalizer including the transmission path
(3)係数トレーニング期間において、上記の実ルート等化器の出力と夕一ゲット 等化器の出力とに基づいて、 ターゲット等化器のタップ係数を更新し、 更新した タップ係数 (の一部) を用いて、 実ルート等化器の一部のタップ係数を設定する 係数更新制御部 (3) During the coefficient training period, the tap coefficients of the target equalizer are updated based on the output of the real root equalizer and the output of the evening get equalizer, and a part of the updated tap coefficients ( ) Is used to set some tap coefficients of the real root equalizer.
上述のごとく構成された本発明の自動等化器では、 係数トレーニング期間にお いて、 実ルート等化器の出力とターゲット等化器の出力とに基づいて、 夕ーゲッ ト等化器のタップ係数を更新し、 係数更新後のターゲット等化器の伝達関数を因 数分解した因数の一部の逆数を伝達関数とするように、 I I R型等化器のタップ 係数を設定することができる。 In the automatic equalizer of the present invention configured as described above, the tap coefficient of the evening equalizer is calculated based on the output of the real root equalizer and the output of the target equalizer during the coefficient training period. Is updated and the transfer function of the target equalizer The tap coefficients of the IIR type equalizer can be set so that some reciprocals of the factorized factor are used as transfer functions.
従って、 伝達関数として分母関数部を有する、 受信信号のゲインの急峻な谷を 補償するのに適した I I R型の等化器を使用することができ、 これにより、 伝送 路の条件に係わらず、常に良い伝送量、即ち、スループットを得ることができる。 図面の簡単な説明  Therefore, it is possible to use an IIR type equalizer having a denominator function part as a transfer function and suitable for compensating for a steep valley in the gain of a received signal, and thereby, regardless of the conditions of the transmission path, It is possible to always obtain a good transmission amount, that is, a throughput. BRIEF DESCRIPTION OF THE FIGURES
図 1及び図 2はいずれも本発明の原理を説明するためのプロック図である。 図 3は本発明の第 1実施形態に係る自動等化器の構成を示すプロック図である。 図 4〜図 7はいずれも図 3に示す係数更新ブロックの機能を説明するためのブ ロック図である。  1 and 2 are block diagrams illustrating the principle of the present invention. FIG. 3 is a block diagram showing a configuration of the automatic equalizer according to the first embodiment of the present invention. 4 to 7 are block diagrams for explaining the function of the coefficient update block shown in FIG.
図 8は図 4〜図 7に示す第 1チヤンネル夕一ゲット等化器の構成例を示すプロ ック図である。  FIG. 8 is a block diagram showing a configuration example of the first-channel first-get equalizer shown in FIGS.
図 9は図 4〜図 7に示す第 2チヤンネル夕ーゲット等化器の構成例を示すプロ ック図である。  FIG. 9 is a block diagram showing a configuration example of the second channel sunset-equalizer shown in FIGS.
図 1 0は図 3に示すトランスバーサル等化器 (T E Q) の構成例を示すブロッ ク図である。  FIG. 10 is a block diagram showing a configuration example of the transversal equalizer (T EQ) shown in FIG.
図 1 1は図 3に示すリカーシブ等化器の構成例を示すブロック図である。  FIG. 11 is a block diagram showing a configuration example of the recursive equalizer shown in FIG.
図 1 2は本実施形態に係る係数トレーニング時のポール判定を説明するための 図である。  FIG. 12 is a diagram for explaining pole determination during coefficient training according to the present embodiment.
図 1 3は図 4〜図 7に示すリカーシブ等化器を T E Qの後段に配置した場合の 構成を示すプロック図である。  FIG. 13 is a block diagram showing a configuration in a case where the recursive equalizer shown in FIGS. 4 to 7 is arranged after TEQU.
図 1 4は本発明の第 2実施形態に係る自動等化器の係数更新プロックの機能を 説明するためのブロック図である。  FIG. 14 is a block diagram for explaining a function of a coefficient update block of the automatic equalizer according to the second embodiment of the present invention.
図 1 5は第 2実施形態に係る係数更新ブロックの動作を説明するための図であ る。  FIG. 15 is a diagram for explaining the operation of the coefficient update block according to the second embodiment.
図 1 6は第 2実施形態に係る係数更新プロックの係数トレーニング終了後の機 能を説明するためのブロック図である。  FIG. 16 is a block diagram for explaining the function of the coefficient update block according to the second embodiment after the completion of coefficient training.
図 1 7は従来のディジタル加入者線 (AD S L ) 伝送システムの構成を示すブ ロック図である。 Figure 17 is a block diagram showing the configuration of a conventional digital subscriber line (AD SL) transmission system. It is a lock figure.
図 18は図 17に示す DMT変調部の構成を示すブロック図である。  FIG. 18 is a block diagram showing a configuration of the DMT modulator shown in FIG.
図 19は図 17に示す TEQの係数更新方法を説明するためのブロック図であ る。 発明を実施するための最良の形態  FIG. 19 is a block diagram for explaining the method of updating the TEQ coefficient shown in FIG. BEST MODE FOR CARRYING OUT THE INVENTION
以下、 図面を参照して、 本発明の実施形態について詳述する。  Hereinafter, embodiments of the present invention will be described in detail with reference to the drawings.
(A) 第 1実施形態の説明  (A) Description of the first embodiment
本発明のポイントは、 FFTに入力される受信系統 (実ルート) 側の I I R型 等化器の係数を、 ターゲット等化器を利用して適応的に求めることである。  The point of the present invention is to adaptively obtain the coefficients of the IIR type equalizer on the receiving system (real route) side input to the FFT using the target equalizer.
例えば、 DMT変調方式のように、 方式的に係数のトレーニング期間が設けら れていて、 加入者側から送られてくる信号が予め分かっている場合は、 擬似伝送 路パス (ターゲットルート) を設けて、 実際の伝送路パス (実ルート) からの信 号と比較できるから、 図 1に示すようなシステムが考えられる。  For example, if the training period of the coefficient is provided systematically as in the DMT modulation method and the signal transmitted from the subscriber side is known in advance, a pseudo transmission path (target route) is provided. Therefore, a system as shown in Fig. 1 can be considered because it can be compared with the signal from the actual transmission path (real route).
即ち、 伝送路の伝達関数を HI (z)
Figure imgf000014_0001
(z— 、 TEQの伝達関数を H
That is, the transfer function of the transmission path is HI (z)
Figure imgf000014_0001
(z—, the transfer function of TEQ is H
2 (Z (Z -1) /Hteqd (z— 、 ターゲット等化器の伝達関数を H 3 (Z)
Figure imgf000014_0002
(z "1) とおくと、 係数引き込み後、 ターゲットル一卜と実ル一 トとの誤差が零になると仮定して、 次式 (1) が成り立つ。
2 (Z (Z- 1 ) / Hteq d (z—, transfer function of target equalizer to H 3 (Z)
Figure imgf000014_0002
If (z " 1 ) is set, the following equation (1) holds, assuming that the error between the target route and the actual route becomes zero after the coefficient is pulled.
