Showing papers on "Low-pass filter published in 1982"

Removing Tidal-Period Variations from Time-Series Data Using Low-Pass Digital Filters

Abstract: Several low-pass, digital filters are examined for their ability to remove tidal Period Variations from a time-series of water surface elevation for San Francisco Bay. The most efficient filter is the one which is applied to the Fourier coefficients of the transformed data, and the filtered data recovered through an inverse transform. The ability of the filters to remove the tidal components increased in the following order: 1) cosine-Lanczos filter, 2) cosine-Lanczos squared filter; 3) Godin filter; and 4) a transform fitter. The Godin fitter is not sufficiently sharp to prevent severe attenuation of 2–3 day variations in surface elevation resulting from weather events.

Nonlinear Restoration of Noisy Images

Abstract: The restoration of images degraded by an additive white noise is performed by nonlinearly filtering a noisy image. The standard Wiener approach to this problem is modified to take into account the edge information of the image. Various filters of increasing complexity are derived. Experimental results are shown and compared to the standard Wiener filter results and other earlier attempts involving nonstationary filters.

Synthesis and Realization of Narrow-Band Canonical Microwave Bandpass Filters Exhibiting Linear Phase and Transmission Zeros

Abstract: The present study is concerned with the synthesis and realization of narrow-band canonical microwave bandpass filters exhibiting additional couplings between nonsuccessive resonant circuits. These bypass couplings effect, on the one hand, attenuation poles at finite frequencies and, on the other hand, make the bandpass filter nonminimum-phase, and therefore offer degrees of freedom for group delay equalization in the passband. The realization of canonical network structures with bypass couplings is possible by extracting from the chain matrix an ideal transformer, the input of which is connected, for example, in shunt with and its output in series with the remainder two-port. The extraction is associated with the reduction by 2 of the degree of the denominator polynomial of the chain matrix elements. To realize a clear correlation between the bypass elements and the electrical performance, canonical structures with sections that are bypassed only once or twice are proposed. Theoretical and measured curves of the examples of realization for 1235 MHz, consisting of coaxial cavities, and for 4000 MHz, consisting of dual-mode cavities, are seen to be in close agreement.

Conversion and Matched Filter Approximations for Serial Minimum-Shift Keyed Modulation

Abstract: Serial minimum-shift keyed (MSK) modulation, a technique for generating and detecting MSK using series filtering, is ideally suited for high data rate applications provided the required conversion and matched filters can be closely approximated. Lowpass implementations of these filters as parallel inphase- and quadrature-mixer structures are characterized in this paper in terms of signal-to-noise ratio (SNR) degradation from ideal and envelope deviation. Several hardware implementation techniques utilizing microwave devices or lumped elements are presented. Optimization of parameter values results in realizations whose SNR degradation is less than 0.5 dB at error probabilities of 10-6.

Direct digital loudspeaker

Abstract: A method of and apparatus for converting a digital signal to sound, the digital signal being encoded in a sequence of code words at a signal encoding frequency, the code words representing the analog sound pressure of an original audio signal, with decoding of the digital signal occurring after electro-acoustic transduction through mechanical rectification and characteristics of a listener's ear, includes utilizing a plurality of substantially identical sound pressure generating elements each having an individual driver associated therewith, and selectively energizing the drivers in a pulsed manner at the signal encoding frequency in combination in response to a respective order of the bits of each code word of a digital signal from a most significant bit to a least significant bit. The sum of the air pressures produced by the sound pressure generating elements in response to each of the successive code words of the digital signal has a magnitude corresponding to the analog value of the respective code word, and the auditory system of the listener has the characteristics of a low pass filter whereby the listener receives the sum of the air pressures as the analog sound pressure of the original audio signal.

