US 5,793,807 AGrant

Multimodulus Blind Eqalization Using Piecewise Linear Contours

Issue Date:1998-08-11

•31 Claims

•10 Drawing Sheets

Abstract

A blind equalization technique--the multimodulus algorithm (MMA)--adapts coefficients of an equalizer so as to minimize dispersion of the output samples of the equalizer around piecewise linear contours of a signal space. The MMA technique is illustrated in the context of both square and non-square signal point constellations.

Metadata

Assignee

Lucent Technologies Inc.

Inventors

Jean-Jacques Werner
Jian Yang

Application Information

Application Number:US 6464041

Filing Date:1996-05-07

Priority Date:1996-05-07

Art Unit:274

Classifications

IPC:

H04L 2701

Field of Search:

375364229;230;231;232;233;234;235;350724.2

Patent Drawings (10 sheets)

Description

BACKGROUND OF THE INVENTION

The present invention relates to communications equipment, and, more particularly, to blind equalization in a receiver.

For bandwidth-efficient digital transmission schemes, the initial training of a receiver consists of the three main steps of setting the automatic gain control (AGC), recovering timing (clock synchronization), and converging the adaptive filters. This initial training of the receiver is called start-up. For many applications, start-up is facilitated by using a training signal representing a known sequence of data. However, there are applications where the usage of a training signal is not desirable or not feasible. In this case, the receiver has to start up under so-called "blind" conditions.

The main challenge in the implementation of a blind start-up is to provide a reliable convergence of the adaptive filters, e.g., fractionally spaced linear equalizers (FSLEs), hereafter simply referred to as equalizers. Blind convergence of an equalizer is also known as "blind equalization." The philosophy of blind equalization is to use an equalizer tap adaptation algorithm that minimizes a predefined cost function to minimize signal point error and, hopefully, provide initial convergence of the equalizer.

Blind equalization takes much more time to converge an equalizer than what is required with a training signal. As a result, blind equalization is not applicable to applications requiring a very fast start-up or applications where the channel is highly time varying, e.g., mobile communications. Otherwise, blind equalization is always a desirable option, because of cost reasons, and sometimes it may be mandatory, because of system design considerations. For example, in a "Point-to-Multipoint Network" such as a fiber-to-the-curb (FTTC) network architecture, the downstream channel conveys a continuously transmitted signal to the receiver sections of all subscriber units. When one of the subscriber units is suddenly turned on, the corresponding receiver has to be trained. However, sending a training signal to this receiver in the downstream channel would disrupt transmission to other active receivers. Thus, this newly-activated receiver has to be started blindly.

As known in the art, there are two techniques for blind equalization: one is referred to herein as the "reduced constellation algorithm" (RCA) (e.g., see Y. Sato, "A Method of Self-Recovering Equalization for Multilevel Amplitude-Modulation Systems," IEEE Trans. Commun., pp. 679-682, June 1975; and U.S. Pat. No. 4,227,152, issued Oct. 7, 1980 to Godard); and the other technique is the so-called "constant modulus algorithm" (CMA) (e.g., see D. N. Godard, "Self-Recovering Equalization and Carrier Tracking in Two-Dimensional Data Communications Systems," IEEE Trans. Commun., vol. 28, no. 11, pp. 1867-1875, November 1980; and N. K. Jablon, "Joint Blind Equalization, Carrier Recovery, and Timing Recovery for High-Order QAM Signal Constellations", IEEE Trans. Signal Processing, vol. 40, no. 6, pp. 1383-1398, 1992.)

The main advantages of the RCA algorithm are its lower complexity and, as a result, its low cost of implementation. However, the main disadvantages of the RCA algorithm are its unpredictability and lack of reliability. It is known in the art that the RCA algorithm often converges to so-called "wrong solutions" that do not allow the receiver to recover the correct data.

In comparison the main advantages of the CMA algorithm are its reliability and predictability. Unlike the RCA algorithm, it rarely converges to wrong solutions. In addition, the CMA algorithm also has the advantage of being able to partially equalize the channel in the presence of carrier phase variations in some applications. However, this performance comes at a price--the CMA algorithm is more complex than the RCA algorithm and, consequently, costs more to implement. Unfortunately, there is another drawback to the CMA algorithm. The latter may additionally phase rotate the equalized signal points, thus requiring the use, and perhaps additional cost, of a phase rotator at the output of the equalizer to compensate for this effect.

SUMMARY OF THE INVENTION

We have realized a new blind equalization technique--the multimodulus algorithm (MMA)--which adapts coefficients of an equalizer so as to minimize dispersion of the output samples of the equalizer around piecewise linear contours of a signal space.

In comparison to the two prior art approaches mentioned above, the MMA algorithm has a number of advantages. With respect to the RCA algorithm, the MMA algorithm converges faster and is more reliable. With respect to the CMA algorithm, the MMA algorithm is less costly to implement since less processing is required to perform the convergence. Also, MMA does not additionally phase rotate the signal points so that a compensating phase rotator is not required as in the CMA approach.

It should also be noted that the MMA algorithm is especially attractive for carrierless AM/PM digital transmission systems, and can also be used for other modulation schemes such as quadrature amplitude modulation.

