Abstract: To more efficiently compensate for modulation imbalance, a mobile device modulates and demodulates a calibration tone and generates digital data representing modulation imbalance effects on the calibration tone. Using digital data enables modulation imbalance effects to be quickly estimated using digital signal processing techniques, increasing the number of estimates of modulation imbalance effects calculated. The modulation imbalance estimates are used to refine one or more compensation parameters which are applied to transmitted and/or received signals to compensate for errors caused by modulation.

Abstract: Two dimensional interpolation techniques are used to compensate for time-varying channel gain H in an OFDM system. The channel gain can be reasonably estimated at various times and at various frequencies due, for example, to the use of pilot tones. These channel estimates are used to estimate the channel gain at other times and/or frequencies by two-dimensional interpolation, interpolating first with respect to time (e.g., with respect to a symbol index s) and then with respect to frequency (e.g., with respect to a sub-carrier index c).

Abstract: To more efficiently compensate for modulation imbalance, a mobile device modulates and demodulates a calibration tone and generates digital data representing modulation imbalance effects on the calibration tone. Using digital data enables modulation imbalance effects to be quickly estimated using digital signal processing techniques, increasing the number of estimates of modulation imbalance effects calculated. The modulation imbalance estimates are used to refine one or more compensation parameters which are applied to transmitted and/or received signals to compensate for errors caused by modulation.

Abstract: Architectures of multi-stage sample rate converters are disclosed. According to one aspect of the present invention, a received signal with a higher sampling rate is converted to a lower sampling rate. To prevent aliasing in the resultant signal, an anti-aliasing filter is introduced. The passband of the anti-aliasing filter is so adjusted according to the conversation rate of a sample rate converter. To keep the implementation relatively simple, the coefficients of the filter are kept constant. Therefore, the conversation rate of a sample rate converter is constrained in a limited range, thus requiring only a constant anti-aliasing filter. A series of halfband filters are then used to convert the signal to a desired sampling rate.