A Novel Class of Generalized Chebyshev Low-Pass Prototype for Suspended Substrate Stripline Filters

Abstract: A novel class of low-pass prototype filters having an equiripple passband response with three transmission zeros at infinity and the remainder at a finite real frequency is presented. The prototypes are synthesized using the alternating pole technique to obtain directly the even-mode or the odd-mode admittance and little accuracy is lost for prototypes up to degree 15. Tables of element values for commonly used specifications are included. The tables are useful for the design of TEM-mode microwave broad-band filters, diplexers, and multiplexers, particularly for a printed circuit form a realization. A design example of a low-pass microwave broad-band filter designed and constructed in suspended substrate stripline (SSS) configuration is given and experimental results are also presented.

Optimization of short digital linear filters for increased accuracy

Abstract: The accuracy of short length digital linear filter operators can be substantially increased if the sampling interval as well as the abscissa shift are properly adjusted. This may be done by a trial and error process of adjustment of these parameters until the error made by the filter operator, applied to a suitably chosen test function, is smallest. As an illustration of the application of this method, 7-, 11- and 19-point filters for the calculation of Schlumberger apparent resistivity from a known resistivity transform are designed. Errors with the new 7-point filter are seen to be less than those with a 19-point filter of conventional design. The errors with the new 19-point filter are two to three orders of magnitude smaller than those made by the conventional 19-point filter. The new method should provide digital linear operators that allow significant improvements in accuracy for comparable computation efforts, or substantial reduction in computation for comparable accuracy of results, or something of both.

Design of Low-Pass Elliptic Filters by Means of Cascaded Microstrip Rectangular Elements

Abstract: A new method for synthesizing nonredundant low-pass elliptic filters in a microstrip configuration is presented. The realization consists of the cascade connection of proper rectangular elements, each one corresponding to four reactive elements of the lumped-constant prototype. This allows an effective control of parasitic and unwanted reactance which results in the possibility of realizing higher order filters with cutoff frequencies up to X-band. Fifth- and seventh-order filters were fabricated on alumina substrates showing very good performance, particularly in the passband.

The truncated second-order nonlinear filter revisited

Abstract: By rederiving the truncated second-order nonlinear filter, it is shown that the original derivations of this filter contain errors, or at least illogical approximations. The so-called modified truncated second-order nonlinear filter is, furthermore, shown to be the correct form of this filter provided a small correction is made in the discrete-time case.

Implementation of active distributed RC anti-aliasing/smoothing filters

Abstract: The design of an active distributed RC filter is described for use as an anti-aliasing/smoothing filter in a PCM channel bank filter. A new filter configuration (known as a Rauch filter) has been chosen which uses the op amp in inverting configuration with a zero common mode signal and thus results in improved high-frequency attenuation characteristics along with good power supply rejection. A conventional Sallen and Key configuration with distributed RC elements is also discussed.

Method and apparatus for monitoring the tool status in a machine tool with cyclic machining

Abstract: The vibration level (a) is measured and applied to controllable filter (F). The filter (F) is a band pass filter that only passes a narrow band around a certain frequency that is determined by the frequency (1) of the machining cycle. The output signal of the filter is then compared to one or several reference levels (7) and an alarm signal is outputted in case of unacceptable deviation from the reference level.

Automatic sound volume control device

Abstract: An automatic volume control device for a car stereo or the like which adjusts the volume according to an environmental noise level includes a microphone and a low pass filter circuit whose characteristic is designed so as to correspond to the human auditory characteristic at a low frequency end, but so as to cut out higher frequency components such as voices, etc. A time constant circuit serves to smooth volume transients in case the environmental noise sharply increases.

Integrated switched-capacitor low-pass filter with combined anti-aliasing decimation filter for low frequencies

Abstract: A new combined antialiasing decimation filter is presented which allows the implementation of a low-frequency switched-capacitor filter on a single chip. Experimental results are presented for a CMOS second-order low-pass filter with 1 dB passband ripple, a cutoff frequency of 2 Hz, and a dynamic range of 84 dB. The decimation filter converts the input clock of 16 kHz into an output clock of 250 Hz. The integrated anti-aliasing filter has a low pole frequency of about 3 kHz.