HI (z) · H'2 (z) -H3 (z) =0 … (1) ここで、 H2 (z) が零点をもたないとして、 両辺を H2 (z) で割ると、  HI (z) · H'2 (z) -H3 (z) = 0… (1) Here, if H2 (z) has no zero and both sides are divided by H2 (z),
HI (Z) -H3 (z) ZH2 (z) =0 … (2) が成り立つ。 これをブロック図に表すと図 2に示すようになる。 このことから、 丁£<3の伝達関数112 (z) =Htequ (z"1) /Hteqd (z"1) の分母関数 Hteqd (z-1) は、 ターゲット等ィ匕器側に移動すると分子関数となり、 ターゲット等化 器の伝達関数 H 3 (Z)
Figure imgf000014_0003
(z"1) の一部とみなすことができることが分 かる。
HI (Z) -H3 (z) ZH2 (z) = 0 ... (2) holds. This is shown in a block diagram in FIG. From this, the transfer function 112 (z) of H << 3 is equal to the denominator function Hteqd (z- 1 ) of Htequ (z " 1 ) / Hteqd (z" 1 ). When it moves, it becomes a molecular function, and the transfer function of the target equalizer H 3 (Z)
Figure imgf000014_0003
(z " 1 ).
ということは、 トレーニング時には夕一ゲット等化器のタップ係数の更新を行 ない、 更新後に、 ターゲット等化器の伝達関数を因数分解した時の因数成分の逆 数に相当する伝達関数が実ルート側に挿入されるように実ル一ト側の等化器の夕 ップ係数を設定すれば、 実ルート側の等化器の伝達関数がもつべき分母関数を適 応的に得ることが可能となる。 This means that during training, the tap coefficients of the one-time equalizer are updated, and after the update, the transfer function corresponding to the reciprocal of the factor component when the transfer function of the target equalizer is factorized is the real root. To be inserted into the real root of the equalizer evening By setting the tap coefficients, it is possible to appropriately obtain the denominator function that the transfer function of the equalizer on the real root side should have.
これを実現する自動等化器構成を図 3に示す。 この図 3に示す構成も例えば図 17に示す局側モデム 210に適用されるもので、 1はリカーシブ等化器、 2は 図 17に示す TEQ205と同等の TEQ、 3はウィンドウブロック、 4は図 1 7に示す F FT206と同等の F F T、 5は係数更新プロックをそれぞれ示す。 ここで、 リカ一シブ等化器 1は、 入力信号 (図 17の場合ならディジタル BP F 204の出力 (伝送路 300からの受信信号) を、 所定の分母関数 (例えば、 上記の Hteqd (z を伝達関数の分母とする、 I I R (Infinite Impulse Response) 型の等化器であり、 ここでは、 例えば図 11に示すような 3タツプの I I R型の等化器として構成される。 Figure 3 shows an automatic equalizer configuration that achieves this. The configuration shown in FIG. 3 is also applied to, for example, the office-side modem 210 shown in FIG. 17, where 1 is a recursive equalizer, 2 is a TEQ equivalent to the TEQ 205 shown in FIG. 17, 3 is a window block, and 4 is a diagram. FFT equivalent to FFT 206 shown in 17 and 5 show a coefficient update block, respectively. Here, the recoverable equalizer 1 converts the input signal (the output of the digital BP F 204 in the case of FIG. 17 (the received signal from the transmission path 300)) into a predetermined denominator function (for example, H teq d (This is an IIR (Infinite Impulse Response) type equalizer using z as the denominator of the transfer function. Here, for example, it is configured as a 3-tap IIR type equalizer as shown in FIG. 11.
即ち、 レジスタ (遅延素子) 13, 14に時系列に 1サンプルずつ保持された データ (加算器 1 1の出力) に対して、 それぞれ所定の係数 (タップ係数) が乗 算器 15, 16にて乗算され、 それらの総和が加算器 12にて求められ、 その結 果が加算器 1 1にて現在の入力信号と加算されることによって、 入力信号に対す る等化出力が得られるようになつている。  That is, for the data (output of the adder 11) held in the registers (delay elements) 13 and 14 one by one in time series, predetermined coefficients (tap coefficients) are respectively multiplied by the multipliers 15 and 16. The sum is obtained by the adder 12, and the result is added to the current input signal by the adder 11, so that an equalized output for the input signal can be obtained. ing.
一方、 TEQ (実ルート等化器) 2は、 入力信号 (リカーシブ等化器 1の出力) を、 所定の分子関数 (例えば、 上記の Htequ (z-1)) を伝達関数として等化する もので、 例えば図 10に示すような、 公知のトランスバーサル型等化器 (F I R 型の等化器) として構成され、 タップ数に応じた複数のレジスタ (遅延素子) 2 1に時系列に 1サンプルずつ入力信号が保持されることにより得られる時間軸上 の複数点のサンプルデータに対して、乗算器 22にてそれぞれタップ係数 C ( j ) (j =— N〜N) が乗算され、 それらの加算器 23による総和が入力信号 (リカ —シブ等化器 1の出力) に対する等化出力として得られるようになつている。 な お、 TEQ2のタップ数は、 ガード時間長に相当するサンプル数 Lが基準になる が、 それよりも多くても問題ない。 また、 それらの初期値は全て "0. 0" とし てよい。 On the other hand, the TEQ (real root equalizer) 2 equalizes the input signal (the output of the recursive equalizer 1) using a predetermined molecular function (for example, the above H tequ (z- 1 )) as a transfer function. It is configured as a well-known transversal type equalizer (FIR type equalizer) as shown in FIG. 10, for example, and a plurality of registers (delay elements) 21 corresponding to the number of taps are stored in a time-series manner. Multiplier 22 multiplies the sample data at a plurality of points on the time axis obtained by holding the input signal for each sample by a tap coefficient C (j) (j = —N to N). The sum by the adder 23 is obtained as an equalized output with respect to the input signal (the output of the reciprocating equalizer 1). The number of taps in TEQ2 is based on the number of samples L corresponding to the guard time length, but there is no problem if it is larger than that. Also, their initial values may all be "0.0".
さらに、 ウィンドウブロック 4は、 TEQ2の出力を所定時間遅延させて FF T 4に入力するためのものであり、 係数更新ブロック 5は、 リカーシブ等化器 1 及び T E Q 2の係数を T E Q 2の出力に基づいてそれぞれ適応的に更新するため のもので、 本実施形態では、 リカーシブ切替信号及び T E Q切替信号により、 リ カーシブ等化器 1を入力信号通過 (スルー) 状態にしたり、 更新後の係数値をリ 力一シブ等化器 1に反映 (設定) したりすることができるようになつている。 具体的に、 この係数更新ブロック 5は、 係数トレーニング期間において、 図 4 及び図 6に示すような機能部がソフトウェア処理によって実現される。 即ち、 こ れらの図 4及び図 6に示すように、 係数更新ブロック 5は、 擬似ランダムビット シーケンサ 5 1, 遅延ブロック 5 2 , 第 1チャンネルターゲット等化器 5 3 , 第 2チャンネルターゲット等化器 5 4 , 加算器 5 5及び逆第 2チャンネル夕一ゲッ ト関数ブロック 5 6を有して構成されている。 Further, the window block 4 is for delaying the output of TEQ2 by a predetermined time and inputting it to the FFT 4, and the coefficient update block 5 is for recursive equalizer 1 And the coefficient of TEQ 2 are adaptively updated based on the output of TEQ 2. In the present embodiment, the recursive switching signal and the TEQ switching signal pass the recursive equalizer 1 through the input signal (through). ) The state can be changed, and the updated coefficient value can be reflected (set) in the reactive equalizer 1. Specifically, in the coefficient update block 5, during the coefficient training period, the functional units shown in FIGS. 4 and 6 are realized by software processing. That is, as shown in FIGS. 4 and 6, the coefficient update block 5 includes a pseudo-random bit sequencer 51, a delay block 52, a first channel target equalizer 53, and a second channel target equalizer. It has an adder 54, an adder 55 and an inverse second channel evening get function block 56.
