Abstract: A power receiver is provided herein. The power receiver provides in-band communication with a power transmitter. The power receiver recognizes a change in frequency that identifies a training sequence and determines an impulse response with respect to the training sequence. The power receiver also cancels an effect of the impulse response during the in-band communication.

Abstract: A signal processing device includes a bit-pattern output unit and a look-up table storage unit which are configured as follows: The bit-pattern output unit is provided for receiving input 1-bit digital signals generated by ?? modification and aligning bits of the input 1-bit digital signals in a chronological order to output parallel bit pattern. The look-up table storage unit is provided for storing a look-up table that represents a relationship between the bit patterns output from the bit pattern output unit and resulting values of a filtering arithmetic operation on the basis of the bit patterns. In the signal processing device, the bit patterns output from the bit-pattern output unit are provided as indexes. The indexes are referenced to output the resulting values of the filtering arithmetic operation corresponding to the bit patterns listed in the look-up table stored in the look-up table storage unit.

Abstract: This invention is an improvement upon the basic adaptive signal processors, the Multi-Stage Wiener Filter and the cascaded canceller. The invention combines the concepts of soft weighting the adaptive weights of either type of processor disclosed herein with reiteratively processing the outputs by returning them to the input of the chosen adaptive signal processor. The combination of these functions improves the Signal to Interference plus Noise Ratio (SINR), the Probability of Detection (Pd), and/or the Bit Error Rate (BER). The invention improves statistical convergence of these types of metrics such that fewer training data samples are needed to achieve a particular satisfactory value of these metrics than would occur using traditional computational adaptive signal processors.

Abstract: Equalisation of a communication channel is achieved through use of a Wiener filter frequency response mechanism that operates to transform at least a portion of a data stream generated from a plurality of space time coded (STC) symbol streams received from a plurality of transmit antenna elements into a packet spectrum. A training sequence for a channel through which the symbol streams have been sent is also transformed to a channel impulse response spectrum in order to assess the channel impulse response for the channel. The packet spectrum is equalised with the channel impulse response spectrum to produce an equalised packet spectrum in the transform domain. This is then converted into a time domain equalised data stream for recovery of originally transmitted information.

Abstract: Limit-cycle oscillations are caused by the compounding of quantization errors that occurs when previous digital filter outputs are used as inputs to the digital filter for the current operation. Where a signal in a digital waveform has become a constant common value applied to the input of the digital filter (indicative that the digital waveform has suspended conveyance of data), limit-cycle oscillations often appear as “random” outputs, with values different from the common value, that occur long after the signal in the digital waveform has become the constant common value. Limit-cycle oscillations are manifested as noise in the filtered digital waveform. Such noise hampers the ability of the system to extract the signal from the filtered digital waveform. The present invention identifies the occurrence of a limit-cycle oscillation as an output different from the common value. The identified limit-cycle oscillation is set equal to the common value.

Abstract: The present invention relates to an interpolation method and an interpolation apparatus for a digital audio signal or a digital image signal that has a predetermined sampling cycle and quantization bit length, and in particular, to an interpolation method and an interpolation apparatus that can effectively reduce quantization noise in a digital signal that is obtained by information compression. In the method and apparatus according to the present invention, it is made to perform interpolation processing of signal levels in a interpolation object interval in a given digital signal in accordance with a predetermined function curve, which monotonously changes, with the interpolation object interval including a discontinuous part that exists between one signal interval, where the same gradation levels continue, and another signal interval, which is adjacent to the one signal interval and in which the same gradation levels that are different continue.

Abstract: In a signal processing integrated circuit having an analog to digital converter and a digital filter having a plurality of taps separated in time, when starting a conversion after a reset or a change of input channel, the filter will have an incomplete set of input data as the delayed inputs to an output calculation are all zero from the reset operation. After reset, during the time that data are filling up the filter pipeline, the calculation of an output value will give a result that holds information about the input, but does not present the data with the same scaling and frequency content as the fully settled filter. The integrated circuit selectively provides two modes, on that provides only fully settled data from the filter or and another that provides all data from the filter, including unsettled data. Knowledge about the filter coefficients can be utilized by a user or user process to extract information about the input from the unsettled data.

Abstract: Initialization of an adaptive equalizer is controlled by varying a gain applied to an input signal. The input signal may be an alternate mark inversion signal. The gain is applied to the input signal to form an amplified signal. An autocorrelation value of the amplified signal is calculated. The autocorrelation value is compared to a predetermined constant. The gain is adjusted, based on which of the group consisting of the autocorrelation value and the predetermined constant is greater. The gain application, autocorrelation calculation, comparison and gain adjustment are repeated, until the autocorrelation value is sufficiently close to the predetermined constant to satisfy a convergence criterion. The amplified signal is then filtered.

Abstract: A method for an adaptive equalization apparatus in a multiple-link hopping radio system includes hopping among a plurality of radio links to receive variable-length bursts of radio signals on the plurality of radio links and equalizing amplitude and phase variations of a slow channel for each radio link from a received burst on the radio link. Further, the method includes storing the estimated tap coefficients pertinent to each radio link and using the tap weights of the current burst of the radio link to reliably pre-compensate the channel amplitude and phase distortion of a next received burst on the radio link.

Abstract: In the Global System for Mobile Communication (GSM), an improvement in estimating the channel impulse response (CIR) is obtained by starting up two channel estimators using the midamble and obtaining channel estimations with split burst. The two estimations are then cross-correlated to obtain the estimated CIR.

Abstract: A method and apparatus for reducing convergence time in a digital filter. When the digital filter is initially run, the coefficients in the digital filter are adjusted to reduce error in the output of the digital filter. When the adjusted coefficients meet a selected error level, these coefficients are stored in a memory and the digital filter filters data. The next time the digital filter is run, the stored coefficients are loaded into the digital filter and a number of iterations are run in which the coefficients are adjusted. Then, a determination is made as to whether the error level meets a threshold that may be the same as the selected error level. If the coefficients meet the threshold, the coefficients are stored in the memory and the filter is then used to filter data.

Abstract: Equalizer training is validated by techniques to detect false training. The validation techniques involve checking the unique time domain energy shaping of a plurality of taps of a trained equalizer of a modem for good training. The energy measure for each equalizer tap of the plurality of taps is approximated by summing the magnitude of a real component and a magnitude of an imaginary component of an equalizer coefficient corresponding to the equalizer tap. A tap with a highest approximate energy (i.e., the main tap) is then determined. Next, the main tap energy is compared to a main tap threshold and a predetermined guarding threshold. The plurality of taps are arranged in a time index ordered sequence. In determining the main tap threshold, a first energy sum of a beginning set of taps in the sequence is compared with a second energy sum of a last set of taps in the sequence. The lesser tap energy sum between the first energy sum and the second energy sum is the main tap threshold.

Abstract: A method of calculating TEQ and FEQ coefficients in a discrete multi-tone (DMT) system includes determining an estimated channel impulse response. A frame delay is determined based upon the estimated channel impulse response. Using these parameters and a known training sequence, the method generates the TEQ coefficients. The method then generates the FEQ coefficients based upon the TEQ coefficients and a version of the estimated channel impulse response.