Abstract: A channel estimation method and a channel estimation system based on time-domain training sequence are provided. The channel estimation system first acquires an initial channel vector used for channel estimation of a current frame, and calculates an algorithm initial vector based on a training sequence vector contained in a received receiving signal vector and the receiving signal vector, then performs estimation based on the initial channel vector and the algorithm initial vector and using a conjugate gradient method based on a preprocessing matrix to acquire each channel estimation value, and eventually perform noise reduction processing on each channel estimation value based on a first predetermined threshold value to acquire a final channel estimation value. Compared with the existing channel estimation methods, the present invention has a shorter delay and lower calculation complexity, and thus can be easily implemented.

Abstract: The present invention relates to a method and system for channel estimation. First, pilot signals are extracted from a received multi-path signal, in which each pilot signal includes a first pilot and a second pilot. Then, an initially estimated channel frequency-domain response is obtained based on the extracted first pilot. Afterward, a frequency-domain response estimate of each pilot frequency in the second pilot is calculated according to the obtained initially estimated channel frequency-domain response, an actual value of each pilot frequency in the second pilot is obtained based on the extracted second pilot, and a deviation between the frequency-domain response estimate and the actual value of each pilot frequency is calculated. When it is detected that aliasing components exist in the initially estimated channel frequency-domain response, a center of each aliasing component is determined according to the deviation, so as to determine an estimated position of each aliasing component.

Abstract: Data rate conversion devices and methods are provided. A method for converting a first digital signal having a first sampling rate into a second digital signal having a sampling rate close to a predetermined second sampling rate comprises the following operations: when the ratio of the first sampling rate to the second sampling rate is a repeating infinite decimal, calculate at least two calibrating coefficient values and output the calibrating coefficient values according to a predetermined rule; conduct overflow operation on the output calibrating coefficient; and interpolate the first digital signal using the output calibrating coefficient and the result of the overflow operation to obtain the second digital signal such that during any period of a certain length along time axis, sampling times of the second digital signal equals to sampling times of the second sampling rate.

Abstract: The present invention relates to a blind adaptive filter for narrowband interference cancellation, which includes an adaptive filter, a delay unit coupled to the adaptive filter for generating a delayed signal with a predetermined delay length from the output signal of the adaptive filter, and an error calculation unit coupled to the adaptive filter and the delay unit. The error calculation unit compares the output signal from the adaptive filter and the delayed signal from the delay unit to extract error information, and feedback the first error information to the adaptive filter. The first error information is formed of a transfer function including a number of coefficients, and used to adjust the adaptive filter and remove interference in the next input signal. The disclosed technique is also applicable in wideband receivers, as well as resisting multiple strong narrowband interferences having a frequency sweep rate of tens of milliseconds.

Abstract: The present invention relates to a method and system for channel estimation. First, pilot signals are extracted from a received multi-path signal, in which each pilot signal includes a first pilot and a second pilot. Then, an initially estimated channel frequency-domain response is obtained based on the extracted first pilot. Afterward, a frequency-domain response estimate of each pilot frequency in the second pilot is calculated according to the obtained initially estimated channel frequency-domain response, an actual value of each pilot frequency in the second pilot is obtained based on the extracted second pilot, and a deviation between the frequency-domain response estimate and the actual value of each pilot frequency is calculated. When it is detected that aliasing components exist in the initially estimated channel frequency-domain response, a center of each aliasing component is determined according to the deviation, so as to determine an estimated position of each aliasing component.

Abstract: A channel estimation method and a channel estimation system based on time-domain training sequence are provided. The channel estimation system first acquires an initial channel vector used for channel estimation of a current frame, and calculates an algorithm initial vector based on a training sequence vector contained in a received receiving signal vector and the receiving signal vector, then performs estimation based on the initial channel vector and the algorithm initial vector and using a conjugate gradient method based on a preprocessing matrix to acquire each channel estimation value, and eventually perform noise reduction processing on each channel estimation value based on a first predetermined threshold value to acquire a final channel estimation value. Compared with the existing channel estimation methods, the present invention has a shorter delay and lower calculation complexity, and thus can be easily implemented.

Abstract: A method and system for canceling an interference signal are provided. In the method, a received signal is transformed from a time domain to a frequency domain at relative low precision to obtain an estimated frequency and an estimated amplitude of the interference signal. A reference signal is generated using the estimated frequency and the estimated amplitude. An error signal is generated using the received signal and the reference signal based on minimum correlation criteria. The reference signal is adjusted using the error signal based on an adaptive algorithm. The reference signal is subtracted from the received signal to obtain the useful signal.

Abstract: A method and apparatus for measuring parameters of a receiver having a mixer for generating an I signal and a Q signal using an input signal, an I channel circuit for processing the I signal, and a Q channel circuit for processing the Q signal. The method includes feeding the receiver a first testing signal before the mixer. The method includes feeding the receiver a second testing signal on the I channel circuit. The method includes feeding the receiver a third testing signal on the Q channel circuit. The method includes measuring I/Q quadrature deviation and I/Q delay imbalance of the receiver using the first, the second, and the third testing signals. This separates the I/Q quadrature deviation and I/Q delay imbalance.

Abstract: The present invention relates to a blind adaptive filter for narrowband interference cancellation, which includes an adaptive filter, a delay unit coupled to the adaptive filter for generating a delayed signal with a predetermined delay length from the output signal of the adaptive filter, and an error calculation unit coupled to the adaptive filter and the delay unit. The error calculation unit compares the output signal from the adaptive filter and the delayed signal from the delay unit to extract error information, and feedback the first error information to the adaptive filter. The first error information is formed of a transfer function including a number of coefficients, and used to adjust the adaptive filter and remove interference in the next input signal. The disclosed technique is also applicable in wideband receivers, as well as resisting multiple strong narrowband interferences having a frequency sweep rate of tens of milliseconds.

Abstract: Data rate conversion devices and methods are provided. A method for converting a first digital signal having a first sampling rate into a second digital signal having a sampling rate close to a predetermined second sampling rate comprises the following operations: when the ratio of the first sampling rate to the second sampling rate is a repeating infinite decimal, calculate at least two calibrating coefficient values and output the calibrating coefficient values according to a predetermined rule; conduct overflow operation on the output calibrating coefficient; and interpolate the first digital signal using the output calibrating coefficient and the result of the overflow operation to obtain the second digital signal such that during any period of a certain length along time axis, sampling times of the second digital signal equals to sampling times of the second sampling rate.

Abstract: A method and apparatus for measuring parameters of a receiver having a mixer for generating an I signal and a Q signal using an input signal, an I channel circuit for processing the I signal, and a Q channel circuit for processing the Q signal. The method includes feeding the receiver a first testing signal before the mixer. The method includes feeding the receiver a second testing signal on the I channel circuit. The method includes feeding the receiver a third testing signal on the Q channel circuit. The method includes measuring I/Q quadrature deviation and I/Q delay imbalance of the receiver using the first, the second, and the third testing signals. This separates the I/Q quadrature deviation and I/Q delay imbalance.

Abstract: A method and system for canceling an interference signal are provided. In the method, a received signal is transformed from a time domain to a frequency domain at relative low precision to obtain an estimated frequency and an estimated amplitude of the interference signal. A reference signal is generated using the estimated frequency and the estimated amplitude. An error signal is generated using the received signal and the reference signal based on minimum correlation criteria. The reference signal is adjusted using the error signal based on an adaptive algorithm. The reference signal is subtracted from the received signal to obtain the useful signal.