ここで、 擬似ランダムビットシーケンサ 5 1は、 図 1 9により前述した送信側 擬似ランダムビットシーケンサ 1 2 1と同じ擬似ランダム信号 (トレーニング信 号) を発生するものであり、 遅延ブロック 5 2は、 この擬似ランダムピットシー ケンサ 5 1から入力されるターゲットルート側の擬似ランダム信号の遅延時間を 調整して、 送信側擬似ランダムビットシーケンサ 1 2 1により伝送路 3 0 0を通 じて T E Q 2に入力される実ルート側の擬似ランダム信号の遅延時間と、 ターゲ ットルート側の擬似ランダム信号の遅延時間とを合わせるためのものである。 な お、 遅延ブロック 5 2は、 実ルート側 (T E Q 2の出力側) に設けてもよい (実 ルート側の擬似ランダム信号の遅延時間を調整するようにしてもよい)。  Here, the pseudo-random bit sequencer 51 generates the same pseudo-random signal (training signal) as the pseudo-random bit sequencer 12 1 on the transmitting side described above with reference to FIG. 19, and the delay block 52 The delay time of the pseudo-random signal on the target route side input from the pseudo-random pit sequencer 51 is adjusted, and the pseudo-random bit sequencer 121 on the transmitting side is input to the TEQ 2 via the transmission path 300 by the pseudo-random bit sequencer 121. This is to match the delay time of the pseudo-random signal on the real route side with the delay time of the pseudo-random signal on the target route side. Note that the delay block 52 may be provided on the real route side (the output side of TEQ 2) (the delay time of the pseudo random signal on the real route side may be adjusted).
また、第 1チャンネル夕一ゲット等化器(以下、第 1ターゲット等化器という) 5 3は、 遅延ブロック 5 2の出力 (擬似ランダム信号) に作用するもので、 例え ば図 8に示すように、 T E Q 2と同様のトランスバーサル型 (F I R型) の等化 器として構成される。  Also, the first channel evening get equalizer (hereinafter referred to as the first target equalizer) 53 acts on the output (pseudo-random signal) of the delay block 52, for example, as shown in FIG. In addition, it is configured as a transversal type (FIR type) equalizer similar to TEQ 2.
即ち、 本第 1ターゲット等化器 5 3では、 タップ数に応じた数のレジスタ (遅 延素子) 5 3 1に時系列に 1サンプルずつ入力信号がそれぞれ保持されることに より得られる時間軸上の複数点のサンプルデータに対して、 乗算器 5 3 2にて夕 ップ係数がそれぞれ乗算され、それらの加算器 5 3 3による総和が、入力信号(遅 延ブロック 5 2の出力) に対する出力として得られるようになつている。 なお、 第 1夕ーゲット等化器 5 3のタップ数は、 前述したガード時間長に相当するサン プル数 Lに近い値で、その初期値は前述の手法により予め設定した値を使用する。 また、 第 2チャンネルターゲット等化器 (従属等化器) 5 4は、 第 1夕一ゲッ ト等化器 5 3の等化後出力に作用するもので、 ここでは、 第 1ターゲット等化器 5 3と同様に F I R型の等化器として構成される。 That is, in the first target equalizer 53, the number of registers (delay elements) 531 corresponding to the number of taps, the time axis obtained by holding the input signals one by one in time series, is obtained. The sample data at the above plural points are multiplied by a multiplier coefficient in a multiplier 532, respectively, and the sum of those multipliers is added to the input signal (output of the delay block 52). It can be obtained as output. The number of taps in the first evening get equalizer 53 is equal to the number of taps corresponding to the guard time length described above. A value close to the number L of pulls, and the initial value uses a value preset by the above-described method. Also, the second channel target equalizer (subordinate equalizer) 54 acts on the post-equalization output of the first evening equalizer 53. Here, the first target equalizer 54 is used. It is configured as an FIR type equalizer in the same way as 53.
ただし、 本第 2チャンネルターゲット等化器 (以下、 第 2ターゲット等化器と いう) 5 4は、 上記の第 1ターゲット等化器 5 3に比して低次 (2〜4次; 3〜 5タップ) の等化器とし、 最終的に、 そのタップ係数を用いて実ルート側の分母 関数部となるリカーシブ等化器 1のタツプ係数を設定することを考慮して、 第 1 タップの係数値は " 1 . 0 " に固定し、 その他のタップ係数は " 0 . 0 " を初期 値とする。 3タップの場合の第 2ターゲット等化器 5 4の構成を図 9に示す。 この図 9に示すように、 第 2ターゲット等化器 5 4においても、 2つのレジス 夕 (遅延素子) 5 4 1に時系列に 1サンプルずつ入力信号がそれぞれ保持される ことにより得られる時間軸上の 3点のサンプルデータに対して乗算器 5 4 2, 5 4 3, 5 4 4にてタップ係数が乗算され、 それらの加算器 5 3 3による総和が、 入力信号 (第 1ターゲット等化器 5 3の出力) に対する出力として得られるよう になっている。  However, the second channel target equalizer (hereinafter, referred to as the second target equalizer) 54 has a lower order (2 to 4 order; 3 to 4) than the first target equalizer 53 described above. Considering that the tap coefficients of the recursive equalizer 1 that will be the denominator function part on the real route side are finally set using the tap coefficients, the tap coefficient of the first tap is considered. The numerical value is fixed to "1.0", and the other tap coefficients are initially set to "0.0". FIG. 9 shows the configuration of the second target equalizer 54 in the case of three taps. As shown in FIG. 9, also in the second target equalizer 54, the time axis obtained by holding the input signals one by one in time series in the two resistors (delay elements) 54 1 is obtained. Multipliers 542, 543, and 544 multiply the sample data of the above three points by tap coefficients, and the sum of those multipliers is added to the input signal (first target equalizer). (The output of the device 53).
さらに、 加算器 5 5は、 T E Q 2の出力と第 2ターゲット等化器 5 4の出力と との差分 (等化誤差信号 e ;以下、 単に 「誤差 e」 と略記する) を得るもので、 その誤差 eが最小となるように、 T E Q 2 , 第 1ターゲット等化器 5 3及び第 2 ターゲット等化器 5 4の係数がそれぞれ更新されるようになっている。  Further, the adder 55 obtains a difference between the output of the TEQ 2 and the output of the second target equalizer 54 (an equalization error signal e; hereinafter, simply abbreviated as “error e”). The coefficients of the TEQ 2, the first target equalizer 53, and the second target equalizer 54 are updated so that the error e is minimized.
ただし、 第 1夕一ゲット等化器 5 3の係数更新には、 加算器 5 5で得られた誤 差 eではなく、 第 2ターゲット等化器での変形を考慮した誤差 e ' を使わなけれ ばない。 これは、 「トランスバーサル型等化器の係数更新に使う誤差はその(直後 の) 出力端の誤差でなければならない」 ためである。 例えば、 第 2ターゲット等 化器 5 4の代わりに反転アンプ (ゲインが " 1 . 0 " であるが、 位相が 1 8 0度 回転) を用いた場合には、 反転アンプ出力でみると正のエラーが反転アンプ入力 では負になるから、 反転アンプの伝達関数 (=ー 1 . 0 ) を考慮しないと、 全く 逆符号の誤差で第 1ターゲット等化器 5 3の係数を更新することになる。  However, in updating the coefficients of the first evening equalizer 53, the error e 'taking into account the deformation in the second target equalizer must be used instead of the error e obtained by the adder 55. No. This is because the error used for updating the coefficients of the transversal equalizer must be the error of the (immediate) output terminal. For example, when an inverting amplifier (gain is “1.0”, but the phase is rotated by 180 degrees) is used instead of the second target equalizer 54, a positive Since the error becomes negative at the input of the inverting amplifier, the coefficient of the first target equalizer 53 is updated with a completely opposite sign error without considering the transfer function of the inverting amplifier (= -1. 0). .
