A digital quadrature demodulation system

Existing digital quadrature demodulation techniques sample the input at either 2B Hz or 4B Hz, select the even samples as the in-phase (I), and interpolate the odd samples to give the quadrature (Q), output. The signal bandwidth is B. We propose a demodulation system to produce I and Q samples at arbitrary sampling rate greater than 2B Hz. The system eliminates the IF downconversion step with a special sampling scheme. The even samples correspond to the I component, while the Q components are the filtered output. The filter can be a lowpass or least squares (LS) filter. The lowpass filter design is based on trade-offs between the filter length and the degree of oversampling. It produces similar results as previous work when the sampling rate is 2B Hz or 4B Hz. Unlike existing methods which assume the input is white, a LS filter, on the other hand, can make use of input signal characteristics to achieve a better result. The higher the correlation in the input the larger the improvement. The cost for LS filtering is a coefficient update step if the input is time varying. A scheme to cancel dc offset from analog to digital (A/D) converters is also given.