BRIEF DESCRIPTION OF THE DRAWINGS

FIG. 1 is an illustrative block diagram of a portion of a communications system embodying the principles of the invention;

FIG. 2 is an illustrative block diagram of a phase-splitting equalizer;

FIG. 3 is an illustrative block diagram of a portion of an adaptive filter for use in an equalizer;

FIG. 4 is an illustrative block diagram of a cross-coupled equalizer;

FIG. 5 is an illustrative block diagram of a four-filter equalizer;

FIG. 6 is an illustrative signal point plot of an output signal of an equalizer before convergence;

FIG. 7 is an illustrative signal point plot of an output signal of an equalizer for a system using the MMA blind equalization method;

FIG. 8 is an illustrative signal point plot illustrating the reduced signal point constellation of the RCA blind equalization method;

FIG. 9 is an illustrative signal point plot illustrating the circular contour of the CMA blind equalization method;

FIG. 10 is an illustrative signal point plot illustrating the piecewise linear contours of the MMA blind equalization method in accordance with the principles of the invention;

FIGS. 11 and 12 are illustrative block diagrams of a portion of a receiver embodying the principles of the invention;

FIGS. 13, 14, and 15, are illustrative signal point plots illustrating the piecewise linear contours of the MMA blind equalization method in accordance with the principles of the invention for a nonsquare constellation;

FIGS. 16 and 17 are illustrative signal point plots of an output signal of an equalizer for a communications system using a two-step blind equalization method in accordance with the principles of the invention;

FIG. 18 shows a table providing a general comparison between the RCA, CMA, and MMA, blind equalization methods; and

FIG. 19 shows a table of illustrative data values for use in the RCA, CMA, and MMA, blind equalization methods.

DETAILED DESCRIPTION

An illustrative high-level block diagram of a portion of a communications system embodying the principles of the invention is shown in FIG. 1. For illustrative purposes only, it is assumed that receiver 10 receives a CAP (carrierless, amplitude modulation, phase modulation) signal, which can be represented by: ##EQU1## where a.sub.n and b.sub.n are discrete-valued multilevel symbols, p(t) and p(t) are impulse responses which form a Hilbert pair, T is the symbol period, and .zeta.(t) is additive noise introduced in the channel. Additional information on a CAP communications system can be found in J. J. Werner, "Tutorial on Carrierless AM/PM--Part I--Fundamentals and Digital CAP Transmitter," Contribution to ANSI X3T9.5 TP/PMD Working Group, Minneapolis, Jun. 23, 1992.

It is assumed that the CAP signal in equation (1) has been distorted while propagating through communications channel 9 and experiences intersymbol interference (ISI). This ISI consists of intrachannel ISI (a.sub.n or b.sub.n symbols interfering with each other) and interchannel ISI (a.sub.n and b.sub.n symbols interfering with each other). The purpose of receiver 10 is to remove the ISI and minimize the effect of the additive noise .zeta.(t) to provide signal r'(t). Since the inventive concept is generally applicable to adaptive filters, the inventive concept will illustratively be described in the context of a number of different equalizer structures which can be used within receiver 10. However, before describing the inventive concept, some background information on equalizers and the above-mention RCA and CMA algorithms is presented.

Equalizer Structures

An illustrative phase-splitting FSLE equalizer 100 is shown in FIG. 2. It is assumed that FSLE equalizer 100 operates on an input signal comprising two dimensions: an in-phase component and a quadrature component. FSLE equalizer 100 comprises two parallel digital adaptive filters implemented as finite impulse response (FIR) filters 110 and 120. Equalizer 100 is called a "phase-splitting FSLE" because the two FIR filters 110 and 120 converge to in-phase and quadrature filters. Some illustrative details of the equalizer structure are shown in FIG. 3. The two FIR filters 110 and 120 share the same tapped delay line 115, which stores sequences of successive Analog-to-Digital Converter (A/D) 125 samples r.sub.k. The sampling rate l/T of A/D 125 is typically three to four times higher than the symbol rate l/T and is chosen in such a way that it satisfies the sampling theorem for real signals. It is assumed that T/T=i, where i is an integer.

The outputs of the two adaptive FIR filters 1 10 and 120 as shown in FIG. 3 are computed at the symbol rate l/T. The equalizer taps and input samples can be represented by a corresponding N-dimensional vector. As such, the following relationships are now defined:

where the superscript T denotes vector transpose, the subscript n refers to the symbol period nT, and k=in.

Let y.sub.n and y.sub.n be the computed outputs of the in-phase and quadrature filters, respectively, and:

An X/Y display of the outputs y.sub.n and y.sub.n or, equivalently, of the complex output Y.sub.n =y.sub.n +jy.sub.n, is called a signal constellation. FIGS. 6 and 17 show an 64-CAP constellation before and after illustrative convergence of the equalizer in accordance with the inventive concept. After convergence, the signal constellation consists of a display of the complex symbols A.sub.n =a.sub.n +jb.sub.n corrupted by some small noise and ISI.

In the normal mode of operation, decision devices (or slicers) 130 and 135 shown in FIG. 2 compare the sampled outputs y.sub.n and y.sub.n of equalizer 100 with valid symbol values a.sub.n and b.sub.n and makes a decision on which symbols have been transmitted. These sliced symbols will be denoted a.sub.n and b.sub.n. The receiver then computes the following in-phase and quadrature errors e.sub.n and e.sub.n :

and the tap coefficients of the two adaptive filters are updated using the familiar least-mean-squares (LMS) algorithm, i.e.,

where .alpha. is the step size used in the tap adjustment algorithm.

Turning now to FIG. 4, a cross-coupled FSLE, 200, is shown. For this equalizer structure, the A/D samples are first fed to two fixed in-phase and quadrature FIR filters, 210 and 205, respectively. In this case, the sampling rate l/T' of A/D 125 is typically equal to four times the symbol rate l/T. The outputs of the two fixed FIR filters are computed at a rate l/T" that is consistent with the sampling theorem for analytic signals as known in the art. The output signals are then fed to equalizer 200 having a so-called cross-coupled structure. Typically, l/T" is twice the symbol rate l/T.