今、誤差 eは、第 2夕ーゲット等化器 5 4の出力端で測定したものであるから、 第 2ターゲット等化器 5 4の係数更新にはそのまま使えるが、 第 1夕一ゲット等 化器 5 3の出力端と誤差 eの測定点の間には、 第 2ターゲット等化器 5 4が存在 するため、 第 1ターゲット等化器 5 3の出力点での誤差を e ' とし、 第 2夕一ゲ ット等化器 5 4の伝達関数を h 2とすると e ' · h 2 = eの関係が成り立つ。 したがって、 誤差 e ' = e /h 2を第 1ターゲット等化器 5 3の係数更新用の エラーとして使わなければならないことになる。 Now, since the error e is measured at the output end of the second evening equalizer 54, The second target equalizer 54 can be used as it is for updating the coefficient of the second target equalizer 54, but the second target equalizer 54 is located between the output end of the first evening equalizer 53 and the measurement point of the error e. Therefore, if the error at the output point of the first target equalizer 53 is e 'and the transfer function of the second evening equalizer 54 is h2, e'h2 = e Holds. Therefore, the error e ′ = e / h 2 must be used as an error for updating the coefficient of the first target equalizer 53.
この誤差 e ' を求めるのが、 逆第 2チャンネルターゲット関数ブロック (夕一 ゲット出力点誤差計算ブロック) 5 6であり、 第 2ターゲット等化器 5 4のもつ 等化器の伝達関数の逆数を伝達関数として有し、 その等化器関数により加算器 5 5で得られた誤差を時間域で等化して、 第 1ターゲット等化器 5 3の出力点での 誤差 e ' を求め、 これを第 1ターゲット等化器 5 3の係数更新に用いるようにな つているのである。  The error e ′ is obtained by the inverse second channel target function block (Yuichi get output point error calculation block) 5 6, and the inverse of the transfer function of the equalizer of the second target equalizer 54 is It has a transfer function, and the error obtained by the adder 55 is equalized in the time domain by the equalizer function to obtain an error e 'at the output point of the first target equalizer 53. It is used for updating the coefficient of the first target equalizer 53.
つまり、 本実施形態の係数更新ブロック 5は、 上記の加算器 5 5, 逆第 2チヤ ンネルターゲット関数ブロック 5 6および第 1夕一ゲット等化器 5 3, 第 2夕一 ゲット等化器 5 4及び T E Q 2の各係数を更新する第 1差分演算更新部で構成さ れているのである。  That is, the coefficient update block 5 of the present embodiment includes the adder 55, the inverse second channel target function block 56, the first evening get equalizer 53, and the second evening get equalizer 5. It is composed of a first difference calculation updating unit that updates each coefficient of 4 and TEQ 2.
以下、 上述のごとく構成された係数更新ブロック 5による係数更新処理 (係数 トレーニング方法) について説明する。  Hereinafter, a coefficient updating process (coefficient training method) by the coefficient updating block 5 configured as described above will be described.
まず、 図 4及び図 6に示すように、 実ルート (T E Q 2 ) とターゲットルート (遅延ブロック 5 2 ) に対して擬似ランダムビットシーケンサ 1 2 1, 5 1から 同じパターンの擬似ランダム信号が加えられる。 そして、 引き込み開始 (係数更 新処理開始) 直後は、 前記のリカーシブ切替信号によって、 リカーシブ等化器 1 はスルー設定とし、 第 1ターゲット等化器 5 3の係数は固定、 T E Q 2 , 第 2夕 一ゲット等化器 5 4の係数は可変として、 適応的に係数更新を行なう (第 1引き 込み処理)。  First, as shown in FIGS. 4 and 6, the pseudo-random bit sequencers 12 1 and 51 add the same pattern of pseudo-random signals to the real route (TEQ 2) and the target route (delay block 52). . Immediately after the start of the pull-in (the start of the coefficient update process), the recursive equalizer 1 is set to the through setting by the recursive switching signal, the coefficient of the first target equalizer 53 is fixed, and the TEQ 2 and the second evening are set. The coefficients of the one-get equalizer 54 are made variable, and the coefficients are updated adaptively (first acquisition process).
この際の係数更新の方法 (自動等化アルゴリズム) は、 加算器 5 5にて実ル一 ト出力 (T E Q 2の出力) からターゲットルート出力 (第 2ターゲット等化器 5 4の出力)を差し引いて得られる誤差 eを用いて、トランスバーサル型等化器 2, 5 4内の乗算器での係数値を適応的に最良値に近づけてゆく良く知られた方法で ある。 In this case, the coefficient updating method (automatic equalization algorithm) is that the target route output (the output of the second target equalizer 54) is subtracted from the actual route output (the output of TEQ 2) by the adder 55. Using the error e obtained by the above, a well-known method of adaptively approaching the coefficient value in the multiplier in the transversal type equalizers 2, 54 to the best value is used. is there.
例えば、 図 10に示す構成をもつ TEQ 2を例にすると、 LMS法と呼ばれる 方法では、係数 C (j )のリ回目の修正アルゴリズムは、次式(3)で表される。  For example, taking TEQ 2 having the configuration shown in FIG. 10 as an example, in a method called the LMS method, the second correction algorithm for the coefficient C (j) is expressed by the following equation (3).
C(;)(v+1) = C( )(v) - α 'e(v)… ( 3 ) C (;) (v + 1) = C () (v) -α 'e (v) … (3)
ただし、 この式 (3) において、 Xj Mは、 TEQ2の入力信号系列、 e (v)は 誤差を表す。 なお、 自動等化アルゴリズムの詳細については、 例えば文献 「適応 信号処理」 (昭晃堂 辻井重男編集 1995年 5月 15日初版発行 24頁)に 記載されており、 上記以外の本文献に記載された各種アルゴリズムを適用するこ とも可能である。 However, in this equation (3), Xj M represents the input signal sequence of TEQ2, and e (v) represents the error. The details of the automatic equalization algorithm are described in, for example, the document “Adaptive Signal Processing” (edited by Shokodo Shigeo Tsujii, first published on May 15, 1995, p. 24), and described in other documents other than the above. It is also possible to apply various algorithms.
さて、 以上の第 1引き込み処理の後、 係数更新ブロック 5は、 次に、 第 1ター ゲット等化器 53の係数も可変にして、 さらに誤差 eが小さくなるようにする第 2引き込み処理を行なう。なお、この際も、リカーシブ等化器 1及び TEQ 2は、 リカーシブ切替信号によって、 スルー設定に維持されている。  Now, after the above-described first pull-in process, the coefficient update block 5 next performs a second pull-in process in which the coefficient of the first target equalizer 53 is also made variable and the error e is further reduced. . At this time, the recursive equalizer 1 and TEQ 2 are maintained in the through setting by the recursive switching signal.
