Showing papers on "Band-stop filter published in 1996"

Unbalance compensation using generalized notch filters in the multivariable feedback of magnetic bearings

Abstract: Unbalance of magnetically levitated rotors causes undesirable synchronous vibrations which may lead to saturation of the magnetic actuator. To avoid this problem we propose a generalized narrow-band notch filter which is inserted into the multivariable feedback without destabilizing the closed loop. The parameters of the generalized notch filter strongly depend on the inverse sensitivity matrix evaluated at rotational speed. This matrix is easily measured a priori and stored in a look-up table. It is shown that a decentralized notch is feasible for weakly gyroscopic rotors. The proposed notch filter approach has advantages in terms of run-time complexity and analytical verification of closed-loop stability. Results from the implementation of the proposed unbalance compensation in industrial magnetic bearing systems are included.

Optimal Surface Smoothing as Filter Design

Abstract: Smooth surfaces are approximated by polyhedral surfaces for a number of computational purposes. An inherent problem of these approximation algorithms is that the resulting polyhedral surfaces appear faceted. Within a recently introduced signal processing approach to solving this problem [7, 8], surface smoothing corresponds to low-pass filtering. In this paper we look at the filter design problem in more detail. We analyze the stability properties of the low-pass filter described in [7, 8], and show how to minimize its running time. We show that most classical techniques used to design finite impulse response (FIR) digital filters can also be used to design significantly faster surface smoothing filters. Finally, we describe an algorithm to estimate the power spectrum of a signal, and use it to evaluate the performance of the different filter design techniques described in the paper.

A comparison of adaptive and nonadaptive filters for reduction of power line interference in the ECG

Abstract: We have investigated the relative performance of an adaptive and nonadaptive 60-Hz notch filter applied to an ECG signal. We evaluated the performance of the two implementations with respect to adaptation rate (or transient response time), signal distortion, and implementation complexity. We also investigated the relative effect of adaptive and nonadaptive 60-Hz filtering on ECG data compression. With a 360 Hz sample rate and an adaptation time of approximately 0.3 s for a 1 mV 60-Hz signal, the adaptive implementation is less complex and introduces less noise, particularly in the ST-segment, into a typical ECG signal. When applied to ECG signals, prior to data compression by average beat subtraction and residual differencing, the residual signal resulting from the adaptively filtered signal had an average entropy 0.31 bits per sample (bps) lower than the unfiltered signal. The nonadaptive 60-Hz filter produced an average entropy decrease of 0.08 bps relative to the unfiltered ECG.

System for adaptively filtering audio signals to enhance speech intelligibility in noisy environmental conditions

Abstract: A method and system are provided for adaptively reducing noise in frames of digitized audio signals that include both speech and background noise. Frames of digitized audio signals are passed through an adjustable, high-pass filter circuit to filter a portion of background noise located in a low frequency range of the digitized signal. The filter circuit is adjusted by a filter control circuit adapted for a current frame to exhibit a selected frequency response curve. The filter control circuit includes a speech detector for detecting the presence or absence of speech in the frames of digitized audio signals. The filter circuit is adjusted when no speech is detected in the current frame. In a first preferred embodiment, the filter control circuit controls the filter circuit by calculating a noise estimate corresponding to the background noise, and adjusting the filter circuit based on the noise estimate. As the noise estimates increase, the filter circuit is adjusted to extract increasing amounts of energy falling in low frequency ranges of speech. In a second preferred embodiment, the filter circuit is adjusted as a function of a noise profile estimate. A noise profile estimate for a current frame is determined as a function of speech detection and is compared to a reference noise profile. Based on this comparison, the filter circuit is adaptively adjusted.

Method and apparatus for adaptive line enhancement in coriolis mass flow meter measurement

Abstract: An apparatus and method for determining frequency and phase relationships of vibrating flow tubes in a Coriolis mass flow meter. Adaptive line enhancement (ALE) techniques and apparatus are used in a digital signal processing (DSP) device to accurately determine frequency and phase relationships of the vibrating flow tube and to thereby more accurately determine mass flow rate of a material flowing through the mass flow meter. In a first embodiment, an adaptive notch filter is used to enhance the signal from each corresponding sensor signal on the vibrating flow tubes. In a second embodiment, a plurality of adaptive notch filters are cascaded to enhance the signal from each corresponding sensor signal. In both embodiments, an antialiasing decimation filter associated with each sensor signal reduces the computational complexity by reducing the number of samples from a fixed frequency A/D sampling device associated with each sensor signal. Computational adjustments are performed to compensate for spectral leakage between the fixed sampling frequency and the variable fundamental frequency of the vibrating flow tubes. Despite this added computational complexity, the present invention is simpler than prior designs and provides better noise immunity due to the adaptive notch filtration. Heuristics are applied to the weight adaptation algorithms of the notch filters to improve convergence of the digital filters and to reduce the possibility of instability of the filters interfering with mass flow measurements.