The cross-coupled equalizer 200 uses two adaptive FIR filters 215a and 215b, each with tap vectors c.sub.n and d.sub.n. For simplicity, the same tap vector notations c.sub.n and d.sub.n (which have been used for the previous described equalizer 100 of FIG. 2) are used again. However, it should be clear to those skilled in the art that the tap vectors are different for the two types of equalizers. These two filters are each used twice to compute the outputs y.sub.n and y.sub.n of the equalizer. Let r.sub.n and r.sub.n be the output vectors of the in-phase and quadrature filters that are used to compute the outputs of the cross-coupled equalizer. The following definitions can be made:

The complex output Y.sub.n of the equalizer can be written in the following compact way:

where the asterisk * denotes complex conjugate. Making the following definitions for the sliced complex symbol A.sub.n and the complex error E.sub.n :

The LMS algorithm for updating the complex tap vector C.sub.n can be written as:

Turning now to FIG. 5, a four-filter FSLE is shown. Four-filter equalizer 300 has the same general structure as cross-coupled FSLE 200 shown in FIG. 4, except that the adaptive portion consists of four different filters rather than two filters which are used twice. For this reason it is called a four-filter FSLE. The two output signals of equalizer 300 are computed as follows:

Using the definitions for the in-phase and quadrature errors e.sub.n and e.sub.n in equations (7a) and (7b), the following tap updating algorithms for the four filters result:

Having generally described the structure of some prior-art equalizers as shown in FIGS. 2-5, a general overview of the concept of blind equalization will now be described using the equalizer structure of FIG. 2.

Concept of Blind Equalization

In the normal (steady-state) mode of operation, the decision devices in FIG. 2, i.e., slicers 130 and 135, compare the equalizer complex output samples, Y.sub.n, (where Y.sub.n =y.sub.n +jy.sub.n), with all the possible transmitted complex symbols, A.sub.n (where A.sub.n =a.sub.n +jb.sub.n), and selects the symbol A.sub.n which is the closest to Y.sub.n. The receiver then computes an error, E.sub.n, where:

which is used to update the tap coefficients of equalizer 100. This type of tap adaptation is called "decision directed", because it uses the decisions of slicers 130 and 135. The most common tap updating algorithm is the LMS algorithm, which is a stochastic gradient algorithm that minimizes the mean square error (MSE), which is defined as:

where E.! denotes expectation and e.sub.n and e.sub.n are in-phase and quadrature errors, respectively.

At the beginning of start-up, the output signal of equalizer 100, Y.sub.n, is corrupted by a lot of intersymbol interference, as illustrated in FIG. 6. The latter represents experimental data obtained for a 64-CAP receiver using a phase-splitting FSLE as represented by FIG. 2. (The term "64-CAP," refers to the number of predefined symbols in the signal space or signal constellation each symbol representing 6 bits since 2.sup.6 =64.)

When a training sequence is used during start-up (i.e., a predefined sequence of A.sub.n symbols), the receiver can compute meaningful errors E.sub.n by using the equalizer output signal Y.sub.n and the known sequence of transmitted symbols A.sub.n. In this case, tap adaptation is said to be done with "ideal reference" to distinguish it from decision directed tap adaptation.

However, when no training sequence is available, equalizer 100 has to be converged blindly. In this case, a decision-directed tap updating algorithm cannot be used to converge the equalizer, because the slicer makes too many wrong decisions, as should be apparent from FIG. 6.

As such, the philosophy of blind equalization is to use a tap adaptation algorithm that minimizes a cost function that is better suited to provide initial convergence of equalizer 100 than the MSE represented by equation (17). The cost functions used in the RCA and CMA algorithms, and those used in accordance with the inventive concept, are described below.

Convergence of an equalizer during blind start-up usually consists of two main steps. First, a blind equalization algorithm is used to open the "eye diagram." (Hereafter, this will be referred to as "it opens the eye.") Once the eye is open enough, the receiver switches to a decision directed tap adaptation algorithm.

Reduced Constellation Algorithm (RCA)

This section provides a general overview of the RCA algorithm. This general overview is then followed with a description of the RCA algorithm in the context of each of the illustrative equalizer structures, described above.

With the RCA algorithm, the error used in the tap updating algorithm is derived with respect to a signal constellation that has a smaller number of points than the received constellation. As illustration, it is again assumed that the signal constellation comprises 64 symbols. In the RCA algorithm, the reduced constellation typically consists of four signal points only, as shown in FIG. 8. It should be noted that the RCA algorithm requires the use of a decision device, e.g., a slicer, to select the closest signal point from the reduced constellation. The error between the received sample Y.sub.n and the closest signal point A.sub.r,n of the reduced constellation is the complex number:

where sgn (.) is the signum function and the expression on the right corresponds to the case where the reduced constellation consists of four points. The reduced constellation algorithm minimizes the following cost function:

where E .! denotes expectation.

Now, consider the phase-splitting equalizer structure shown in FIG. 2. Using equations (5), (6), and (20), the following equations result:

The gradients of the cost function represented by equation (20) with respect to the tap vectors c.sub.n and d.sub.n are equal to:

These gradients are equal to zero when the channel is perfectly equalized, i.e. when the received samples Y.sub.n are equal to the symbol values A.sub.n. This condition leads to the following value of R: ##EQU2##

For example, consider the gradient with respect to the tap vector c.sub.n. From the left of equations (21a) and (21b) there is the condition: E(y.sub.n -R sgn(y.sub.n))r.sub.n !=0. With perfect equalization y.sub.n =a.sub.n. Also, if it is assumed that different symbols are uncorrelated, then: Ea.sub.n r.sub.n !=k.sub.n Ea.sub.n.sup.2 !, where k.sub.n is a fixed vector whose entries are a function of the channel. The above condition can then be written as: Ea.sub.n.sup.2 !-R Esgn(a.sub.n)a.sub.n !=0. Noting that sgn (a.sub.n)a.sub.n =.vertline.a.sub.n .vertline. and solving for R, equation (23) results.