その後、 係数更新ブロック 5は、 図 5及び図 7に示すように、 リカーシブ等化 器 1の伝達関数が、 第 2ターゲット等化器 54の伝達関数の逆数となるように、 リカーシブ等化器 1のタップ係数を設定する。 すなわち、 図 9の乗算器 543, 544で乗算するタップ係数をそれぞれ図 11の乗算器 15, 16で乗算するタ ップ係数に設定する。 また、 ターゲットルート側の第 2ターゲット等化器 54は スルーとする処理を行ない、 リカーシブ切替信号によってリカーシブ等化器 1は 非スルー設定とする。  Thereafter, as shown in FIGS. 5 and 7, the coefficient update block 5 performs the recursive equalizer 1 so that the transfer function of the recursive equalizer 1 becomes the reciprocal of the transfer function of the second target equalizer 54. Set the tap coefficient of. That is, the tap coefficients to be multiplied by multipliers 543 and 544 in FIG. 9 are set to the tap coefficients to be multiplied by multipliers 15 and 16 in FIG. 11, respectively. In addition, the second target equalizer 54 on the target route side performs a process of setting to through, and the recursive equalizer 1 is set to non-through by a recursive switching signal.
さらにその後、 係数更新ブロック 5は、 TEQ 2と第 1ターゲット等化器 53 の係数を可変として、 さらに誤差 eが小さくなるように係数更新を行なう。 リカ 一シブ等化器 1の係数は固定しておく。  Further, thereafter, the coefficient update block 5 makes the coefficient of the TEQ 2 and the coefficient of the first target equalizer 53 variable, and updates the coefficient so that the error e is further reduced. The coefficient of the Rica equalizer 1 is fixed.
もちろん、 第 2ターゲット等化器 54のタップ係数の更新においては、 後にそ のタップ係数がリカーシブ等化器 1のタップ係数として設定され、 リカーシブ等 化器 1の伝達関数の極点が複素平面上の単位円 (以下、 単に 「単位円」 という) の外に配置されてしまうことがないように、 その更新値をチェックする必要があ る。ここで、タツプ係数設定後のリカーシブ等化器 1の伝達関数を、例えば H ( z ) =1/ (l + a z-^b z'1) のように、 0次係数を "1. 0" (乗算器 542 で乗算するタップ係数 =1. 0) とすると (図 1 1参照)、 ポール判定演算に考慮 すべき変数が、 aと b (乗算器 543, 544で乗算する夕ップ係数) だけにな るので、 第 2ターゲット等化器 54のタップ係数のチェックのための演算量が減 る。 Of course, in updating the tap coefficient of the second target equalizer 54, the tap coefficient is later set as the tap coefficient of the recursive equalizer 1, and the pole of the transfer function of the recursive equalizer 1 is set on the complex plane. It is necessary to check the updated value so that it is not placed outside the unit circle (hereinafter simply referred to as “unit circle”). Here, the transfer function of the recursive equalizer 1 after tap coefficient setting, as for example, H (z) = 1 / ( l + a z- ^ b z '1), the 0-order coefficient "1.0" (Multiplier 542 If the tap coefficient to be multiplied by is = 1.0 (see Fig. 11), the only variables to be considered in the pole determination operation are a and b (the evening coefficient to be multiplied by the multipliers 543 and 544). Therefore, the amount of calculation for checking the tap coefficient of the second target equalizer 54 is reduced.
即ち、 ポール位置を判定する場合、 上記関数を変形して、 z 2+a z +b = 0 を満たす解が、 複素平面上の単位円内に存在することを確認すればよい。 この条 件を考慮すると、 b^O. 9, b≥a— 0. 9, b≥— a— 0. 9が導出される (実際には、 0. 9ではなく 1. 0であるが、 実機動作の場合、 係数丸め等の影 響を考慮し、 余裕を少しもたせて 0. 9としている)。 これは、 図 12に示す三角 形領域内に aと bが存在すればよいことを示している。 That is, when determining the pole position, the above function may be modified to confirm that a solution satisfying z 2 + az + b = 0 exists in a unit circle on the complex plane. Considering this condition, b ^ O. 9, b≥a—0.9, b≥—a—0.9 is derived (actually, it is 1.0 instead of 0.9, In the case of actual operation, the margin is set to 0.9 in consideration of the effects of coefficient rounding, etc.). This indicates that it suffices that a and b exist within the triangular area shown in FIG.
係数更新プロック 5は、 この条件を満足するようにソフトウェアにより係数更 新を制御する。 即ち、 係数更新ブロック 5は、 ポールが図 12に示す三角形領域 外に存在することとなる場合は、 乗算器 542, 543におけるタップ係数を乗 算器 15, 1 6にそのまま設定し、 ポールが図 12に示す三角形領域外に存在す ることとなる場合、 図 1 2に示す三角形の頂点が、 それぞれ、 (-1.8,0.9), (1.8,0.9), (0.0,-0.9) であるから、 ポール (a, b) が当該三角形のどの辺 6〜 8に近いかによつて、下記の 3通り (1)〜(3) の処理を行なってから、 a' , b ' を乗算器 15, 16にそれぞれ設定する。  The coefficient update block 5 controls the coefficient update by software so as to satisfy this condition. That is, the coefficient update block 5 sets the tap coefficients in the multipliers 542 and 543 to the multipliers 15 and 16 as they are when the pole exists outside the triangular area shown in FIG. If they exist outside the triangle area shown in Fig. 12, the vertices of the triangle shown in Fig. 12 are (-1.8,0.9), (1.8,0.9), and (0.0, -0.9), respectively. Depending on which side 6 to 8 of the triangle the pole (a, b) is near, the following three processes (1) to (3) are performed, and then a 'and b' are multiplied by multipliers 15, 16 Set to each.
(1) 上の辺 6に近い場合  (1) When it is close to the upper side 6
b≥0.9, a≥1.8のとき、 a' =1.8, b' =0.9  When b≥0.9, a≥1.8, a '= 1.8, b' = 0.9
b≥0.9, 1.8>a≥-l,8のとき、 a' =a, b' =0.9  When b≥0.9, 1.8> a≥-l, 8, a '= a, b' = 0.9
b≥0.9, -1.8>aのとき、 a' =-1.8, b' =0.9  When b≥0.9, -1.8> a, a '= -1.8, b' = 0.9
(2) 右の辺 7に近い場合  (2) When close to the right side 7
b<0.9, a>0, a+b≥2.7のとき、 a' =1.8, b' =0.9  When b <0.9, a> 0, a + b≥2.7, a '= 1.8, b' = 0.9
b<0.9, a>0, 2.7>a≥-0.9のとき、 a' =(a+b+0.9)/2, b' =(a+b-0.9)/2 b<0.9, a>0, -0.9〉a+bのとき、 a' =0.0, b' =-0.9  When b <0.9, a> 0, 2.7> a≥-0.9, a '= (a + b + 0.9) / 2, b' = (a + b-0.9) / 2 b <0.9, a> 0, -0.9> a + b, a '= 0.0, b' = -0.9
(3) 左の辺 8に近い場合  (3) Close to the left side 8
b<0.9, aく 0, -a+b≥2.7のとき、 a' ="1.8, b' =0.9  When b <0.9, a <0, -a + b≥2.7, a '= 1.8, b' = 0.9
b<0.9, a<0, 2.7>-a+b≥-0.9のとき、 a' =-(-a+b+0.9)/2, b' =(-a+b-0.9)/2 b<0.9, -0.9>-a+bのとき、 a' =0.0, b' =-0.9 When b <0.9, a <0, 2.7> -a + b≥-0.9, a '=-(-a + b + 0.9) / 2, b' = (-a + b-0.9) / 2 When b <0.9, -0.9> -a + b, a '= 0.0, b' = -0.9
以上のように、 本実施形態によれば、 TEQ2に加えて、 分母関数を伝達関数 (等化器関数) として有する I I R型のリカーシブ等化器 1を導入したことによ り、 少ない次数、 つまり、 少ないハードウェア量で、 TEQ 2が不得意とする急 峻なゲインの谷の補償を効果的に行なうことができる。 したがって、 伝送路 30 0の条件に係わらず、常に良い伝送量、即ち、スループットを得ることができる。 そして、 リカーシブ等化器 1の係数については、 トレーニング時において、 夕 ーゲットル一ト側の等化器のタップ係数を利用するため、 パラメ一夕を高速且つ 安定に得ることができる。  As described above, according to the present embodiment, in addition to TEQ2, the introduction of the IIR type recursive equalizer 1 having the denominator function as the transfer function (equalizer function) reduces the number of orders, With a small amount of hardware, it is possible to effectively compensate for a steep gain valley that TEQ 2 is not good at. Therefore, a good transmission amount, that is, a throughput can always be obtained regardless of the condition of the transmission path 300. As for the coefficient of the recursive equalizer 1, the tap coefficient of the equalizer on the evening gate side is used at the time of training, so that the parameter can be obtained quickly and stably.