All-fiber tunable filter and laser based on two-mode fiber.

Abstract: We demonstrate an all-fiber acousto-optic tunable filter based on two-spatial-mode coupling, with improved ruggedness and efficiency, by using a new acoustic-transducer design. We use a rigorous modeling of the flexural acoustic wave to analyze the mode coupling with better accuracy. Using the acousto-optic tunable filter, we demonstrate a novel all-fiber tunable laser with a tuning range of more than 20 nm and a linewidth of 0.2 nm.

Multirate receive device and method using a single adaptive interpolation filter

Abstract: A device for receiving a source digital signal transmitted at one or more source symbol frequencies Fs samples a received analog signal transposed into the baseband and delivers an entirely demodulated sample signal. The sampling frequency Fe complies with the generalized sampling condition. A single filter module filtering the sampled signal comprises a digital filter controlled by a filter coefficient vector. The filter coefficients are determined according to a determination criterion allowing for the ratio of the sampling frequency Fe to the source frequency Fs, to interpolate the sampled signal, and for an analysis of the transmission path, in order to limit interference introduced by the path.

Efficient demodulator for bandpass sampled AM signals

Abstract: A simple nonlinear (quadratic) filter is shown to demodulate bandpass sampled AM signals efficiently. The filter is based upon a discrete version of the recently introduced Teager-Kaiser energy operator, but also closely resembles a complex digital sampling demodulator. Such a filter can also be implemented in analogue circuitry.

EKG based heart rate monitor with digital filter and enhancement signal processor

Abstract: A convenient low-cost heart rate monitor. In one embodiment, a digital filter structure includes a low pass filter having a notch at 60 Hz and a bandpass filter which amplifies signals in a frequency range from 10-40 Hz and has a notch at 60 Hz. This digital filter has a recursive structure and uses integer coefficients to simplify and speed up the calculations. A four bit microcontroller may implement the digital filter. The output of the digital filter is subject to enhancement signal processing to emphasize QRS complexes indicative of human heartbeats.

Recursively splitting the low-frequency band with successively fewer filter taps in methods and apparatuses for sub-band encoding, decoding, and encoding and decoding

Abstract: A sub-band encoding method for splitting the frequency spectrum of an input signal into plural bands, encoding the signals of the respective bands and transmitting the encoded signal, is disclosed. The encoding method includes a first step of splitting the input signal into a signal of a high frequency band and a signal of a low frequency band using a first-stage low-pass filter and a first-stage high-pass filter, a second step of downsampling signals of respective frequency bands obtained by the first step, a third step of splitting the frequency spectrum of the low frequency band signal downsampled by the second step, using recursively a pre-set low-pass filter and a high-pass filter, for generating signals of a plurality of frequency bands, and a fourth step of encoding the signals of respective frequency bands obtained by the second step and the third step. The number of taps of the second-stage low-pass filter and the second-stage high-pass filter used in the third step is set so as to be smaller than the number of taps of the first-stage low-pass filter and the first-stage low-pass filter. In the case of splitting the frequency spectrum of a two-dimensional picture signal, the method may include high pass and low pass filtering at each stage in both horizontal and vertical directions, and the relationship involving the number of taps at each stage may be similar.

End-coupled microstrip slow-wave resonator filter

Abstract: A novel microwave narrow-band bandpass filter using end-coupled microstrip slow-wave resonators is proposed. The new filter is not only compact in size due to the slow-wave effect, but it also exhibits a wide upper stopband resulting from the dispersion property. A three-pole microstrip filter of this type has, for the first time, been designed and fabricated. The longitudinal dimension of the filter is smaller than the half-wavelength of a 50 /spl Omega/ line at a midband frequency of 1.53 GHz. No spurious response occurs for the frequency up to 5.5 GHz.