The nonaveraged gradients in equations (22a) and (22b) can be used in a stochastic gradient algorithm to adapt the tap coefficients of the equalizer, so that the following tap updating algorithms result:

Turning now to the cross-coupled FSLE structure illustrated by FIG. 4, the complex output Y.sub.n of this equalizer is computed from equation (10). Using this expression in equation (20), the gradient of the cost function with respect to the complex tap vector C.sub.n is:

Assuming a perfectly equalized channel the following expression for R results: ##EQU3## where the expression on the right is the same as the one in equation (23) for the usual case where E.vertline.a.sub.n .vertline.!=E.vertline.b.sub.n .vertline.!. The tap updating algorithm for the complex tap vector C.sub.n is given by

Turning now to the four-filter FSLE structure illustrated by FIG. 5, the outputs y.sub.n and y.sub.n of this four-filter equalizer structure are computed from equations (13a) and (13b). The gradients of the cost function in equation (20) with respect to the four tap vectors are similar to the ones given in equations (22a) and (22b) and will not be repeated here. The tap updating algorithms are given by:

where the constant R is the same as in equation (23).

The main advantage of RCA is its low cost of implementation because it is typically the least complex of blind equalization algorithms. The tap updating algorithms represented by equations (24a), (24b), (27) and (28) are the same as the standard LMS algorithms represented by equations (8a) and (8b) except that the slicer uses a different number of points.

The main disadvantages of RCA are its unpredictability and lack of robustness. The algorithm is known to often converge to so-called "wrong solutions." These solutions are quite acceptable from a channel equalization perspective, but do not allow the receiver to recover the right data. It should be pointed out that the equalizer structure in FIG. 2 is much more likely to converge to wrong solutions than the structure in FIG. 4. This is due to the fact that the former has many more degrees of freedom than the latter.

A wrong solution that is often observed with the equalizer structure in FIG. 2 is the so-called diagonal solution. In this case, the in-phase and quadrature filters both converge to the same filter, so that they both generate the same output samples. As a result, the signal constellation at the output of the equalizer consists of points clustered along a diagonal.

Other wrong solutions can occur when the in-phase and quadrature filters introduce propagation delays which differ by an integral number of symbol periods. As an example, at a given sampling instant, a, may appear at the output of the in-phase filter while b.sub.n-1 appears at the output of the quadrature filter. This kind of wrong solution can generate points in the signal constellation at the output of the equalizer that do not correspond to transmitted symbols. For example, a 32-point signal constellation may be converted into a 36-point constellation and the 128-point constellation in FIGS. 13, 14, and 15 may be converted into a 144-point constellation.

Constant Modulus Algorithm (CMA)

This section provides a general overview of the CMA algorithm. This general overview is then followed with a description of the CMA algorithm in the context of each of the illustrative equalizer structures, described above.

The CMA algorithm minimizes the dispersion of the equalized samples Y.sub.n with respect to a circle with radius R. This is graphically illustrated in FIG. 9. The CMA algorithm minimizes the following cost function:

where L is a positive integer. The case L=2 is the most commonly used in practice. The cost function in equation (29) is a true two-dimensional cost function which minimizes the dispersion of the equalizer complex output signal Y.sub.n with respect to a circular two-dimensional contour.

Now, consider the phase-splitting equalizer structure shown in FIG. 2. The gradients of the cost function with respect to the tap vectors c.sub.n and d.sub.n are given by:

Assuming a perfectly equalized channel the following value for R.sup.L results: ##EQU4## where the expression on the right holds for the usual case where the statistics of the symbols a.sub.n and b.sub.n are the same. For L=2, the following stochastic gradient tap updating algorithms results:

Turning now to the cross-coupled FSLE structure illustrated by FIG. 4, the gradient of the cost function represented by equation (29) with respect to the complex tap vector C.sub.n is equal to:

For L=2, the tap updating algorithm for the complex tap vector becomes:

where R is given by the expression on the right in equation (31).

Turning now to the four-filter FSLE structure illustrated by FIG. 5, the gradients of the cost function represented by equation (29) with respect to the four tap vectors are similar to the ones given by equations (30a) and (30b). For L=2, the tap updating algorithms become:

The constant R is the same as in equation (31).

The main advantages of CMA are its robustness and predictability. Unlike RCA, it rarely converges to wrong solutions. For some applications, other than those considered here, it also has the advantage of being able to partially equalize the channel in the presence of carrier phase variations. The main disadvantage of CMA is its cost of implementation. The CMA tap updating algorithm is more complex than that of the RCA algorithm and the inventive concept (described below), and, in addition, the CMA algorithm requires a so-called "rotator" at the output of the equalizer. As a result, once a certain degree of convergence is achieved, the output signal of the equalizer must be counter-rotated before switching to a decision-directed tap adaptation algorithm. The need to use a rotator after the equalizer increases the cost of implementation of CMA for the some type of applications. It should be pointed out, however, that there are other applications, such as voiceband and cable modems, where the rotator function is required anyway for other purposes, such as tracking frequency offset introduced in the channel. In these latter cases, the need to do a rotation does not increase the cost of implementation, and CMA becomes a very attractive approach.

Multimodulus Algorithm (MMA)

In accordance with the inventive concept, the new blind equalization algorithm presented here minimizes the dispersion of the equalizer output samples y.sub.n and y.sub.n around piecewise linear in-phase and quadrature contours. This algorithm is called the multimodulus algorithm (MMA). For the special case of square signal constellations of the type used for 16-, 64-, and 256-CAP systems, the contours become straight lines. This is graphically illustrated in FIG. 10 for a 64-point constellation. The multimodulus algorithm minimizes the following cost function:

where L is a positive integer and R(Y.sub.n,) and R(Y.sub.n,) take discrete positive values, which depend on the equalizer outputs Y.sub.n.