なお、 上記のように 3段階に分けて引き込みを行なうことが不可欠というわけ ではない。 例えば、 図 7に示すリカーシブ等化器 1導入後の TEQ2および第 1 ターゲット等化器の係数更新による特性向上は小さいから省略することもできる。 また、 第 1ターゲット等化器 53の F (周波数) 特性は、 主に、 送信側、 受信側 に適用されるアナログフィルタ (例えば図 17に示すフィル夕 104, 202) の特性に左右され、 ケーブル長に応じた伝送路の周波数特性を、 主に、 TEQ2 で補償する場合には、 実質的に、 固定係数の夕一ゲット等化器を使用できるから 第 2引き込み処理は省略できる場合もある。  It should be noted that it is not indispensable to perform the attraction in three stages as described above. For example, since the improvement of the characteristics of the TEQ2 and the first target equalizer after the recursive equalizer 1 shown in FIG. The F (frequency) characteristic of the first target equalizer 53 mainly depends on the characteristics of an analog filter (for example, filters 104 and 202 shown in FIG. 17) applied to the transmission side and the reception side. When the frequency characteristics of the transmission path according to the length are mainly compensated by TEQ2, the second pull-in process may be omitted in some cases because an evening get equalizer with a fixed coefficient can be used substantially.
さらに、 上述したリカーシブ等化器 1の配置位置は、 例えば図 13に示すよう に、 TEQ 2の後段にすることも考えられるが、 この場合、 TEQ2のタップ係 数更新のための誤差 e〃 の計算が必要となり、誤差 e〃 を求めるために、図 4 (図 6)に示す構成と同様の理論(「トランスパーサル型等化器の係数更新に使う誤差 はその (直後の) 出力端の誤差でなければならない」) により、 逆リカ一シブ等化 器 1 ' が必要となる。  Further, the arrangement position of the above-described recursive equalizer 1 can be considered to be after the TEQ 2 as shown in FIG. 13, for example. In this case, the error e〃 for updating the tap coefficient of the TEQ 2 can be considered. In order to obtain the error e〃, the same theory as the configuration shown in Fig. 4 (Fig. 6) ("The error used for updating the coefficients of the trans-persal equalizer is the output immediately after it) The inverse recovery equalizer 1 'is required.
(B) 第 2実施形態の説明  (B) Description of the second embodiment
本第 2実施形態では、 ターゲット等化器を第 1実施形態のように第 1ターゲッ ト等化器 53及び第 2ターゲット等化器 54と分けないで、図 14に示すように、 リカーシブ等化器 1の分母関数となる次数分 L' だけ次数を上げた 1つの (L + L' — 1次) チャンネルターゲット等化器 57として引き込み処理を行ない、 係 数トレーニングが進んだ段階で、 このターゲット等化器 57の伝達関数を因数分 解して (図 1 5のステップ S 1 )、 1次または 2次の z—1関数の積に変換し、 そ の中で適当なものを選択して (L ' — 1次関数の抽出:図 1 5のステップ S 2 )、 図 1 6に示すように、 実ルート側の係数固定のリカーシブ等化器 1の伝達関数の 分母関数となるようにする処理を係数更新ブロック 5が行なう。 なお、 移設設定 しない残りの項については関数展開を行ない(図 1 5のステップ S 4 )、所定次数 L以下の夕ーゲット等化器 5 7 ' の伝達関数となるようにそのタップ係数を設定 'しなおす。 In the second embodiment, the target equalizer is not separated from the first target equalizer 53 and the second target equalizer 54 as in the first embodiment, and the recursive equalization is performed as shown in FIG. One (L + L '— first-order) channel target equalizer 57 whose order is increased by the order L', which is the denominator function of unit 1, performs the pull-in process, and at the stage when the coefficient training progresses, this target Factor the transfer function of equalizer 57 (Step S 1 in Figure 15), convert it to the product of a first-order or second-order z- 1 function, and select an appropriate one (L '— Extraction of linear function: As shown in step S 2) of FIG. 15 and FIG. 16, the coefficient updating block 5 performs processing to make the denominator function of the transfer function of the recursive equalizer 1 with the fixed coefficients on the real root side. For the remaining terms that are not set for relocation, function expansion is performed (step S4 in Fig. 15), and the tap coefficients are set so as to be the transfer function of the evening-equalizer 57 'with a predetermined order L or less. Try again.
つまり、 本第 2実施形態における係数更新ブロック 5は、 T E Q 2の出力と該 ターゲット等化器 5 7の出力との差分が最小となるように夕ーゲット等化器 5 7 のタップ係数を更新する第 2差分演算更新部としての機能を有しており、 トレー ニング期間の終了後、 そのターゲット等化器 5 7の伝達関数を因数分解して、 そ の零点が単位円の中に存在する所望の z関数の逆関数が係数固定のリカーシブ等 化器 1の伝達関数となるようにリカーシブ等化器 1のタツプ係数を設定するよう になっているのである。 That is, the coefficient update block 5 in the second embodiment updates the tap coefficients of the evening equalizer 57 so that the difference between the output of the TEQ 2 and the output of the target equalizer 57 is minimized. It has a function as a second difference calculation update unit, and after the training period ends, the transfer function of the target equalizer 57 is factorized and the zero point is desirably present in the unit circle. The tap coefficient of the recursive equalizer 1 is set so that the inverse function of the z function becomes the transfer function of the recursive equalizer 1 with fixed coefficients.
ただし、 z = ej ' 2 7t i/ f sであり、 f sはサンプリング周波数、 fは周波数をそ れぞれ表し、 j = -1である。 Here, z = ej ' 27t i / fs , fs represents the sampling frequency, f represents the frequency, and j = -1.
ここで、 適当なものを選択する基準は次のようにすべきである。  Here, the criteria for selecting an appropriate one should be as follows.
