Reduced complexity time-frequency trained equalizer for discrete multi-tone based DSL systems
View Patent ↗The present invention provides a system and method for implementing a new TEQ training approach that trains TEQ coefficients by exploiting both time-domain and frequency-domain information. An advantage of this technique is that it reduces memory usage due to the training process. In addition, the complexity of the training process is simplified, and the associated computational work is reduced. The reduction of memory usage and computational work in turn may lead to cost savings, power consumption savings and other advantages.
1. A method for determining equalizer coefficients for channel equalization in Discrete-Multi-Tone based Digital Subscriber Line systems, the method comprising:
determining an impulse response value in a time domain;
applying windowing to the impulse response value to determine a windowed impulse response value;
determining a residual error signal based upon the windowed impulse response value; and
adjusting an equalizer coefficient by an amount that is based upon the residual error signal;
wherein determining a residual error signal further comprises determining the difference between the impulse response value and the windowed impulse response value.
2. The method of claim 1 wherein determining the residual error signal further comprises subtracting the windowed impulse response value from the impulse response value.
3. The method of claim 1 wherein adjusting an equalizer coefficient further comprises:
transforming the residual error signal to a frequency domain to generate a transformed residual error signal;
modifying the transformed residual error signal by an estimated channel response value to form an interim value;
transforming the interim value to a time domain to generate a correction factor; and
adjusting the equalizer coefficient by an amount proportional to the correction factor.
4. The method of claim 3 wherein the transforming the residual error signal to a frequency domain to generate a transformed residual error signal further comprises transforming the residual error signal via a Fast Fourier Transform.
5. The method of claim 3 wherein the transforming the interim value to a time domain to generate a correction factor further comprises transforming the interim value via an Inverse Fast Fourier Transform.
6. The method of claim 3 wherein the amount proportional to the correction factor is determined at least in part by multiplying the correction factor by an amount proportional to a step size.
7. A system for determining equalizer coefficients for channel equalization in Discrete-Multi-Tone based Digital Subscriber Line systems, the system comprising:
an impulse response module for determining a impulse response value in a time domain;
a windowing module for applying windowing to the impulse response value to determine a windowed equalized impulse response value;
a residual error module for determining a residual error signal based upon the windowed impulse response value; and
an equalizer coefficient module for adjusting an equalizer coefficient by an amount that is based upon the residual error signal;
wherein the residual error module further comprises a difference module for determining the difference between the impulse response value and the windowed impulse response value.
8. The system of claim 7 wherein the residual error module further comprises a subtraction module for subtracting the windowed impulse response value from the impulse response value.
9. The system of claim 7 wherein the equalizer coefficient module further comprises:
a first transformation module for transforming the residual error signal to a frequency domain to generate a transformed residual error signal;
a modification module for modifying the transformed residual error signal by an estimated channel response value to form an interim value;
a second transformation module for transforming the interim value to a time domain to generate a correction factor; and
wherein the equalizer coefficient module adjusts the equalizer coefficient by an amount proportional to the correction factor.
10. The system of claim 9 wherein the transforming the residual error signal to a frequency domain to generate a transformed residual error signal further comprises transforming the residual error signal via a Fast Fourier Transform.
11. The system of claim 9 wherein the transforming the interim value to a time domain to generate a correction factor further comprises transforming the interim value via an Inverse Fast Fourier Transform.
12. The system of claim 9 wherein the amount proportional to the correction factor is determined at least in part by multiplying the correction factor by an amount proportional to a step size.
13. A method for determining equalizer coefficients for channel equalization in Discrete-Multi-Tone based Digital Subscriber Line systems, the method comprising:
determining a impulse response value in a time domain;
applying windowing to the impulse response value to determine a windowed impulse response value;
determining a residual error signal based upon the windowed impulse response value; and
adjusting an equalizer coefficient by an amount that is based upon the residual error signal;
wherein adjusting an equalizer coefficient further comprises:
transforming the residual error signal to a frequency domain to generate a transformed residual error signal;
modifying the transformed residual error signal by an estimated channel response value to form an interim value;
transforming the interim value to a time domain to generate a correction factor; and adjusting the equalizer coefficient by an amount proportional to the correction factor.
14. The method of claim 13 wherein determining the residual error signal further comprises subtracting the windowed impulse response value from the impulse response value.
15. The method of claim 13 wherein the transforming the residual error signal to a frequency domain to generate a transformed residual error signal further comprises transforming the residual error signal via a Fast Fourier Transform.
16. The method of claim 13 wherein the transforming the interim value to a time domain to generate a correction factor further comprises transforming the interim value via an Inverse Fast Fourier Transform.
17. The method of claim 13 wherein the amount proportional to the correction factor is determined at least in part by multiplying the correction factor by an amount proportional to a step size.
18. A system for determining equalizer coefficients for channel equalization in Discrete-Multi-Tone based Digital Subscriber Line systems, the system comprising:
an impulse response module for determining a impulse response value in a time domain;
a windowing module for applying windowing to the impulse response value to determine a windowed equalized impulse response value;
a residual error module for determining a residual error signal based upon the windowed impulse response value; and
an equalizer coefficient module for adjusting an equalizer coefficient by an amount that is based upon the residual error signal;
wherein the equalizer coefficient module further comprises:
a first transformation module for transforming the residual error signal to a frequency domain to generate a transformed residual error signal;
a modification module for modifying the transformed residual error signal by an estimated channel response value to form an interim value;
a second transformation module for transforming the interim value to a time domain to generate a correction factor; and
wherein the equalizer coefficient module adjusts the equalizer coefficient by an amount proportional to the correction factor.
19. The system of claim 18 wherein the residual error module further comprises a subtraction module for subtracting the windowed impulse response value from the impulse response value.
20. The system of claim 18 wherein the transforming the residual error signal to a frequency domain to generate a transformed residual error signal further comprises transforming the residual error signal via a Fast Fourier Transform.
21. The system of claim 18 wherein the transforming the interim value to a time domain to generate a correction factor further comprises transforming the interim value via an Inverse Fast Fourier Transform.
22. The system of claim 18 wherein the amount proportional to the correction factor is determined at least in part by multiplying the correction factor by an amount proportional to a step size.