Illustrative embodiments of the inventive concept are shown in FIGS. 11 and 12. FIG. 11 illustrates an embodiment representative of a digital signal processor 400 that is programmed to implement an FSLE in accordance with the principles of the invention. Digital signal processor 400 comprises a central processing unit (processor) 405 and memory 410. A portion of memory 410 is used to store program instructions that, when executed by processor 405, implement the MMA algorithm. This portion of memory is shown as 411. Another portion of memory, 412, is used to store tap coefficient values that are updated by processor 405 in accordance with the inventive concept. It is assumed that a received signal 404 is applied to processor 405, which equalizes this signal in accordance with the inventive concept to provide a output signal 406. For the purposes of example only, it is assumed that output signal 406 represents a sequence of output samples of an equalizer. (As known in the art, a digital signal processor may, additionally, further process received signal 404 before deriving output signal 406.) An illustrative software program is not described herein since, after learning of the MMA algorithm as described herein, such a program is within the capability of one skilled in the art. Also, it should be noted that any equalizer structures, such as those described earlier, can be implemented by digital signal processor 400 in accordance with the inventive concept.

FIG. 12 illustrates another alternative embodiment of the inventive concept. Circuitry 500 comprises a central processing unit (processor) 505, and an equalizer 510. The latter can be any equalizer structure, e.g., such as those described earlier. It is assumed that equalizer 510 includes at least one tap-coefficient register for storing values for corresponding tap coefficient vectors (e.g., as shown in FIG. 3). Processor 505 includes memory, not shown, similar to memory 410 of FIG. 11 for implementing the MMA algorithm. Equalizer output signal 511, which represents a sequence of equalizer output samples, is applied to processor 505. The latter analyzes equalizer output signal 511, in accordance with the inventive concept, to adapt values of the tap coefficients to minimize dispersion of the equalizer output samples around piecewise linear contours.

Multimodulus Algorithm (MMA)--Square Constellations

For square constellations, R(Y.sub.n)=R(Y.sub.n )=R=constant, so that the cost function of equation (36) becomes:

Unlike the cost function for CMA represented by equation (29), this is not a true two-dimensional cost function. Rather, it is the sum of two independent one-dimensional cost functions CF.sub.I and CF.sub.Q. The application of the MMA algorithm in the context of the three illustrative types of equalizers (described above) will now be described.

For the phase-splitting equalizer structure shown FIG. 2, the gradients of the cost function in equation (37) with respect to the tap vectors c.sub.n and d.sub.n are equal to:

Assuming a perfectly equalized channel, the following value for R.sup.L results: ##EQU5##

The best compromise between cost and performance is achieved with L=2, in which case the tap updating algorithms become

Turning now to the cross-coupled FSLE structure illustrated by FIG. 4, the gradient of the cost function represented by equation (37) with respect to the complex tap vector C.sub.n is given by:

where,

Assuming a perfectly equalized channel, the value for R.sup.L is: ##EQU6## which reduces to equation (39) for the usual case where the symbols a.sub.n and b.sub.n have the same statistics. For L=2, the tap updating algorithm for the complex tap vector C.sub.n becomes:

where,

Turning now to the four-filter FSLE structure illustrated by FIG. 5, the gradients of the cost function represented by equation (37) with respect to the four tap vectors are similar to the ones given in equations (38a) and (38b). For L=2, the tap updating algorithms become:

The constant R is the same as in equation (39).

In accordance with the inventive concept, the above-mentioned two-step blind equalization procedure utilizing the MMA algorithm is graphically illustrated by FIGS. 6, 16, and 17 for equalizer 100. The output signal of equalizer 100, before any form of convergence, is shown in FIG. 6. As noted above, FIG. 6 represents experimental data obtained for a 64-CAP receiver using a phase-splitting FSLE as represented by FIG. 2. FIG. 7 illustrates the beginning of the MMA process convergence. As shown in FIG. 16, and in accordance with the inventive concept, the MMA technique converges the equalizer enough to clearly illustrate the 64-symbol signal space as 64 noisy clusters. Although these noisy clusters would, typically, not be acceptable for steady-state operation, the eye is open enough to allow the receiver to switch to a 64-point slicer and a decision-directed LMS algorithm. The end result is a much cleaner constellation, as shown in FIG. 17. Typically, a clean transition can be made between the two modes of adaptation, MMA and decision directed, when the symbol error rate is better than 10.sup.-2, although successful transitions have been observed for worse symbol error rates. It should be pointed out that the noisy clusters in FIG. 16 could be further reduced by decreasing the step size in the MMA tap adjustment algorithm. Indeed, in some applications it may be possible to eliminate the switching to a decision directed tap adaptation algorithm. However, it should be noted that this would increase the start-up time and the required amount of digital precision.

The MMA algorithm for square constellations can be used without modification for nonsquare constellations. In this case, caution has to be exercised in the computation of the constant R, because the discrete levels for the symbols a.sub.n and b.sub.n do not all have the same probability of occurrence (described below). However, it has been found through computer simulations that convergence of the MMA algorithm is somewhat less reliable for nonsquare constellations than for square constellations. This can be corrected by using the modified MMA discussed in the following section.

