Showing papers on "Root-raised-cosine filter published in 1982"

Median filters: Some modifications and their properties

Abstract: Some modifications of the median filter are given and their properties are derived. In addition, some results for standard median filters are given. It is shown that for nonmedian nth ranked-order operations, repeated application of the operation will reduce any signal to a constant. Also, it is proved that the output of a recursive median filter is invariant to subsequent passes by the same filter.

Recursive adaptive filter design using an adaptive genetic algorithm

Abstract: An adaptive genetic algorithm for determining the optimum filter coefficients in a recursive adaptive filter is presented. The algorithm does not use gradient techniques and thus is appropriate for use in problems where the function to be optimized is non-unimodal or non-quadratic, such as the mean-squared error surface in a recursive adaptive filter. The mechanisms of the algorithm are inspired by adaptive processes observed in nature. After an initial set of possible filters is randomly selected, each filter is mapped to a binary string representation. Selected bit strings are then transformed using the operations of crossover and mutation to build new "generations" of filters. The probability of selecting a particular bit string to modify and/or replicate for the next "generation" is inversely proportional to its estimated mean-squared error value. Hence, the process not only examines new filter coefficient values, but also retains the advances made in previous "generations". Computer simulations of the algorithm's performance on unimodal and bimodal error surfaces are presented.

Optimum FIR Transmitter and Receiver Filters for Data Transmission Over Band-Limited Channels

Abstract: The paper deals with the design of digital transmitter and receiver filters with finite impulse response (FIR) for data transmission over band-limited channels. The filters are matched and satisfy a zero intersymbol interference constraint when cascaded. For baseband transmission, the filters achieve optimum spectral concentration in the frequency range [-(1+\beta)/2T, (1+\beta)/2T] . Mathematically, the filter design leads to a generalized eigenvalue problem which is solved numerically by a projected gradient Procedure. For transmission over bandpass channels by combined amplitude and phase modulation, the design technique is modified so that filters with complex-valued impulse response and optimum spectral concentration in the range of positive bandpass frequencies [f_{c} - (1+\beta)/2T, f_{c} + (1+\beta)/2T] are obtained. In addition, the complex formulation allows the design of impulse responses with enhanced spectral attenuation in the corresponding range of negative frequencies in order to minimize imageband interference. Results are shown in terms of filter coefficients, signal spectra, and spectral concentrations obtained. For example, filters designed for a voiceband data modem operating at a symbol rate of 2400 baud achieve a spectral concentration of 98.5 percent with 24 coefficients and \beta = 0.1 , and with only 0.001 percent of the total energy in the imageband region.

State description for the root-signal set of median filters

Abstract: Median filtering is a simple digital technique for smoothing signals. One main characteristic of the filter is that it maps the input signal space into a root signal space, where signals invariant to median filters are called roots of the signal. In this paper, we develop the theory for the root signal set of median filters. A tree structure for the root signal set is obtained for binary signals. The number of roots R (n) for a signal of length "n" and window size filter "2s- 1" is exactly represented by the difference equation R(n) = R(n - 1) + R(n - s). A general solution is obtained in a Z domain approach. Finally, a method for faster one dimensional median filter operation is introduced.

ϵ‐separating nonlinear digital filter and its applications

Abstract: This paper proposes an ϵ-separating nonlinear digital filter (called an ϵ-filter). This filter is intended for effective filtering of low-amplitude noise superposed on the signal with sharp discontinuities and can be realized by combining a simple nonlinear element with a conventional linear filter. In this paper, the basic ϵ-filter and its modifications to a trend-adaptive filter and a two-dimensional filter are described. The effectiveness of the new filter is demonstrated by computer simulation. Some of its application to EEG analysis, image processing and coding are also presented.

Reduction of noise in TV rate electron microscope images by digital filtering

Abstract: SUMMARY Digital framestores can be used to reduce noise in TV rate electron microscope images. Three operating modes: averaging, recursive filtering and Kalman filtering have been investigated. It is shown that there is an optimum choice of filter parameters.