①夕ーゲット等化器の場合は伝達関数は分母関数がなく、 分子関数のみである からその零点は単位円の内部でも外でも許されるが、 リカーシブ等化器 1の伝達 関数の分母関数の場合は単位円の外は許されないから、 因数分解した関数の零点 をチェックして、零点が単位円の中にある項から選択しなければならない。なお、 零点のチェックは、 第 1実施形態にて前述したとおりである。  (1) In the case of the evening get equalizer, the transfer function has no denominator function and is only a numerator function, so its zero is allowed inside and outside the unit circle. However, in the case of the denominator function of the transfer function of recursive equalizer 1, Since is not allowed outside the unit circle, we must check the zeros of the factorized function and select from the terms whose zeros are inside the unit circle. The check of the zero point is as described above in the first embodiment.
②最終的なターゲットルートのゲインの周波数特性は伝送周波数帯域でできる だけ平坦または単線路特性に近い、 周波数ァップとともに単調にゲインが下がる 特性であることが望ましいから、 ターゲットル一トの周波数特性に大きなゲイン のくぼみ又は山が残らないように、 もしくは、 ゲインのくぼみ又は山が小さくな るように選択する。  ② The frequency characteristics of the gain of the final target route should be as flat as possible in the transmission frequency band or as close to single-line characteristics as possible, and the gain should decrease monotonically with the frequency gap. Choose so that no large gain dips or peaks remain, or small gain dips or peaks.
なお、 本例のように当初のターゲット等化器 5 7にリカーシブ等化器 1で対応 する項も含ませる場合には、 ターゲット等化器 5 7の初期値を特定することは難 しいから、 この場合は、 T E Q 2に初期値を与え、 引き込み当初は T E Q 2の係 数は固定で、 ターゲット等化器 5 7の係数を可変にするのが望ましい。 また、 図 1 6においては、係数更新ブロック 5は、リカーシブ等化器 1の係数を固定して、 T E Q 2 , ターゲット等化器 5 7 ' の係数更新を行ない、 誤差 eの更なる圧縮を 行なう。 If the initial target equalizer 57 also includes a term corresponding to the recursive equalizer 1 as in this example, it is difficult to specify the initial value of the target equalizer 57. Therefore, in this case, it is desirable to provide an initial value to TEQ 2 and to initially fix the coefficient of TEQ 2 and make the coefficient of the target equalizer 57 variable. In FIG. 16, the coefficient update block 5 updates the coefficient of the TEQ 2 and the target equalizer 5 7 ′ while fixing the coefficient of the recursive equalizer 1, and further compresses the error e. .
本第 2実施形態においても、 第 1実施形態と同様に、 トレーニング時に夕一ゲ ットル一ト側で係数更新した係数固定のリカーシブ等化器 1を導入したことによ り、 少ないハードウェア量で、 T E Q 2が不得意とする受信信号の急峻なゲイン の谷を十分に補償することができる。 したがって、 伝送路 3 0 0の条件に係わら ず、 常に良い伝送量、 即ち、 スループットを得ることができるとともに、 等化器 系の全てのパラメータを高速且つ安定に得ることができる。  In the second embodiment, as in the first embodiment, a fixed-coefficient recursive equalizer 1 updated at the evening-get-point side during training is introduced, thereby reducing the amount of hardware. Therefore, it is possible to sufficiently compensate for the steep gain valley of the received signal that TEQ 2 is not good at. Therefore, irrespective of the conditions of the transmission path 300, a good transmission amount, that is, a throughput can always be obtained, and all parameters of the equalizer system can be obtained at high speed and in a stable manner.
なお、 本発明は、 上述した実施形態に限定されず、 本発明の趣旨を逸脱しない 範囲で種々変形して実施することができる。  The present invention is not limited to the above-described embodiment, and can be implemented with various modifications without departing from the spirit of the present invention.
例えば、 等化器系の係数更新には、 時間域または周波数域計算法のいずれを適 用することもできる。 産業上の利用可能性  For example, either the time domain or the frequency domain calculation method can be applied to update the coefficients of the equalizer system. Industrial applicability
以上のように、 本発明によれば、 トレーニング時にターゲットルート側で安定 して係数更新した等ィヒ器のタップ係数を利用する係数固定の I I R型の等化器を 導入したことにより、 F I R型の等化器のみでは受信信号の急峻なゲインの谷を 十分に補償できないという課題を解決することができ、 加入者ペアケーブル等の 伝送路にプリッジタップが存在する等の条件に関わらず、 良好なスループットを 確保することができる。 したがって、 AD S Lサービス等において高速, 高品質 なサービスを加入者に提供することができ、 その有用性は極めて高いものと考え られる。  As described above, according to the present invention, the FIR type is realized by introducing a fixed coefficient IIR type equalizer that uses the tap coefficients of the Eich device that is updated stably on the target route side during training. Can eliminate the problem that the steep gain valley of the received signal cannot be sufficiently compensated for using the equalizer alone, and it is good regardless of conditions such as the presence of a bridge tap in the transmission path such as a paired subscriber cable. High throughput can be secured. Therefore, high-speed, high-quality services can be provided to subscribers in ADSL services and the like, and their usefulness is considered to be extremely high.

Claims

請 求 の 範 囲 The scope of the claims
1 . 所定の伝送路からの受信信号について適応的に等化処理を施す自動等化器 であって、 1. An automatic equalizer that adaptively performs equalization processing on a received signal from a predetermined transmission path,
伝送路からの受信信号に作用する等化手段と、  An equalizing means that operates on a reception signal from the transmission path;
該伝送路を含む該等化手段の出力までのィンパルス応答特性を擬似する夕ーゲ ット等化器と、  An evening equalizer that simulates an impulse response characteristic up to an output of the equalizing means including the transmission path;
該伝送路からのトレーニング信号に基づく該等化手段の出力と、 該ターゲット 等化器の出力とに基づいて、 該等化手段及び該タ一ゲット等化器のタップ係数を 更新し、 更新後のターゲット等化器のタップ係数を用いて該等化手段の一部の夕 ップ係数を設定する係数更新制御部とをそなえたことを特徴とする、自動等化器。  The tap coefficients of the equalizer and the target equalizer are updated based on the output of the equalizer based on the training signal from the transmission path and the output of the target equalizer. An automatic equalizer, comprising: a coefficient update control unit that sets a tap coefficient of a part of the equalizing means using a tap coefficient of the target equalizer.
2 · 該等化手段が、 2 · The equalization means
該受信信号を時間域で等化する、 異なる 2以上の等化器を含み、  Including two or more different equalizers for equalizing the received signal in a time domain,
該係数更新制御部が、 前記設定を行なうのは、 上記異なる等化器のうち前段の 等化器であることを特徴とする、 請求の範囲第 1項に記載の自動等化器。  2. The automatic equalizer according to claim 1, wherein the coefficient update control unit performs the setting by a pre-stage equalizer among the different equalizers.