Multimodulus Algorithm (MMA)--NonSquare Constellations

The principle of the modified MMA is illustrated in FIGS. 13, 14, and 15, with respect to a 128-CAP signal constellation. (A 128-point signal constellation is obtained in the following way. First define a 144-point signal constellation using the symbol levels .+-.1,.+-.3,.+-.5,.+-.7,.+-.9,.+-.11, and then remove the four corner points in each quadrant.) Minimization of the dispersion of the equalizer output samples y.sub.n and y.sub.n is now done around piecewise straight lines. Again, this is done independently for y.sub.n and y.sub.n. The quadrature cost functions derived from equation (37) are:

The in-phase cost functions derived from equation (37) are:

The constant K is a function of the signal constellation under consideration and is determined empirically. In computer simulations for 128-CAP, a suggested value is K=8. Two different moduli R.sub.1 and R.sub.2 are used in equations (47) because the symbols a.sub.n and b.sub.n used in the 128-point constellation have two sets of levels {.+-.1,.+-.3,.+-.5,.+-.7} and {.+-.9, .+-.11} which have a different probability of occurrence. More moduli can be used if there are more than two sets of symbol levels with different statistics.

The moduli R.sub.1 and R.sub.2 in equations (47) are computed from equation (39) by evaluating the moments of the symbols over the set of symbol levels to which a given modulus applies (additionally described below). As an example, consider FIG. 13, which illustrates the moduli for the in-phase dimension, and which applies to the real symbols a.sub.n of a 128-CAP signal constellation. The moments of the symbols can be computed by considering the first quadrant only. Consider the subset of 24 symbols in this quadrant that applies to R.sub.1. For these symbols a.sub.n =1, 3, 5, 7, 9, 11; and b.sub.n =1, 3, 5, 7; so that each value of a.sub.n occurs with probability 4/24=1/6. Similarly, the R.sub.2 subset has 8 symbols for which a.sub.n =1, 3, 5, 7 and b.sub.n =9, 11, so that each value of a.sub.n occurs with probability 2/8=1/4. Thus, the variance of the symbols becomes: ##EQU7##

Other moments of the symbols are computed in a similar fashion and then used in equation (39) to evaluate the values of the various moduli.

The tap updating algorithms for the modified MMA algorithm are the same as the ones given in equations (40), (44), and (46), except that the constant R is replaced by either R.sub.1 or R.sub.2 depending on which equalizer output sample Y.sub.n is received. FIG. 14 illustrates the moduli for the quadrature dimension, and which applies to the symbols b.sub.n of the 128-CAP signal constellation. It should be apparent from FIG. 15, which represents the union of FIGS. 13 and 14, that the in-phase and quadrature tap updating algorithms need not use the same moduli R.sub.1 or R.sub.2 in a given symbol period.

As described above, the inventive concept provides an alternative approach to performing blind equalization. A general comparison of the RCA, CMA, and MMA techniques is shown in the table of FIG. 18. In addition, the table shown in FIG. 19 shows illustrative values, for signal constellations of different sizes, of the constants R, R.sub.1, and R.sub.2, which are used in the tap updating algorithms of the RCA, CMA, and MMA, blind equalization techniques described above. The data shown in FIG. 19 assumes that the symbols a.sub.n and b.sub.n take the discrete values .+-.1, .+-.3,.+-.5,.+-.7, . . . . The closed-form expressions for these constants are derived as described below.

It should be noted that, although particularly noticeable in the RCA algorithm (as described above), both the CMA and MMA algorithms can, on some occasions, also converge to the wrong solution under certain conditions, e.g., the above-mentioned "diagonal solution." However, the co-pending, commonly assigned, U.S. patent application of Werner et al., entitled "Technique for Improving the Blind Convergence of a Two-Filter Adaptive Equalizer," Ser. No. 08/717,582, filed on Sep. 18, 1996, describes a technique that improves convergence across all blind equalization methods for phase-splitting FSLEs.

Moments of Data Symbols

The following description discusses the concept of "moments of data symbols." In particular, the closed-form expressions for the moments E.vertline.a.sub.n .vertline..sup.L !, E.vertline.b.sub.n .vertline..sup.L !, and E.vertline.A.sub.n .vertline..sup.L ! when the symbols a.sub.n and b.sub.n take values proportional to the odd integers .+-.1,.+-.3,.+-.5,.+-.7, . . . , are presented. These expressions are then used to get closed-form expressions for the constants R used in the three blind equalization algorithms and illustrated in the table of FIG. 19.

First, it is assumed that the symbols an and b, have the same statistics, so that Ea.sub.n .vertline..sup.L !=E.vertline.b.sub.n .vertline..sup.L !. Consider first the following known summations of powers of integers: ##EQU8##

These summations can be used to find closed-form expressions for sums of powers of odd integers. For example, for power one: ##EQU9## where the two summations in the middle have been evaluated by using the closed-form expression of equation (49a). Similar series manipulations can be used for other sums of powers of odd integers.

Now, consider square signal constellations which use symbols a.sub.n and b.sub.n with values .+-.1,.+-.3,.+-.5,.+-.7, . . . .+-.(2m-1), where m is the number of different symbol levels (in magnitude). As an example, for the 4-CAP, 16-CAP, 64-CAP, and 256-CAP square signal constellations, m=1, 2, 4, and 8, respectively. It is also assumed that all the symbol values are equiprobable. As a result, the moments of the symbols a.sub.n are: ##EQU10##

Next, consider the complex symbols A.sub.n =a.sub.n +jb.sub.n. Assuming, that the symbols a.sub.n and b.sub.n are uncorrelated, the following expressions for the even moments of the complex symbols result:

Using equations (52) and (54) in equation (55b), results in: ##EQU11##

The above results can now be used to get closed-form expressions for the constants R used in the various blind equalization algorithms. The following (remarkably simple) expressions for these constants result: ##EQU12##

With respect to nonsquare signal constellations, the various symbol levels 2k-1 for a.sub.n and b.sub.n have a different probability of occurrence, even when all the complex symbols A.sub.n are equiprobable. This should be apparent from the 128-point constellation illustrated by FIG. 15. In this case, the moments of the symbols have to be computed according to the general formula: ##EQU13## where P.sub.1 is the probability of occurrence of the symbol levels appearing in the corresponding summation. For typical 32-CAP and 128-CAP constellations the expression in (60) is restricted to two different probabilities P.sub.1 and P.sub.2.

