摘要:
Noise is removed from the digitized output of a sensor, subject to undesired resonance, even when the resonant frequency is unknown or drifts, with sufficiently low phase delay for the sensor to be used in closed-loop control. A very narrow notch filter which removes the resonance-induced noise is recursive (IIR) and therefore has a low phase delay. However, the apparatus which determines the center frequency of the notch filter is non-recursive, and therefore stable. It includes a tunable FIR filter which tracks the same resonance that we wish the IIR filter to remove. Tuning the FIR filter to minimize the output of the FIR filter therefore tunes the notch frequency to align with the resonant frequency. The tuning parameter which adaptively produces this result is suitably scaled and biased, and is applied to the IIR filter.
摘要:
Noise may be reduced or eliminated from a digital sawtooth signal representing the phase of a periodic signal. This may be done precisely, even when inexpensive fixed-point arithmetic is used. In one aspect of the invention, the input signal (noise plus true signal) 12 is filtered to produce, in succession: (a) mod one differentiated noise plus slope of true phase signal 28; (b) mod one differentiated noise plus slope of residual phase signal (true phase signal minus estimated slope of true phase signal) 36; (c) mod one differentiated noise 46; (d) estimated noise 62; and (e) smoothed phase signal 72. In a second aspect, a noisy phase signal 12 is extracted from a first arbitrary periodic signal and the above steps are used to generate a noise-reduced phase signal 72. The noise-reduced phase signal 72 is then used to generate a second arbitrary periodic signal of the same frequency.
摘要:
Noise is removed from the digitized output of a sensor, subject to undesired resonance, even when the resonant frequency is unknown or drifts, with sufficiently low phase delay for the sensor to be used in closed-loop control. A very narrow notch filter which removes the resonance-induced noise is recursive (IIR) and therefore has a low phase delay. However, the apparatus which determines the center frequency of the notch filter is non-recursive, and therefore stable. It includes a tunable FIR filter which tracks the same resonance that we wish the IIR filter to remove. Tuning the FIR filter to minimize the output of the FIR filter therefore tunes the notch frequency to align with the resonant frequency. The tuning parameter which adaptively produces this result is suitably scaled and biased, and is applied to the IIR filter.
摘要:
An apparatus for controlling an amplitude of a signal generated from a digitized sinusoid of rapidly and widely varying amplitude is described herein. The apparatus includes a two stage gain adjuster which produces a gain adjusted signal, a phase shifter which converts the gain adjusted signal into two gain adjusted output signals separated in phase by 90 degrees, a power estimation unit to estimate the power of the gain adjusted signal, and an adjusting unit to adjust a gain of the gain adjuster according to a power estimate from the power estimation unit and a desired output signal power.
摘要:
A system and method for a directional microphone system is disclosed. The directional microphone system can adaptively track and detect sources of sound information, and can reduce background noise. A first monolithic detection unit for detecting sound information and performing local signal processing on the detected sound information is provided. In the detection unit, an integrated transducer is provided for receiving acoustic waves and for generating sound information representative of the waves. A processor is coupled to the transducer for receiving the sound information and for performing local digital signal processing on the sound information to generate locally processed sound information. A base unit is coupled to the first monolithic detection unit and includes a global processor which receives the locally processed sound information and performs global digital signal processing on the locally processed sound information to generate globally processed sound information.
摘要:
This detector provides a computationally simple digital low power detector of symbol rate, also called baud rate. It uses an approximate Hilbert transform function to create approximate in-phase and quadrature signals. An approximate envelope detector (feature extractor) processes these signals to produce a signal with a strong frequency component at the symbol rate. This signal is then filtered, accumulated, and threshold detected. The approximate in-phase and quadrature signals are formed by a linear sequence of six delay elements, the output of the third delay element being the in-phase signal. A first summer receives the output of the second delay element at a minus input and the output of the fourth delay element at a plus input. A second summer receives the signal input at a minus input and the output of the sixth delay element at a plus input, and drives a right two bit shifter. A third summer receives the output of the right two bit shifter and the output of the first summer and drives both a right one bit shifter and a right three bit shifter, the outputs of which are summed to form the quadrature signal.