On Computational Complexity in Digital Transmultiplexer Filters

Abstract: The quality standards recommended by the CCITT are the reference for the design of digital transmultiplexer filters. Their impact on filter specifications is analyzed, and expressions are given for the filter order, coefficient, and data word lengths in FIR and IIR approaches. Improvements on the polyphase filter bank design are pointed out, and the method is presented as a hybrid FIR-IIR technique. Finally, it turns out that the multiplication rate is the crucial parameter for complexity comparison between the approaches considered.

Presampling filtering, sampling and quantization effects on the digital matched filter performance

Abstract: Due to the increased capability and reduced cost of digital devices, there has recently been a growing trend to digitize the matched-filtering data detector in the receiver. Comparing with an idealized integrate-and-dump analog matched filter, the digital matched filter (DMF) requires more Eb /No in order to achieve the same bit error rate performance because of the presampling filtering, sampling, and quantization effects. This paper analyzes the performance degradation resulting, separately and jointly, from these three effects. Quantitative results are provided for commonly chosen sets of design parameters. For a given performance degradation budget and complexity limitation, these results could be applied to choose the optimum DMF design parameters including the presampling filter bandwidth, the sampling rate, the number of quantization bits, and the spacing between adjacent quantization levels. 1.0 INTRODUCTION The study of sampling and quantization effects on the digital matched filter (DMF) has recently received much attention in evaluating the performance of digital receivers that employ matched-filter detection.[1, 2, 3] A fundamental case of interest is the case when the input to the DMF consists of (1) an NRZ-L PCM baseband signal and (2) an additive white Gaussian noise process. The NRZ-L PCM signal appears in the time domain as a train of rectangular pulses of voltage levels +V or -V (see Figure 1), depending on whether the transmitted data bit is a 0 or a 1. For such a signal plus noise, it is well known that the integrate-and-dump filter is the optimum (or the matched-filtering) detector which results in the minimum error probability as shown in Figure 2. The increased stability, reliability, and flexibility, as well as the decreased size and cost make the digital implementations of many analog matched filters highly desirable. Figure 3 illustrates one possible digital implementation of the integrate-and-dump filter. As evident from the figure itself, the performance of this digital integrate-and-dump (matched) filter depends upon three system parameters: (1) B (Hz), the bandwidth of the presampling low-pass filter (2) fs (samples/bit), the sampling rate of the sampler in samples per data bit (3) m (bits), the number of bits of the quantizer Because of presampling filtering, sampling and quantization effects, the DMF requires more Eb/No than the analog matched filter. Thus, the degradation factor D of the DMF may be defined as the required increase in Eb/No for the DMF in order to yield the same error probability as the analog matched filter. In what follows, the degradation factor is derived in detail with quantatative results presented for commonly chosen sets of design parameters. 2.0 ANALYSIS This section is devoted to deriving the error probabilities and hence the degradations for the DMF. Refer to the block diagram of the DMF in Figure 3. Let the received signal plus noise at the input to the DMF be expressed as x(t) = s(t) + n(t) (1) where n(t) = a stationary white Gaussian noise process of two sided spectral density No/2 = a rectangular pulse train of voltage levels +V or -V and u(t) = a rectangular pulse of amplitude V ana duration T The energy per bit to one-sided noise density ratio is hence given by

Filter and demodulation circuit

Abstract: A filter and demodulation circuit is proposed which, when used in a radio-frequency receiver, produces an increase in the sensitivity of the receiver. In the filter and demodulation circuit, the intermediate frequency is divided into at least two parallel channels (11, 12) at the input (10). Each channel contains a series circuit comprising a mixing and oscillator circuit (13, 15; 14, 16), a controllable IF filter (17, 18), a demodulator (19, 20) and a high-pass filter (21) or low-pass filter (22). One transmission channel essentially transmits the modulation frequencies of a first frequency range only and the other transmission channel transmits the modulation frequencies of a second frequency range. The AF voltage at the output of each transmission channel re-adjusts the IF filter of this channel and the oscillator circuit of the mixing and oscillator circuit of the other channel.