Everything else being equal (i.e. symbol rate, shaping filters, etc.), it is possible to guarantee a constant average power at the output of a CAP transmitter if Ea.sub.n.sup.2 !=Eb.sub.n.sup.2 !=constant, independently of the type of signal constellation that is used. Of course, different signal constellations will have to use different symbol values if the average power constraint has to be satisfied. Thus, in general, a signal constellation will use symbol values .lambda.(2k-1) where .lambda. is chosen in such a way that the average power constraint is satisfied. For simplicity, it is assumed that Ea.sub.n.sup.2 !=1. For square constellations, the value of .lambda. can then be determined from equation (52), to result in: ##EQU14##

Using this expression of .lambda. in equations (57), (58), and (59), the following expressions for the normalized constants R result: ##EQU15##

Similar expressions can be obtained for nonsquare constellations in a similar fashion. When the number of points in the signal constellation becomes very large, the following asymptotic values for the normalized constants result: ##EQU16##

The foregoing merely illustrates the principles of the invention and it will thus be appreciated that those skilled in the art will be able to devise numerous alternative arrangements which, although not explicitly described herein, embody the principles of the invention and are within its spirit and scope.

For example, although the invention is illustrated herein as being implemented with discrete functional building blocks, e.g., an equalizer, etc., the functions of any one or more of those building blocks can be carried out using one or more appropriate programmed processors.

In addition, although the inventive concept was described in the context of an FSLE, the inventive concept is applicable to other forms of adaptive filters, such as, but not limited to, a decision feedback equalizer (DFE). Indeed, the inventive concept can be easily extended to provide joint blind convergence of an adaptive equalizer and an adaptive echo or near-end cross-talk (NEXT) canceller.

The inventive concept is applicable to all forms of communications systems, e.g., broadcast networks, e.g., high-definition television (HDTV), point-to-multipoint networks like fiber to the curb (mentioned above), signal identification or classification applications, like wire-tapping, etc.

Claims

What is claimed:

1. A blind equalization method comprising the steps of: equalizing a signal to provide a sequence of equalizer output samples; and adapting coefficients of the equalizer to minimize dispersion of the equalizer output samples around piecewise linear contours of a signal space, each output sample having a signal point position in the signal space.

2. The method of claim 1 wherein the adapting step includes the step of calculating a new set of tap coefficient values as a function of a difference between a) a value of each signal point, of each output sample, raised to a power L and b) a magnitude of a distance of at least one piecewise linear contour from the origin of the signal space raised to the power L, where L>0.

3. The method of claim 1 wherein the piecewise linear contours comprise at least a set of quadrature contours and a set of in-phase contours.

4. The method of claim 1 wherein the adapting step minimizes the dispersion as a function of N independent one-dimensional cost functions.

5. The method of claim 1 further comprising the steps of: determining a mean squared error of the received signal; comparing the determined mean squared error with a predetermined value; and changing the adapting step when the determined mean squared error is less than a predefined value such that the adapting step now performs decision directed adaptation.

6. The method of claim 5 wherein decision directed adaptation includes the steps of: slicing each equalizer output sample to provide an estimated received symbol; determining an error value between the estimated received symbol and the corresponding equalizer output sample; and adapting coefficients of the equalizer as a function of a least-mean-squares function that uses the determined error value.

7. The method of claim 5 wherein the decision directed adaptation step includes the steps of: slicing each equalizer output sample to provide an estimated received symbol; and adapting the coefficients of the equalizer as a function of a difference between the estimated received signal and the corresponding output sample.

8. A method for converging a finite impulse response filter, the method comprising the steps of: processing a received signal with the finite impulse response filter to provide a sequence of output samples over a time period; and adapting coefficients of the finite impulse response filter to minimize dispersion of the sequence of output samples around piecewise linear contours of a signal space.

9. The method of claim 8 wherein the adapting step includes the step of calculating a new set of coefficient values as a function of a difference between a) a value of each output sample in the signal space raised to a power L and b) a magnitude of a distance of at least one piecewise linear contour from the origin of the signal space raised to the power L, where L>0.

10. The method of claim 8 further comprising the steps of: determining a mean squared error of the received signal; comparing the determined mean squared error with a predetermined value; and changing the adapting step when the determined mean squared error is less than a predefined value such that the adapting step now performs decision directed adaptation of the coefficients of the finite impulse response filter.

11. The method of claim 10 wherein decision directed adaptation includes the steps of: slicing each one of the sequence of output samples to provide a corresponding estimate of a received symbol; and adapting the coefficients of the finite impulse response filter as a function of a difference between each one of the sequence of output samples and the corresponding estimate.

12. The method of claim 11 wherein the adapting step adapts the coefficients as a function of a least-mean-squares function.

13. A method for use in converging an adaptive filter that processes a received signal to provide an output signal representative of a sequence of output samples, each output sample corresponding to a signal point within a signal space and the adaptive filter having a set of tap coefficient values, the method comprising the steps of: a) establishing a number of piecewise linear contours in the signal space; and b) calculating a new set of tap coefficient values as a function of a difference between 1) a value of each signal point, of each output sample, raised to a power L, and 2) at least one corresponding value of one of the number of piecewise linear contours raised to the power L, where L>0.

14. The method of claim 13 wherein step (a) generates a number of piecewise linear contours, each a distance from an origin of the signal space, where the magnitude of the distance of each contour from the origin of the signal space is the corresponding value of each piecewise linear contour.

15. The method of claim 14 wherein the number of piecewise linear contours comprises at least two in-phase contours and at least two quadrature contours.

