In a parallel time-interleaved data sampling system, timing and amplitude mismatches of this structure degrade the performance of the whole ADC system. In this paper, an adaptive blind synthesis calibration algorithm is proposed, which could estimate the timing, gain and offset errors simultaneously, and calibrate automatically. With no need of an extra calibration signal and redesign, it could efficiently and dynamically track the changes of mismatches due to aging or temperature variation. A fractional delay filter is developed to adjust the timing mismatch, which simplifies the design and decreases the cost. Computer simulations are also included to demonstrate the effectiveness of the proposed method.
In this paper the capacity of non-uniform sampling rate conversion techniques, involving different interpolation methods, aimed at wow defect reduction, is examined. Involved are: linear interpolation, four polynomial-based interpolation methods and the windowed sincbased method. The examined polynomial methods are: Lagrange interpolation, polynomial fitting with additional noise reduction, Hermitan and Spline. The performance of an artificially distorted audio signal, restored using non-uniform resampling, is evaluated on the basis of standard audio defect measurement criteria and compared for all of the aforementioned interpolation methods. The chosen defect descriptors are: total harmonic distortion, total harmonic distortion plus noise and signal to noise ratio.