A new IIR Nyquist filter with zero intersymbol interference and its frequency response approximation

Abstract: A new infinite impulse response (IIR) Nyquist filter with zero intersymbol interference is proposed. The necessary and sufficient conditions for the transfer function are obtained. The proposed IIR Nyquist filter requires only frequency-domain optimization. Multistep optimization, using the iterative Chebyshev approximation, is proposed. This method is able to design a new kind of IIR Nyquist filter with the minimum order. Numerical examples for 30- and 15-percent rolloff rates are illustrated. From these examples, it is confirmed that the IIR approach can reduce the filter order and hardware size, compared with the conventional finite impulse response (FIR) Nyquist filters. Its efficiency becomes marked for high Q Nyquist filters.

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.

Combination speech synthesis and recognition apparatus

Abstract: A reconfigurable lattice filter is employed to permit the same circuitry to function as a speech synthesizer and as a speech analyzer or recognizer. The lattice filter can be configured both as an all-pole filter (for synthesis) and as an all-zero filter (for analysis), with the choice being determined by the state of an analysis/synthesis signal (i.e., mode control signal) provided thereto. The connections between various elements in the circuitry are controlled by the analysis/synthesis signal, also. In synthesis mode, partial correlation coefficients are supplied to the filter from a microprocessor. The filter is excited by a one of a number of stored patterns simulating a glottal pulse for voiced sounds and by a pseudo-random noise generator for unvoiced sounds. In analysis mode, appropriate feedback control paths are enabled so as to provide to the filter coefficients which change in response to changes in the input speech waveform. Coefficient values thus determined are averaged over fixed intervals and successions of such averaged coefficient sets produce representations of words or phrases which can then be used for speech recognition.

Telephony apparatus having filter capacitor switched to undergo discrete phase jumps

Abstract: A telephony audio signal is transmitted as a digital signal by a transmitter that derives: (1) a master clock of frequency F MC , (2) a time slot signal of frequency F TS =F MC ÷2 N' , and (3) a transmit control signal of frequency F WT =F MC ÷2 N . The frequencies F TS and F WT control the digital signal transmission frequency. A receiver responsive to the digital signal derives the master clock frequency and includes a switched capacitor filter. The receiver derives a reception synchronization signal having a frequency F RS that is much lower than and asynchronous with the master clock frequency but which has approximately the same frequency as the time slot signal. In response to the information signal and the master clock frequency at the receiver and to the reception synchronization signal, a pulse amplitude modulated signal indicative of the information signal to which the receiver is responsive is supplied to the switched capacitor filter. In response to the master clock frequency at the receiver and the reception synchronization signal a switching control signal for the filter is derived. The filter is switched a predetermined number of times during each period of the reception synchronization signal. The switching times undergo discrete phase jumps that occur at times such that the filter is switched in a plesiosynchronous manner with the reception synchronization signal.

Adaptive echo canceller with AGC for transmission systems

Abstract: An adaptive echo canceller for a full duplex transmission system comprises a cascade arrangement of an adaptive digital filter and an automatic gain control device. The AGC device receives the output signal from the adaptive digital filter and delivers the estimate of the echo signal to the receive line for subtraction. It has a multiplier receiving the output signal and a signal representative of the multiplication factor from a gain adaptation circuit. A separate gain change circuit is connected to receive the output signal from the adaptive digital filter and simultaneously modifies the tap coefficients of said filter and the multiplication factor of the AGC device in opposite directions for maintaining the output of the adaptive digital filter in a predetermined range.

Communications receiver having a switchable filter input stage

Abstract: A multichannel communications receiver, e.g. a CATV converter, includes a switched filter for reducing third order intermodulation distortion by limiting the number of channels processed by the receiver. In a preferred embodiment, the filter is switchable between a highpass and a lowpass configuration by a control signal supplied to two switching diodes. The input and output of the filter are coupled by a mutual inductance which partially compensates for the effect of parasitic inductance at higher frequencies.

