摘要:
Stable adaptive filter and method are disclosed. The invention solves a problem of instability associated with Fast Affine Projection adaptive filters caused by error accumulation in an inversion process of an auto-correlation matrix. The Stable FAP provides updating of the adaptive filter coefficients by solving at least one system of linear equations, whose coefficients are the auto-correlation matrix coefficients, by using one of the descending iterative methods having an inherent stability of operation due to intrinsic feedback adjustment. The results of the solution are used to update the filter coefficients. The above approach is applicable for any value of a normalized step size ranging from zero to unity. It allows either direct determining of an updating part of the filter coefficients without determining an inverse auto-correlation matrix, or determining the inverse auto-correlation matrix by descending iterative methods.
摘要:
An adaptive Infinite Impulse Response (IIR) filter is provided that can adaptively detect the presence of one or more tones in its input stream. The tones to be detected may be of arbitrary frequency, subject only to a limitation that such tones fall within a frequency band consistent with accepted sampling principles ( e.g. , a maximum frequency of interest no greater than one-half the sampling frequency -- Nyquist sampling criteria). An IIR filter developed according to the method of the invention will adaptively locate the frequencies of tones to be detected, thereby allowing for frequency drift from nominal expected frequency values with no loss in accuracy. Such a filter will also process the input signal sample-by-sample, thereby avoiding the blocking problem of FFT-based filter approaches. With the filter of the invention, an application can identify the frequencies, associated power levels, SNR and duration of the tones. Thus, such an application can use a simple user specified library of tone parameters to decide if tones of interest are present in the input stream.
摘要:
Circuitry and concomitant methodology for demodulating Direct-Sequence, Spread-Spectrum Code-Division Multiple-Access (DS/SS CDMA) channel signal using multiple samples per transmitted symbol and a minimum mean squared error criterion to suppress interference. In one embodiment, a bank of cyclically shifted filters (502) determined with reference to the conventional matched filter (401) for CDMA is used to demodulate the channel signal. In another embodiment, a bank of sub-filters (601, 602) determined with reference to the conventional matched filter (401) for CDMA is employed to demodulate the channel signal. In yet another embodiment, the output of a conventional matched filter is oversampled to demodulate the channel signal. Each embodiment utilizes a set of adaptive coefficients selected to minimize the mean square error between the transmitted symbol and detected symbol.
摘要:
A method and apparatus for separating and removing distortion from interfering co-channel singals and suppressing adjacent-channel interfering signals of the Gaussian Minimum-Shift Keyed (GMSK) or other MSK type with filtering structures that exploit the cyclostationarity of the received GMSK or other MSK signals in order to accommodate a greater number (or the same number, but with greater quality) of transmitted signals received by one or more antennas than can be accommodated by existing filters. The parameters in these filtering structures are adapted by either of two adaptation apparatus that exploit both the known training sequence that is transmitted in most wireless communications systems, and the constant modulus property exhibited by each of the transmitted GMSK or other MSK signals.
摘要:
In a video conference facility, an echo cancelling device is provided which comprises an adaptive finite impulse response filter (38) operable to sample the input signal to a loudspeaker (30) and to model, on the basis of that signal, the signal fed back to a microphone (32). The echo cancelling device further comprises a combiner (36) for subtracting the model feedback signal from the microphone output signal to provide a corrected microphone output signal, and a microprocessor (40) programmed to read the corrected and uncorrected microphone output signal and to compute updates to the weights of the filter (38), the computation including a multiplication by a variable scaling factor which varies in accordance with the ratio of a first value indicative of the long-term average power of the sound being fed back to a second value indicative of the short-term average power of the sound being fed back. In this way, undesirable fluctuations in the modelled output signal which cause unnatural sounding echoes at the other video conference site are reduced.
摘要:
A decision feedback equalizer is suitable for use with a bipolar return-to-zero receiver. The equalizer determines an output Y(n) (160) based on a compensated received value X(n) (104) and a correction factor, D(n) (141). After receiving X(n), the equalizer then forms an equalized received value X'(n) (107) based on combining X(n) with D(n). The equalizer then determines the output value Y(n) based on comparing X'(n) with a positive threshold, V1 and a negative threshold, V2. When Y(n) is determined to be zero, the equalizer adjusts the stored correction value D(n) by a predetermined value, DELTA , based on whether X'(n) is positive or negative.
摘要:
A digital communication system and corresponding feedback equalization method are described for use in reducing multi-path distortions, such distortions being diminished by employment of an adaptive conjugate gradient algorithm wherein weight coefficients of feed-forward and feedback filter taps are updated recursively during transmission of each data symbol. Weight coefficients are updated based on converging transmission error values, the weight coefficients being updated until either a predetermined number of updates are made, or until the current transmission error value is less than or equal to a predetermined percentage of the initial transmission error value for the given symbol.
摘要:
An extended coefficient resolution range is provided in an adaptive filter arrangement by utilizing several filters, in combination, to represent the extended resolution range. The coefficients of a first filter represents a high portion of the extended range, while the coefficients of the second filter represent low portions of the extended range. The two filters operate in parallel to filter an input signal, and the outputs of the filters are combined to produce the filtered signal. A low portion of the coefficients of the first filter overlap an upper portion of the coefficients of the second filter. The coefficients of the first filter are periodically updated from the overlapping portions of the coefficients of the second filter, after which the overlapping portions of the coefficients of the second filter are reset. In accordance with another aspect of the invention, the amount of overlap may be adjusted to dynamically change the resolution range of the filter. The coefficients of the first and second filters are capable of variably overlapping to expand or contract the resolution range in response to changes in the system being monitored. The degree of overlap can also be adjusted to provide an appropriate resolution range corresponding to when the system is stable. An adaptive process is used to up date filter coefficients.
摘要:
A measure of a degree of convergence in an adaptive filter arrangement is derived from the comparison of an amount of adaptation occurring in the adaptive filter arrangement, over a predetermined period of time, with a normalizing value accumulated for the same period. Supplemental signal processing may be invoked, modified or withdrawn based upon the degree of convergence indicated.