Current-mode universal filters using unity-gain cells

Abstract: Two new current-mode universal filters are presented. The proposed filters use unity gain current and voltage followers. The first filter has three inputs and one output and can realise lowpass, highpass and bandpass responses without any changes in the circuit topology. Realisation of notch and allpass responses can be easily achieved without adding any additional active elements. The second filter has three inputs and one output and can simultaneously realise lowpass, highpass and bandpass responses without any changes in the circuit topology. Realisation of notch and allpass responses can be easily achieved without adding any additional active elements. The proposed circuits enjoy low active and passive sensitivities.

Microwave channelized active filters-a new modular approach to achieving compactness and high selectivity

Abstract: The new class of microwave active filters being presented offers a convenient way to realize miniature filter circuits with sharp passband-to-stopband transitions. The approach, which lends itself to a broad range of narrowband and wideband filtering applications, involves parallel connections of frequency-selective, unilateral network branches that contain both passive and active subcircuits. Highly selective filtering action derives from controlled interferences among branch signal components. Attributes of the new technique include unconditional circuit stability, tolerance for large passive-circuit-element losses, practicability of narrowband lumped-element configurations, graceful performance degradation with active element parameter changes, and the advantage of module-based procedures for design and implementation. The broad applicability of the new approach is illustrated with three experimental demonstration circuits that employ off-the-shelf MMIC amplifier chips. The circuits comprise a 10-GHz notch filter of one quarter percent bandwidth, a 10-GHz bandpass filter of two percent bandwidth, and a 7.5-GHz lowpass filter.

Method and apparatus for adjusting a spectrum shape of a speech signal

Abstract: Adjusting the shape of a spectrum of a speech signal includes the steps of using a first filter with pole-zero transfer function A(z)/B(z) for subjecting a speech signal to a spectrum envelop emphasis and a second filter cascade-connected with the first filter, for compensating for a spectral tilt due to the first filter, independently deriving two filter coefficients used in the second filter for compensating for the spectral tilt from the pole-zero transfer function, and compensating for the spectral tilt corresponding to the pole-zero transfer function according to the derived filter coefficients.

Filter device and dual-band radio system in which the filter device is used

Abstract: A filter device includes at least two filter elements provided in a package, each of the filter elements passing only signals within a predetermined frequency band, the predetermined frequency bands of the filter elements having center frequencies which are distinct from each other. An input terminal is connected to and shared by respective inputs of the filter elements. An output terminal is connected to and shared by respective outputs of the filter elements.

Intersymbol interference cancellation with reduced complexity

Abstract: A signal processing channel and method for reducing intersymbol interference in a sequence of data symbols includes a pre-shaping filter for sharpening leading edges of data symbols and providing trailing edges that approximate a decaying exponential. The output of the pre-shaping filter is combined with a cancellation output from a resistance-capacitance circuit having a time constant selected to provide pulse responses that complement the exponentially decaying trailing edges. The combination of the shaped output and the cancellation output is input to a slicer or other decision device. The 2-level output from the slicer is an input to a decision feedback filter that generates the cancellation output. In addition to the resistance-capacitance circuit, the decision feedback filter includes a delay to properly time the coincidence of the cancellation output with the shaped output from the pre-shaping filter. Preferably, the signal processing channel and method include a gain circuit for setting the peaks of the shaped signals and include adaptive adjustment of the gain, the sharpening by the pre-shaping filter, and the RC time constant of the decision feedback filter.

Switchable duplex filter

Abstract: A radio frequency filter has a band-pass type frequency response which is controllable in a manner such that the frequency response may be moved between the transmission frequency (TX') and reception frequency (RX') of associated radio equipment. Thus the radio frequency filter may be used both as a transmission filter and as receiving filter, provided that the transmission and reception take place at different times. The radio frequency filter includes a change-over switch, which connects the radio frequency filter to a transmitter when the pass band of the radio frequency filter is in the transmission frequency, and which connects to the receiver when the pass band of the radio frequency filter is in the reception frequency.

Ceramic filter with integrated harmonic response suppression using orthogonally oriented low-pass filter

Abstract: A ceramic filter (100) with integrated harmonic response suppression has a ceramic monolithic block filter having a predetermined passband defined by tuned resonators located between an input and an output (116); and at least one of a harmonic trap filter, a lowpass filter and a lowpass microstrip filter, each having an inductive and a capacitive component. This is achievable with a design which incorporates an integrated harmonic response suppression filter directly in or on the dielectric ceramic monolithic block. This can result in a substantial savings in space, cost, and part count in an electronic telecommunications device.