Design and application of finite impulse response digital filters

Abstract: The finite impulse response (FIR) digital filter is a spatial domain filter with a frequency domain representation. The theory of the FIR filter is presented and techniques are described for designing FIR filters with known frequency response characteristics. Rational design principles are emphasized based on characterization of the imaging system using the modulation transfer function and physical properties of the imaged objects. Bandpass, Wiener, and low-pass filters were designed and applied to 201T1 myocardial images. The bandpass filter eliminates low-frequency image components that represent background activity and high-frequency components due to noise. The Wiener, or minimum mean square error filter ‘sharpens’ the image while also reducing noise. The Wiener filter illustrates the power of the FIR technique to design filters with any desired frequency reponse. The lowpass filter, while of relative limited use, is presented to compare it with a popular elementary ‘smoothing’ filter.

Design of separable denominator 2-dimensional digital filters possessing real circularly symmetric frequency responses

Abstract: A new formulation of the approximation procedure to design 2-dimensional digital filters with separable denominators possessing real circularly symmetric frequency-response characteristics is described. In this scheme, a 1-dimensional function approximating the response on the axes of the 2-dimensional plane is first obtained. The 2-dimensional function is then constructed from the 1-dimensional function by adding extra terms, whose coefficients are determined employing an optimisation procedure, so as to yield the desired response, as closely as possible, in the rest of the 2-dimensional plane. Techniques to reduce the number of unknown coefficients to be determined by optimisation are also discussed.

Pulse detecting circuit of receiver for use in digital radio communication system

Abstract: In a pulse detector of a receiver for use in a digital radio communication system comprising a low pass filter supplied with an input signal, a first resistor and a capacitor which are connected in series between an output terminal of the low pass filter and ground to constitute a longer time constant integrating circuit, a second resistor connected in parallel with the first resistor through a switch controlled by a control input, the first and second resistors and the capacitor constituting a shorter time constant integrating circuit, and a comparator with one input connected to the output of the low pass filter and the other input to a juncture between the first and second resistors and the capacitor, the potential of the juncture being utilized as a reference potential of the comparator, there is provided a switch connected between the juncture and the ground in series with the capacitor. According to this invention, pulse detection can be made more accurately and power can be much more saved than a prior art pulse detecting circuit.

Sampled data filter system as for a digital TV

Abstract: Low pass sampled data filters tend to produce pre- and post- over/undershoots in the output signal in response to input signal transitions. These over/undershoots or sequence of artifacts are eliminated in the present system by the following method. The input signal is monitored for transitions exceeding an amplitude which tends to generate objectionable artifacts. The values of such transitions are stored and applied to a plurality of weighting circuits to generate a plurality of signal values corresponding to respective ones of the sequence of artifacts. The weighted samples are selectively combined with the filter output samples in time correspondence with the occurrence of the output artifacts to cancel the effects of the artifacts.

Miniaturized antenna duplexer using saw filter

Abstract: An antenna duplexer is made compact by a use of a SAW filter, and yet it eliminates the possibility that the SAW filter might be burned, and it avoids additional circuits, e.g., an impedance compensation circuit. The antenna duplexer comprises a local oscillation filter. A reception filter is coupled to the local oscillation filter. The coupled side is partly constituted by a SAW filter. A transmission filter is coupled to the reception filter and an antenna is coupled between the reception filter and the transmission filter.

Method and apparatus for digital time-variant filtering

Abstract: A method and apparatus for digital time-variant filtering are disclosed. The digital time-variant filters are implemented using the Hilbert transform and a time-varying instantaneous frequency control function for controlling the cutoff frequency. The digital time-variant filters are adaptable to various filtering processes, that is, as either a high-cut, low-cut, band-pass, or reject filter. The various digital time-variant filters are stable in that no unusual transients occur in the output signal; are flexible in that the cutoff frequency is controlled by a selectable time-varying control function signal; and are efficient in that computational requirements are only twice to 16 times that of known digital time-invariant filters.