摘要:
For use with a quartz angular rate sensor, a frequency and phase-locked synthesizer recovers a reference signal virtually free of phase noise, and generates a quadrature-phase reference signal for complex demodulation of the angular rate signal. The synthesizer also ensures a precisely adjusted phase shift of approximately zero across the drive tines of the sensor. Moreover, the digital synthesizer provides a precise numerical indication of the drive frequency, which can be used for compensation and automatic tuning of filters, such as a tracking filter, a filter in an automatic gain control, and notch filters in the phase and/or frequency detectors in the digital synthesizer. The tracking filter is used as a pre-filter for the synthesizer, and is responsive to a passband-width control signal generated from the magnitude of the frequency and phase error signal controlling the frequency generated by the synthesizer. Preferably the synthesizer has an oscillator controller for producing a pair of frequency control signals that are the sine and cosine of a frequency control parameter (.phi.), and one of these control signals is generated from the other by a polynomial approximation. To compensate for roundoff error, when one of the in-phase or quadrature-phase outputs has a magnitude less than a limit value, a compensated value for the other output is computed from an even polynomial of the magnitude.
摘要:
A MOVING VEHICLE CLASSIFIER WITH ITERATIVE DECONVOLVER ESTIMATOR deconvolves a signal in which a plurality of sensed signals, which originate from a common signal source (namely, a moving vehicle) and traverse differing signal paths, are used to formulate an initial unconstrained estimation of the signal source as a reference signal. Preselected constraints are placed on the reference signal to generate an estimate of the source signal, and of a source signal matrix, for the plurality of input sensors. A pseudoinverse of the source matrix is then used to form an estimate of the impulse response of the propagation paths for the signals. The estimates of signal path impulse response are used in combination with the input signals to again estimate the unconstrained reference signal. This is in turn used to estimate a new source signal matrix and path responses. These estimation steps are repeated in a series of iterative steps until a point of minimum variance from the received signals is reached. At this point, the source signal estimation is chosen as a deconvolved signal output. The moving vehicle which produced the source signal may then be classified by conventional means.
摘要:
An apparatus and method are disclosed for classifying a signal among a plurality of potential signal sources or events in which a signal embodying values for predefined source features is received and conditional likelihoods are determined for the correspondence of each feature value with each of the plurality of events. The individual conditional likelihood values for each feature which corresponds to a common event are multiplied together to form an overall conditional likelihood that any event in question is the source of the signal. The total likelihood products are then sorted into maximum and minimum values, with the maximum likelihood forming a tentative output designation. The ranked values are then examined by a decision logic element to determine if they satisfy certain minimum absolute and relative value threshold limits to verify if the maximum or next higher level likelihood values designate proper output choices. Depending on the results of the threshold tests, one, two, or no signal sources or events are chosen for a signal classification output. At the same time, a conditional probability for the chosen signal is used to indicate a confidence level in the selected choice.
摘要:
To demonstrate a signal, the signal is sampled and each sample of the signal is multiplied by the sine and the cosine of a phase angle indicating when the sample was taken from the signal. The sinusoidal signal for producing an in-phase demodulated signal is Chebychev-approximation derived and computed as a selected even or odd polynomial, depending on whether the phase angle falls within one of a plurality of angular ranges. So that the error in the synthetic sinusoid is minimax and so that the even and odd polynomials have similar computational complexity, the angular ranges for the even polynomial exceed the angular ranges for the odd polynomial. Preferably, the sinusoid for producing a quadrature-phase demodulated signal is computed as a differential of the sinusoid spliced from the odd and even polynomials. Therefore the quadrature-phase demodulated signal can be provided with a minimal increase in computational complexity. The demodulation method permits a signal to be demodulated by a reference frequency signal to produce a demodulated digital signal that is synchronized to a system clock. The system clock need not be synchronized to the reference frequency signal. The demodulation method is computationally efficient, and permits a digital signal processor to be programmed for demodulating an angular rate signal from a quartz angular rate sensor vibrating at about 10 kilohertz.