Fast recursive estimation using the lattice structure

Abstract: This paper presents the theory for a rapidly converging adaptive linear digital filter. The filter weights are updated for every new input sample. This way the filter is optimal (in the minimum mean square error sense) for all past data up to the present, at all instants of time. This adaptive filter has thus the fastest possible rate of convergence. Such an adaptive filter, which is highly desirable for use in dynamical systems, e.g., digital equalizers, used to require on the order of N2 multiplications for an N-tap filter at each instant of time. Recent “fast” algorithms have reduced this number to like 10 N. One of these algorithms has the lattice form, and is shown here to have some interesting properties: It decorrelates the input data to a new set of orthogonal components using an adaptive, Gram-Schmidt like, transformation. Unlike other fast algorithms of the Kalman form, the filter length can be changed at any time with no need to restart or modify previous results. It is conjectured that these properties will make it less sensitive to digital quantization errors in finite word-length implementation.

New Filter Technology in Picture Processing

Abstract: In this paper we describe a new filtering technique for processing video signals called the "LOGICAL FILTER". With the Logical Filter, video signals in the time domain are assumed to be a set of "patterns" and the pattern recognition logic is applied to the filtering of the video signals.

Nonlinear filtering using linear combinations of order statistics

Abstract: In this paper we consider a nonlinear filter whose output is given by a linear combination of the order statistics of the input sequence. Assuming a constant signal in white background noise, the coefficients in the linear combination are chosen to minimize the output MSE for several noise distributions. This new general filter is superior to the well-known median filter, since the median is just a special case.

Narrow band digital filter

Abstract: A narrow band digital filter is disclosed for rejecting all undesired frequencies, providing a binary signal indicative of when the input frequency is within an acceptance band. The digital filter may be placed within a stereo AM receiver to monitor the pilot tone of the incoming signal with clocking pulses provided by the IF stage of the radio receiver.

Jump-phenomenon in active-RC filters

Abstract: The dynamic range of an active- RC filter is limited by the nonlinearity of the active device used, which causes abrupt transition in the output response, resulting in what is known as the jump-phenomenon. A new algorithm is presented to compute the signal level and the frequency at the jump-point. It is relatively simple compared to the existing methods and is applicable to any filter configuration, using BJT or FET-input operational amplifiers (OA's).

An expandable single-IC digital filter/Correlator

Abstract: The commercial success of single-integrated circuit multipliers and correlators has prompted the design of a new integrated circuit for video-speed digital convolution and correlation. This device is based on two concepts: 1.) the merging of delay and pipeline registers and 2.) "bit-slicing" the filter array in both the signal data word and coefficient dimensions. The device can update one coefficient per clock cycle. The result is an expandable "building block" which can be cascaded or paralleled to give the desired length, signal data word length, and coefficient data word length for any desired filter up to 20 MSPS.

The design of optimal partial response data transmission filters

Abstract: The time response of a Class I partial-response filter to an impulse should be large at two adjacent sample points,impulse should be large at two adjacent sample points, and near-zero at all other sample points, in order to minimize intersymbol interference.This paper gives an analytic design procedure for a realizable filter that meets this requirement in an optimal manner, and also optimizes other frequency and time domain criteria. Numerical examples are given, which illustrate the performance that can be attained with this class of filter.

Structures for digital filter banks

Abstract: Signal processing structures, incorporating one and two digital filter banks (DFB's), are examined with the aid of a polyphase transform that is defined. Convenient matrix descriptions for DFB structures and the short-time Fourier transform are obtained. A condition for decimation in DFB's without aliasing is obtained and shown to be equivalent to shift invariance. Certain commonly employed filter banks with real and complex structures age seen to admit realization with the low noise, memory efficient SIDO filter structure.

Self-corrected electric filters

Abstract: A self-corrected electric filter with localized constant elements having two inputs, two outputs, input and output matching means, a group of an even number of filter elements in cascade for filtering the wide band signal, group delay time correction means comprising adjacent secondary couplings connected between at least two successive filter elements and secondary non-adjacent couplings connected between at least two non-adjacent filter elements

Application of digital filters in the processing of eye movement data

Abstract: Digital filter techniques have been applied to the analysis of eye movement data. Methods were developed to calculate eye velocity and eye acceleration in real-time from an electronystag-mogram (ENG) signal that was recorded using a one-pole RC high-pass filter in the preamplifier. Nonrecursive, finite impulse-response digital filters were designed to remove the effects of the RC high-pass filter and calculate the first and second time derivatives of the ENG signal, as well as remove high-frequency noise. Applying these new techniques to the analysis of vestibular nystagmus enables estimation of the transfer characteristics of the vestibuloocular system.

