U.S. patent application number 08/997491 was filed with the patent office on 2001-11-15 for time domain equalizer for adsl.
Invention is credited to NAFIE, MOHAMMED, WU, SONG.
Application Number | 20010040920 08/997491 |
Document ID | / |
Family ID | 26710838 |
Filed Date | 2001-11-15 |
United States Patent
Application |
20010040920 |
Kind Code |
A1 |
NAFIE, MOHAMMED ; et
al. |
November 15, 2001 |
TIME DOMAIN EQUALIZER FOR ADSL
Abstract
An equalization technique for an ADSL data communications system
is disclosed. The effective channel overall response is made a
window of length I equal to the length of the cyclic prefix. The
response energy output of filtering is monitored against a preset
threshold value to adaptively position the window.
Inventors: |
NAFIE, MOHAMMED;
(MINNEAPOLIS, MN) ; WU, SONG; (PLANO, TX) |
Correspondence
Address: |
TEXAS INSTRUMENTS INCORPORATED
P O BOX 655474, M/S 3999
DALLAS
TX
75265
|
Family ID: |
26710838 |
Appl. No.: |
08/997491 |
Filed: |
December 23, 1997 |
Related U.S. Patent Documents
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Application
Number |
Filing Date |
Patent Number |
|
|
60034341 |
Dec 23, 1996 |
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Current U.S.
Class: |
375/229 ;
375/222 |
Current CPC
Class: |
H04L 25/0212 20130101;
H04L 2025/03522 20130101; H04L 2025/03414 20130101; H04L 25/03159
20130101 |
Class at
Publication: |
375/229 ;
375/222 |
International
Class: |
H03H 007/30 |
Claims
What is claimed is:
1. A time domain equalizer for an ADSL telecommunications system,
substantially as shown and described.
Description
[0001] This application claims priority under 35 U.S.C. .sctn.
119(e)(1) of provisional U.S. Ser. No. 60/034,341 filed Dec. 23,
1996, entitled "Time Domain Equalizer for ADSL," the entirety of
which is incorporated herein by reference.
[0002] This invention relates to digital subscriber loop
telecommunications systems and, in particular, to channel
equalizers for the same.
BACKGROUND OF THE INVENTION
[0003] The current ADSL standard is based on discrete multi-tone
signalling (DMT) protocol, wherein each frame is comprised of the
IFFT of a number of samples. In ADSL channels, and at such high
data rates, it is computationally expensive to fully equalize the
channel. Hence, only partial equalization is implemented to
constrain the overall effective channel length to a certain number
of samples, say v. Then the last v samples of each frame are used
as a prefix to the frame, so that in effect the time domain data is
cyclically convolved with the effective overall channel response,
composed of the physical channel and the equalizer. FIG. 1 shows
the overall situation.
[0004] The current equalization technique relies on frequency
domain lms (least mean square) is computationally expensive, is
nonlinear, and its steady state error is high.
SUMMARY OF THE INVENTION
[0005] The invention provides an equalization technique that relies
on time domain lms, has lower computational complexity and has
lower steady state error.
BRIEF DESCRIPTION OF THE DRAWINGS
[0006] Embodiments of the invention have been chosen for purposes
of illustration and description and are described with reference to
the accompanying drawings, wherein:
[0007] FIG. 1 is a block diagram showing the DMT signalling
system.
[0008] FIG. 2 is a block diagram showing a prepared equalizer
implementation.
[0009] FIG. 3 shows the system response after application of the
equalizer of FIG. 2.
[0010] FIGS. 4 and 5 illustrate the convergence of both
equalization methods.
DETAILED DESCRIPTION OF PREFERRED EMBODIMENTS
[0011] General and additional details concerning digital subscriber
loop modem telecommunications are given in copending patent
application Ser. No. 08/667,267, filed Jun. 20, 1996, entitled
"Multi-Mode Digital Modem," the entirety of which is hereby
incorporated by reference.
[0012] If the channel response is assumed to be given by h and the
equalizer by w, then the overall response will be given by b=h*w.
The objective is to make this effective channel overall response of
length equal to the length of the cyclic prefix, I, sent as an
overhead on each DMT frame. In other words, since achieving this
exactly would probably be impractical computationally--or even
theoretically impossible--with a finite length equalizer, what is
wanted is to have a window with size I with most of the energy of b
inside this window, or to decrease any energy outside that window.
Now, from the training sequence a pretty good estimate of h can be
obtained. This can then be convolved with the equalizer to get an
estimate of b.
[0013] Therefore, if it can be known beforehand where the best
place is to position the maximum energy window over the overall
length of b, the lms algorithm can be applied with the input being
h and the error signal being the components of b outside the
window. One may choose to position the window to coincide with the
position of the maximum energy window over h but this doesn't
always work. Another method would be to run the lms algorithm
several times for several window positions. This might require a
large amount of computation. So, instead of that the value of b
that is output from filtering h with w can be monitored, and when
this value exceeds a certain preset threshold, the window applied.
Thus, this window will change its position adaptively with the
adapted w, and better performance should be expected.
[0014] The method can be summarized as follows:
[0015] 1. Perform FFT on the received training data to get Y.
[0016] 2. To get an estimate of the frequency response of the
channel, divide Y by the training sequence X. Perform an IFFT to
get an estimate of h. Averaging a large number of frames will give
a good estimate of the true h.
[0017] 3. Start with any initial estimate of w.
[0018] a. Filter h using w.
[0019] b. Update w using the lms algorithm, with the error equal to
the output of the filter, and the input equal to h . The equation
is thus, w=w+2*.about.*b(n)*h, where h is the vector composed of
the components of h currently in the equalizer. While doing so,
monitor the output of this filtering process.
[0020] c. When the output exceeds a preset threshold, assume the
lms error equal to zero for a number for outputs equal to 1.
[0021] d. Continue filtering and updating for the whole length of
h.
[0022] e. Repeat the above steps until the lms converges.
[0023] FIG. 1 illustrates an implementation of such a
procedure.
[0024] Although this technique converges faster and offers
performance which is better than frequency domain lms, it can be
made to converge even faster. This is achieved by sticking an all
1's filter before the w filter and then using time domain Ims with
the error equal to the output when not in the window, and the input
is the output of the all 1's filter.
[0025] FIG. 3 shows the results of applying the previous two
techniques to a channel.
[0026] FIGS. 4 and 5 show the convergence of both methods.
[0027] Those skilled in the art to which the invention relates will
appreciate that various substitutions and modifications may be made
to the foregoing embodiments, without departing from the spirit and
scope of the invention as defined in the claims.
* * * * *