3 . 該係数更新制御部は、 前記設定後における前記設定を行なった対象外の等 化器のタップ係数の更新の際には、 前記設定を行なった対象の等化器のタップ係 数は固定とすることを特徴とする、 請求の範囲第 2項に記載の自動等化器。 3. When updating the tap coefficients of the equalizers that are not subjected to the setting after the setting, the coefficient update control unit sets the tap coefficient of the equalizer to be set to be fixed. 3. The automatic equalizer according to claim 2, wherein:
4 . 該等化手段が、 4. The equalizing means is
該受信信号を時間域で等化する I I R (Infinite Impulse Response) 型等化器 と、 該 I I R型等化器の出力を時間域で等化する F I R (Finite Impulse Response) 型等化器とをそなえて構成されるとともに、  An IIR (Infinite Impulse Response) type equalizer for equalizing the received signal in the time domain, and a FIR (Finite Impulse Response) type equalizer for equalizing the output of the IIR type equalizer in the time domain. As well as
該係数更新制御部の行なう前記設定は、  The setting performed by the coefficient update control unit includes:
前記更新後のターゲット等化器の伝達関数を因数分解した因数の逆数に相当す る伝達関数をもつように該 I I R型等ィ匕器のタップ係数を設定することにより行 なわれることを特徴とする、 請求の範囲第 1項に記載の自動等化器。 This is performed by setting tap coefficients of the IIR type equalizer so as to have a transfer function corresponding to the reciprocal of a factor obtained by factorizing the transfer function of the target equalizer after the update. The automatic equalizer according to claim 1.
5 . 該係数更新制御部が、 5. The coefficient update control unit
前記更新後、 前記ターゲット等化器の伝達関数が前記因数を除算したものとな るように該夕ーゲット等化器を制御することを特徴とする、 請求の範囲第 4項に 記載の自動等化器。  The automatic equalizer according to claim 4, wherein after the update, the evening equalizer is controlled such that a transfer function of the target equalizer is obtained by dividing the factor. Chemist.
6 . 前記ターゲット等化器は複数の等化器が直列接続されてなり、 6. The target equalizer has a plurality of equalizers connected in series,
前記因数は該複数の等化器のいずれかの伝達関数に相当し、  The factor corresponds to a transfer function of any of the plurality of equalizers;
前記更新後、 上記相当する伝達関数を有する等化器をスルーとする設定を行な うように該ターゲット等化器を制御することを特徴とする、 請求の範囲第 4項に 記載の自動等化器。  The automatic equalizer according to claim 4, wherein after the update, the target equalizer is controlled so as to perform setting such that an equalizer having the corresponding transfer function is set to be through. Chemist.
7 . 該係数更新制御部が、 7. The coefficient update control unit:
該等化手段の出力と上記相当する伝達関数を有する等化器の出力との差分を等 化誤差信号として求める加算器と、  An adder for obtaining, as an equalization error signal, a difference between the output of the equalization means and the output of the equalizer having the above-described transfer function;
該加算器により得られた等化誤差信号から上記相当する伝達関数を有する等化 器の出力点についての等化誤差信号を求める夕ーゲット出力点誤差計算器とをそ なえ、  A target output point error calculator for obtaining an equalization error signal for an output point of the equalizer having the above equivalent transfer function from the equalization error signal obtained by the adder;
該加算器により得られた等化誤差信号に基づいて該 F I R型等化器及び上記相 当する伝達関数を有する等化器の各タップ係数を更新し、 該ターゲット出力点誤 差計算器により得られた等化誤差信号に基づいて上記相当する伝達関数を有する 等化器以外の等化器のタツプ係数を更新するように構成されたことを特徵とする、 請求の範囲第 6項に記載の自動等化器。  Based on the equalization error signal obtained by the adder, each tap coefficient of the FIR equalizer and the equalizer having the above-mentioned corresponding transfer function is updated, and obtained by the target output point error calculator. 7. The method according to claim 6, wherein a tap coefficient of an equalizer other than the equalizer having the corresponding transfer function is updated based on the obtained equalization error signal. Automatic equalizer.
8 . 該係数更新制御部が、 8. The coefficient update control unit:
一定期間は該ターゲット等化器の上記相当する伝達関数を有する等化器以外の 等化器をスルー設定とし上記相当する伝達関数を有する等化器及び該等化手段の 各タップ係数を更新し、 該一定期間経過後、 上記相当する伝達関数を有する等化 器以外の等化器を非スルー設定とし該ターゲット等化器を構成する各等化器及び 該 F I R型等化器の各係数をそれぞれ更新するように構成されたことを特徴とす る、 請求の範囲第 6項に記載の自動等化器。 During a certain period, the equalizers other than the equalizer having the corresponding transfer function of the target equalizer are set to the through setting, and the tap coefficients of the equalizer having the corresponding transfer function and the equalizer are updated. After the lapse of the certain period, the equalizers other than the equalizer having the above-described corresponding transfer function are set to the non-through setting, and the equalizers constituting the target equalizer and 7. The automatic equalizer according to claim 6, wherein each coefficient of said FIR equalizer is configured to be updated.
9 . 該係数更新制御部が、 9. The coefficient update control unit:
前記更新後のターゲット等化器の伝達関数を因数分解した因数の極点が複素平 面上の単位円の中に存在する所望の z関数の逆関数をもつように該 I I R型等化 器の夕ップ係数を設定するように構成されたことを特徴とする、 請求の範囲第 4 項に記載の自動等化器。  The IIR equalizer is set so that the poles of the factor obtained by factorizing the transfer function of the target equalizer after the update have the inverse function of the desired z function existing in the unit circle on the complex plane. 5. The automatic equalizer according to claim 4, wherein the automatic equalizer is configured to set a tap coefficient.
1 0 . 所定の伝送路からの受信信号について適応的に等化処理を施す自動等化 器の係数トレーニング方法であって、 10. A coefficient training method for an automatic equalizer that adaptively performs equalization processing on a received signal from a predetermined transmission path,
該伝送路からの受信信号に作用する等化手段の出力と、 該伝送路を含む該等化 手段の出力までのインパルス応答特性を擬似するターゲット等化器の出力とに基 づいて、 該等化手段及び該ターゲット等化器のタップ係数を更新し、  Based on the output of the equalizing means acting on the received signal from the transmission path and the output of the target equalizer simulating the impulse response characteristics up to the output of the equalization means including the transmission path, Updating the tap coefficients of the equalizing means and the target equalizer,
更新後の夕ーゲット等化器のタップ係数を用いて該等化手段の一部のタップ係 数を設定することを特徴とする、 .自動等化器の係数トレーニング方法。  A coefficient training method for an automatic equalizer, characterized in that a tap coefficient of a part of the equalizing means is set using a tap coefficient of the evening get equalizer after update.
PCT/JP2002/001085 2002-02-08 2002-02-08 Automatic equalizer and coefficient training method thereof WO2003075482A1 (en)

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Citations (5)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
JPS583414A (en) * 1981-06-30 1983-01-10 Fujitsu Ltd Impulse responsive automatic equalizer
JPH0548391A (en) * 1991-01-23 1993-02-26 Fujitsu Ltd Adaptive equalizer
JPH05152893A (en) * 1991-07-29 1993-06-18 Oki Electric Ind Co Ltd Adaptive equalizer
JPH05152894A (en) * 1991-07-29 1993-06-18 Oki Electric Ind Co Ltd Adaptive equalizer
JP2000244777A (en) * 1999-02-23 2000-09-08 Matsushita Electric Ind Co Ltd Waveform equalizing device

Patent Citations (5)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
JPS583414A (en) * 1981-06-30 1983-01-10 Fujitsu Ltd Impulse responsive automatic equalizer
JPH0548391A (en) * 1991-01-23 1993-02-26 Fujitsu Ltd Adaptive equalizer
JPH05152893A (en) * 1991-07-29 1993-06-18 Oki Electric Ind Co Ltd Adaptive equalizer
JPH05152894A (en) * 1991-07-29 1993-06-18 Oki Electric Ind Co Ltd Adaptive equalizer
JP2000244777A (en) * 1999-02-23 2000-09-08 Matsushita Electric Ind Co Ltd Waveform equalizing device

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