Abstract: A canceler for decorrelating one set of signal samples called the main set rom another set called the auxiliary set taken from the same domain. It frequency-translates the signals on their intermediate frequency (i.f.) carrier down close to DC using a local oscillator outside one band edge of the signal and samples the resultant signal at a rate equal to twice the information bandwidth or greater and permits using real rather than complex samples. These samples are then converted to digital information in analog-to-digital converters and are processed digitally at a clock rate equal to twice the information bandwidth or greater. The main signal is processed as above but the auxiliary signal is translated down using the same local oscillator signal and, with the same local oscillator signal phase shifted 90.degree. to obtain an inphase and quadrature signal on the same carrier frequency prior to analog-to-digital conversion.

Abstract: A pulse compression system for use with step approximation to linear FM and rank coded signals to eliminate sampling errors and range time grating lobes while providing large pulse compression ratios comprising: a receiving circuit for receiving echo signals, a converting circuit for converting echo signals from the receiver to I and Q baseband signals without clock sampling, a sliding window discrete Fourier Transform or fast Fourier Transform (FFT) circuit including a taped delay line and a plurality of resistor-type phase weighting networks and adders for generating a plurality of output signals representing the different frequency steps in the signals, a delay circuit for differentially delaying the output frequency steps from the sliding window DFT circuit so the the output steps occur simultaneously, and a summer for adding the differentially delayed outputs to yield a short pulse with a peak amplitude when a coded echo pulse is correctly indexed within the delay line of the DFT circuit.

Abstract: A pulse compression system based on the transmission of a step approximat to linear FM pulse, and the subsequent processing of echos therefrom by converting the echo pulses to I and Q baseband signals and then sampling these I and Q signals at the Nyquist sampling rate. It has been discovered that this processing yields a signal with a spectrum substantially identical to the spectrum of a properly sampled Frank coded pulse. These sampled I and Q signals are then compressed by a Fast Fourier Transform compressor to yield a pulse with low sidelobes.

Abstract: A digital pulse expander-compressor for use in pulse compression radars and having the advantage of precompression bandwidth tolerance. The pulse expander-compressor exploys a discrete Fourier transform circuit and multi-stage delay line feeding inputs x(n) to the discrete Fourier transform circuit to generate outputs in accordance with the formula ##EQU1## where n is the sequence number of the clock pulse; N is the number of delay stages plus one; and k is the number of the output subpulse from the transform circuit. An arrangement of delay stages differentially delays the output subpulses from the discrete Fourier transform circuit, and a coherent summer adds the real and imaginary parts of the signals from the delay lines. The delay stages delay the subpulse s(k), O.ltoreq.k.ltoreq.N/2-1, by ##EQU2## clock pulse intervals, while delaying the subpulse s(k), N/2+1.ltoreq.k.ltoreq.N-1, by ##EQU3## clock pulse intervals. The subpulse s(k), k=N/2, is not used.

Abstract: A digital pulse compression processor for reducing the processing loss in rget-echo signals caused by sampling time errors comprising a first processing channel for sampling the echo signal in accordance with pulses from a first clock signal and a second processing channel for sampling the echo signal in accordance with pulses from a second clock signal at the same sampling frequency as the first clock signal but with its pulses interlaced, in time, approximately midway between the pulses of the first clock signal to ensure that the largest sampling error will be one-quarter of a sampling period or less. The sampled signals in each channel are then digitized and compressed, and then processed to form the compressed signal envelope. The resultant signal envelopes from each channel are multiplied together to form a low-sidelobe narrow output pulse.

Abstract: A pulse compression decoder which oversamples by two the signal elements of received phase-coded signal, performs a pairwise average on the signal elements at intervals equal to 1/2 the width of a signal element, reverse-codes the pairwise averages, combines the pairwise averages to form a sub-accumulation signal, and performs a pairwise average on the sub-accumulation signals at intervals equal to 1/2 the width of a signal element to produce a compressed pulse having a high peak-to-side lobe ratio.

Abstract: The invention is a configuration of digital open-loop cancelers, each of ch uses a batch window sampling technique, for decorrelating a main input signal from a plurality of auxiliary input signals by using one or more iterations of cancellation. The main signal includes a desirable signal and undesirable interference signals. The auxiliary signals include the undesirable interference signals which are correlated with the interference of the main signal. Samples of the main signal and a first auxiliary signal are fed to a first digital canceler. The canceler batches them, (forms the samples in a group) and measures the correlation between the main signal and auxiliary signal. This measurement, or weight, is applied to each of the samples of the auxiliary signal which were used to establish the weight. The weighted auxiliary samples are subtracted from the main input signal to produce an output residue signal that is uncorrelated with the auxiliary signal.

Abstract: A digital open-loop canceller is used to decorrelate one signal from anot in such applications as coherent sidelobe cancellers and moving target indicators. The invention measures the correlation between a complex main input signal and a complex auxiliary input signal in digital form. It divides the correlation coefficient by the average value of the auxiliary signal magnitude squared, multiplies the ratio by the auxiliary signal and subtracts the result from the main input signal to cancel correlated components.