Varactor tuned helical resonator for use with duplex filter

Abstract: A radio frequency filter comprising several resonator circuits, such as duplex filter, is provides with a given passband. Signal frequencies generating spurious response, mirror frequencies, etc. can be effectively attenuated so that the filter comprises one more resonator circuit connected as a bandstop circuit. That comprises a transmission line resonator (Res), a series connection of an inductive element (MLIN2) and a capacitance diode (D), one end of the series connection being couples to a coupling point in the transmission line resonator, dividing it into two parts (TLIN1, TLIN2), and the other end being couples to the output connector (1) of the radio frequency filter. In addition, means are provided for carrying the direct voltage (V+) to the cathode of the capacitance diode, whereby the series resonance frequency changes according to the control voltage.

Passband disturbance reduction in periodic active noise control systems

Abstract: For narrowband active noise control (ANC), the filtered-X least-mean-square (LMS) algorithm with a sinusoidal reference signal forms an adaptive notch filter. Unfortunately, the effect of the secondary path on the uncorrelated noise is to introduce an undesired passband disturbance. To reduce this disturbance problem, this paper analyzes the passband amplification condition and develops an algorithm that consists of two interconnected adaptive filters using the same internally generated reference signal. The first adaptive filter eliminates the uncorrelated noise components in the residual error, and its output is used to update the second adaptive filter for noise control. Computer simulations were conducted to verify the analysis results and to demonstrate the disturbance reduction for periodic ANC while using a relatively large step size to improve the convergence rate.

Digital filters for high performance audio delta-sigma analog-to-digital and digital-to-analog conversions

Abstract: A multi-stage decimation filter is described which consists of comb filters and a single-stage finite impulse response (FIR) filter. A multi-rate comb filter structure is used to reduce the data path width. A multi-stage interpolation filter consists of a half-band filter and a short FIR filter. The short FIR filter is designed with a nonlinear phase to realize an overall constant group delay-signal path. The coefficients of the FIR filters are optimally quantized to canonical-signed-digit form to realize multiplier-free (shift-and-add) filter implementations.

A multiple sampling, single A/D conversion technique for I/Q demodulation in CMOS

Abstract: We have developed a CMOS A/D converter for I/Q demodulation with an analog mirror signal suppression filter in the sampling unit The circuit directly converts a modulated 30 MHz IF signal to digitized I and Q values in the base band with an accuracy of more than 10 b The output data rate is 2 MHz and the power consumption is 270 mW By placing the I/Q split mirror suppression filter on the analog side, we can get a highly integrated system solution for a coherent receiver The circuit uses multiple sampling, that gives the input values to the filter The sizes of the sampling capacitors determine the coefficients for the filter multiplications The sampled charges are then added in order to get the filter additions This total charge is then converted to digital form in a single conversion By requiring the filter to block DC, the filter subtraction becomes a part of the active offset reduction using correlated double sampling Careful layout and very simple circuit solutions make the design possible

Removing RF interferences from P-band airplane SAR data

Abstract: This paper approaches the problem of canceling the disturbances due to RF interferences in P-band, airborne SAR missions. Two techniques are introduced: one exploits MUSIC to estimate the interferences' frequencies, and then performs notch filtering at that frequencies; whereas the other adaptively estimate the interference contributions and cancel them by means of in-phase subtraction. Both techniques have been successfully tested on the data acquired by the DLR E-SAR sensor over urban areas.

An adaptive analog notch filter using log filtering

Abstract: An adaptive analog notch filter is proposed for applications requiring high frequency adaptive signal processing. Based on a log filter circuit topology, this filter design suggests new circuitry for the implementation of analog adaptive filters. Simulation results demonstrate that the filter is capable of rejecting a sinusoid, and with a second notch filter section attached, is capable of rejecting a pair of sinusoids. The circuit design and the adaptation method are described.