Transmultiplexer Realization with Multistage Filtering

Abstract: Starting from the general principle of frequency shifting a 4 kHz bandwidth digital filter to perform the TDM-FDM conversion, a new decomposition of the filter is introduced. This decomposition allows for the use of a DFT of much lower dimension than the number of channels to be multiplexed, together with a few modulators (multipliers) working at the highest bit rate, and a tree configuration of filters and simple modulators. The filters are of a few different types and can be designed to have very simple coefficients, so avoiding the use of true multipliers; furthermore, wave digital filters, with their known advantages, can be used for some of the more severe filtering steps.

A Kalman filter procedure for the processing of the electroencephalogram

Abstract: In the present paper a Kalman filter procedure is illustrated for the reduction of muscular noise superimposed to the electroencephalografic traces (EEG). Such a noise, in fact, has a bandwidth which overlaps the signal carrying the information content useful for the clinical standpoint and, therefore, can not be removed by means of classical digital filtering. A Markov model is used for identifying the signal model (supposed generated by an ARMA process) and the noise model (conceived on the basis of experiments of neurophysiological evidence). The experimental results show a good performance of the filter on the discrete-time EEG signal which is also quantified by the spectral information and the values of the prediction error of the filter itself. Comparison is then carried on with a classical low-pass FIR filter (mostly used in practice) which can not be aggressive enough towards the noise contained in the signal bandwidth but which can undoubtly ameliorate the performance of the Kalman filter.

Adaptive control signal filter for audio signal expander

Abstract: A resistance-capacitance filter smoothes the control voltage of an audio signal expander. An analog gate couples the greater, in a given sense, of the smoothed control voltage and a further voltage to the control terminal of a variable gain device in the audio signal path, the further voltage being equal to the filter input voltage less a constant equal to Vbe. A time constant modifier circuit reduces the filter time constant when the filter input voltage differs in either sense from the smoothed control voltage by a fraction, less than unity, of Vbe. The resultant, relatively "narrow" dead zones of the adaptive filter enable operation of the expander with relatively low signal voltage levels thereby enabling a corresponding reduction in supply voltage requirements and providing further advantages such as reduced power dissipation, reduced heat build-up and improved reliability.

Transmitter for angle-modulated signals

Abstract: Transmitter for angle-modulated signals having an input for binary signals, a premodulation filter and a frequency modulation arrangement. In order to improve the error rate of the system of transmitter and receiver a premodulation filter is used having a pulse response h(t): ##EQU1## wherein g(t) is the pulse response of a Gaussian low-pass filter and T the duration of a binary signal element. The postdemodulation filter 6-4 (6-5) has a pulse response of the same general form for optimum results.

"Optimum" filter for speech enhancement using integrated digital signal processors

Abstract: An algorithm for adaptive filtering of speech in additive acoustic noise is presented In comparison with other speech enhancement algorithms a very low computational cost is required The efficiency of the algorithm is achieved by proper modifications of a Wiener optimum filter with a filter bank Due to the use of digital Lerner band pass filters a very compact filter bank structure is obtained The speech enhancement system has been implemented with INTEL 2920 signal processors As a result, the quality of noisy speech signals band-limited to (300-3300) Hz can be improved for input signal to noise ratios as low as -5dB

Analysis and implementation of the adaptive notch filter for frequency estimation

Abstract: This paper enhances some theoretical and implementation aspects of a constrained autoregressive moving average model, the notch filter model developed in [1] for the estimation of sinusoidal signals in additive, uncorrelated noise, colored or white. This model is shown to approximate the actual signal plus noise model. In addition, the parameter estimates obtained by minimization of the output power of the notch filter approximate the maximum likelihood estimate of the model parameters. The relationship of the notch filtering approach to the existing autoregressive and Pisarenke methods is established. Next, a scheme to combine fast convergence and unbiased estimation is suggested. Lastly, certain implementation aspects of the filter are considered and the method is shown to be amenable to parallel processing.