16. The method of claim 15 wherein each piecewise linear contour is the same distance R from an origin of the signal space, where the magnitude of the distance R is the corresponding value of each piecewise linear contour.

17. The method of claim 14 wherein the number of piecewise linear contours comprises a plurality of in-phase contours and a plurality of quadrature contours.

18. The method of claim 17 wherein the plurality of in-phase contours comprise a plurality of in-phase subsets and the plurality of quadrature contours comprise a plurality of quadrature subsets such that those contours within the same subset have the same corresponding value.

19. The method of claim 14 wherein the number of piecewise linear contours are grouped into a plurality of subsets such that those contours within the same subset have the same corresponding value.

20. The method of claim 13 further comprising the steps of c) calculating an estimated error rate of the received signal; and d) performing steps b) and c) until the calculated error rate is less than a predetermined value.

21. The method of claim 20 further comprising the step of: e) performing decision directed adaptation of the tap coefficient values when the calculated error rate is less than the predetermined value.

22. An improved equalizer for use in a receiver for performing blind equalization, the improvement comprising: a processor that adapts coefficients of the equalizer to minimize dispersion of output samples of the equalizer around piecewise linear contours of a signal space.

23. The improvement of claim 22 wherein the processor is a digital signal processor.

24. The improvement of claim 22 wherein the processor adapts the coefficients of the equalizer as a function of a difference between a) a value of each output sample in the signal space raised to a power L and b) a magnitude of a distance of at least one piecewise linear contour from the origin of the signal space raised to the power L, where L>0.

25. An improved apparatus for use in a receiver for performing blind equalization, the improvement comprising: a processor for a) providing an equalizer function for equalizing a received signal, and b) adapting coefficients of the equalizer function to minimize dispersion of equalizer output samples around piecewise linear contours of a signal space.

26. The improvement of claim 25 wherein the processor is a digital signal processor.

27. The improvement of claim 25 wherein the processor adapts the coefficients as a function of a difference between a) a value of each output sample in the signal space raised to a power L and b) a magnitude of a distance of at least one piecewise linear contour from the origin of the signal space raised to the power L, where L>0.

28. Apparatus for use in performing blind equalization, the apparatus comprising: an equalizer having a set of tap coefficients and for providing an output signal representing a sequence of output samples, each output sample corresponding to a position in a signal space; a processor responsive to the output signal for adapting the set of tap coefficients in such a way as to minimize dispersion of the output samples around piecewise linear contours of the signal space.

29. The apparatus of claim 28 wherein the processor adapts the coefficients as a function of a difference between a) a value representing the position of each output sample in the signal space raised to a power L and b) a magnitude of a distance of at least one piecewise linear contour from the origin of the signal space raised to the power L, where L>0.

30. Apparatus for use in performing blind equalization, the apparatus comprising: a memory for storing a multimodulus algorithm and for storing a set of tap coefficients; and a processor a) for filtering an input signal as a function of the stored set of tap coefficients to provide an output signal, which represents a stream of output samples, each output sample having a position in a signal space, and b) for executing the multimodulus algorithm to adapt values of the set of tap coefficients in such a way as to minimize dispersion of the output samples around piecewise linear contours of the signal space.

31. The apparatus of claim 30 wherein the processor adapts values of the set of tap coefficients as a function of a difference between a) a value representing the position of each output sample in the signal space raised to a power L and b) a magnitude of a distance of at least one piecewise linear contour from the origin of the signal space raised to the power L, where L>0.

Patent Citations (1)

Patent	Date	Inventor	Cited By
US4227152	1980-10-01	Godard et al.

Non-Patent Literature (10)

Sethares, W., Rey, G.A. and Johnson, C.R., Jr., "Approaches to Blind Equalization of Signals with Multiple Modulus," Proc. 1989 Int'l Conf. on Acoustics, Speech and Signal Processing, pp. 972-975, May 1989.
Axford, R.A., Jr., Milstein, L.B. and Zeidler, J.R., "The Transversal and Lattice Multiple Modulus Algorithms for Blind Equalization of QAM Signals," Proc. IEEE Military Communications Conference, pp. 589-591, Nov. 1995.
IEEE Transactions Communications, vol. Com-28, No. 11, Nov. 1980, "Self-recovering equalization and carrier tracking in two dimensional data communications systems", by Dominique N. Godard.
K.N. Oh and Y.O. Chin, "New Blind Equalization Techniques Based on Constant Modulus Algorithm", IEEE, May 1995, pp. 865-869.
Y. Sato, "A Method of Self-Recovering Equalization for Multilevel Amplitude-Modulation Systems, " IEEE Trans. Commun., pp. 679-682, Jun. 1975.
A. Benveniste and M. Goursat, "Blind Equalizers", IEEE Trans. Commun., vol. 32, No. 8, pp. 871-883, 1984.
N. K. Jablon, "Joint Blind Equalization, Carrier Recovery, and Timing Recovery for High-Order QAM Signal Constellations", IEEE Trans. Signal Processing, vol. 40, No. 6, pp. 1383-1897, 1992.
S. Chen, S. McLaughlin, P.M. Grant and B. Mulgrew, "Multi-Stage Blind Clustering Equaliser", IEEE Trans. Commun., vol. 43, No. 2/3/4, pp. 701-705, Feb./Mar./Apr. 1995.
J. K. Tugnait, "Blind Equalization and Estimation of Digital Communication FIR Channels Using Cumulant Matching", IEEE Trans. Commun., vol. 43, No. 2/3/4 pp. 1240-1245, Feb./Mar./Apr. 1995.
J. J. Werner, "Tutorial on Carrierless AM/PM--Part I--Fundamentals and Digital CAP Transmitter", Contribution to ANSI X3T9.5 TP/PMD Working Group, Minneapolis, Jun. 23, 1992.

Abstract

Description