Apparatus for receiving reflected ultrasonic waves in ultrasonography

Abstract: An echo signal receiver receives higher level echo signals reflected from strong reflections sources or reflecting sources at areas close to a probe in a range of higher frequencies, and also receives lower level echo signals reflected from weak reflection sources or sources at areas remote from the probe in a range of low frequencies. The echo signal receiver comprises a multiple stage amplifier circuit for receiving the echo signal. Outputs from amplifiers at front stages are delivered via filters having higher frequency bands, and outputs from amplifiers at rear stages are delivered via filters having lower frequency bands. The outputs are passed through the filters and then combined into a single processed echo signal. As an alternative, a video signal derived from an echo signal which has passed through a circuit for varying frequency characteristics is compared with a reference voltage by a comparator, which produces an output that is rounded off by removing high frequency components in a low pass filter. The rounded off signal is fed to a circuit for varying frequency characteristics to control the same to change its frequency ban while the signal is being received. In another embodiment, the echo signal is received through a variable low cut filter having a cutoff frequency which can be increased or reduced dependent on the level of the echo signal.

Phase-locked loop circuit with variable bandwidth filter

Abstract: A phase-locked loop circuit comprises an oscillator, a phase comparator which compares the phase of an input signal with the phase of an oscillator signal, a detector which detects when the phase-locked loop circuit is locked within a predetermined frequency range and produces a corresponding lock detecting signal, and a filter circuit including a variable current source which produces a variable current in response to a change of state of the lock detecting signal to control the bandwidth of the filter circuit, a filter element which receives the variable current, a differential amplifier which receives the phase-compared signal, a current mirror circuit which receives the variable current from the variable current source, and a buffer circuit connected to the filter element which supplies an output signal to the oscillator to lock the frequency of the oscillator signal to the frequency of the input signal.

A digital impulse noise removal filter

Abstract: Sensitive absorption measurements may be contaminated by the presence of particulates in the sample. Scattering by these particles can generate impulse noise (spikes) on the spectrum. This problem is especially serious in spectroscopy with a focused laser source, since the laser probes a small cross section of the sample. Often, this noise ruins an otherwise usable spectrum. If the data are stored digitally, however, it is often possible to remove noise spikes from a spectrum numerically. The commonly used smooth method of Savitzky and Golay does a poor job of removing these spikes. Since they are isolated and unidirectional errors, the smooth method simply transforms them to the impulse response of the filter function itself. Similarly, other low pass digital filters will also transform spikes into bands which may interfere with the analysis of real spectral features.

Video signal processing unit

Abstract: A processing circuit for a composite color television signal formed of a chrominance component and a luminance component includes a comb filter of the type including a 1H delay line. In order to process the chrominance component, the comb filter is followed by an operational circuit providing a difference signal representing vertical correlation of the input chrominance component for consecutive line intervals, and a feedback loop circuit for combining the input chrominance component with the feedback difference signal before application to the comb filter. A detecting circuit detects correlation of video information in the luminance component in vertically aligned portions of successive horizontal line intervals, and provides a detecting signal which is used to adjust the feedback loop gain on the feedback loop circuit in accordance with the detecting signal. The processing circuit can further comprise a luminance comb filter including the delay line and an additive combining circuit, which is followed by a subtractive combining circuit and a band pass filter having an input connected to an output of the subtractive combining circuit and an output connected to an input of the latter. This circuit gives the luminance comb filter a flat frequency transfer characteristic. In the detecting circuit, a delay circuit is provided formed of a pre-emphasis circuit, an AM modulator, a delay line, an AGC circuit, an AM detecter, a de-emphasis circuit, and a low pass filter.

Color channel signal-to-noise improvement in digital television

Abstract: Circuitry included in a TV receiver for enhancing the signal-to-noise ratio of the chrominance signal includes an adaptive linear phase, low pass filter which has its bandwidth controlled responsive to the upper frequency components of the current chrominance signal. A frequency analyzer coupled to the chrominance channel counts the number of chrominance signal excursions during predetermined periods to generate a number related to the maximum signal frequency during such periods. The number is applied to a decoder which generates address codes for application to a look up table which provides filter coefficients for altering the filter bandwidth in accordance with the current maximum signal frequency components.