Self-adaptive trans-hybrid balance filter

Abstract: An adaptive balance filter includes a transmission path, a reception path, and an adaptive filter having a signal output, a coefficient output for outputting filter coefficients, an error signal input, and a signal input coupled to the transmission path. There is a main filter which has a signal output, a signal input coupled to the transmission path, and a coefficient input terminal. There is also a first subtractor which has one input coupled to the reception path, another input coupled to the signal output of the main filter, and an output forming a further course of the reception path. A second subtractor has one input coupled to the reception path, another input coupled to the signal output of the adaptive filter, and an output coupled to the error signal input of the adaptive filter. A transfer device is connected between the adaptive filter and the main filter and has a control input for loading the filter coefficients of the adaptive filter into the main filter upon command of a corresponding copy signal. A transfer control device having first, second and third inputs is provided. The transfer control device ascertains echo attenuations of the adaptive filter and the main filter from the first, second and third inputs, comparing two echo attenuations with one another, and sending a corresponding copy signal to the transfer device in the event that the echo attenuation of the adaptive filter is higher than the echo attenuation of the main filter.

Performance of optically preamplified receivers with Fabry-Perot optical filters

Abstract: A technique is presented for evaluating the probability of error for an optically preamplified receiver with a fiber Fabry-Perot optical filter of arbitrary bandwidth between the preamplifier and photodiode. The technique permits simultaneous consideration of the distortion due to modulator chirp and fiber dispersion, the optical filter and the baseband electrical filter. The optical filter is characterized by a Lorentzian passband response, and the quadrature noise processes due to the amplified spontaneous emission are represented by Karhunen-Loeve expansions. For an arbitrary optical filter bandwidth, the evaluation of the probability of error depends explicitly on the details of the Karhunen-Loeve expansion. The solution procedure is general and consequently can, in principle, be applied to other filter passband responses. The influence of the bandwidths of the optical and electrical filters on the performance of a 10 Gb/s system is assessed.

An interference reduction scheme and method

Abstract: This invention relates to telecommunications systems. The present invention provides a system and method operable to the cancel interference from digital subscriber line systems. A data signal demodulator having first and second inputs arranged to receive differential data signals and local field RFI signals respectively, wherein a digital adaptive notch filter is formed by DSP means which locates an interferer in the differential signal by adapting its bandwidth and centre frequency by adaptation means and wherein the centre frequency and bandwidth of the notch filter are used to generate a bandpass filter centred on the interferer and of approximate bandwidth to the interferer; whose output, after processing forms a feedback signal which is sampled, processed, weighted and then combined with the local field RFI input signal of the demodulator, which combined signal is summed with the differential input signal to thereby cancel interference coupled onto the first input.

Digital receiver with noise filter which also serves as a feedback filter providing intersymbol interference reduction

Abstract: A receiver for transmitted signals which represent digital symbol values includes a low-pass filter to minimize transmission noise by reducing the receiver bandwidth. However, this creates pulse distortion which can cause intersymbol interference when the filtered signal is supplied to a symbol detector. Such interference is minimized by the inclusion of a feedback loop between the output of the symbol detector and the input to the filter, in which a detected symbol is multiplied by correction factor and supplied to a subtractor where it is subtracted from the signal to be filtered. The result of the subtraction is supplied to the filter. The noise reduction filter is thus also used for matching the intersymbol distortion characteristic of a filtered detected symbol to the intersymbol distortion characteristic of a filtered received signal pulse.

Method and apparatus for detecting speech at a near-end of a communications system, a speaker-phone system, or the like

Abstract: A method and communication system (10), which can be used in a full, pseudo full, or half duplex speaker-phone system, includes a circuit (14) to remove a frequency component from a received signal to output a modified received signal, and a speaker (18) to convert the modified received signal to an acoustic signal (20). The circuit (14) may be, for example, a notch filter, which may remove a sharp frequency range between about 1250 Hz and 1550 Hz. A microphone (26) converts a second acoustic signal (24) to an electrical signal for transmitting, and a detector (30) produces a detector output if the removed frequency is present in the acoustic signal (24). A circuit (34) is also provided to modify a circuit parameter applied to the received signal when the detector (30) produces a detector output. The parameter may be, for instance an attenuation applied to the received signal.

Optical notch filter manufacture in optical fibre waveguide by plastic deformation

Abstract: An optical notch filter is created in a bared length of single mode fibre by using a microburner and longitudinal stretching of the fibre to form a set of equispaced sharply localized non-adiabatic biconical tapers at a pitch corresponding to the beat length between the core-guided HE 11 mode and the cladding-guided HE 12 mode, the latter being stripped downstream of the tapers by a plastics protective coating surrounding the fibre.