Showing papers on "Root-raised-cosine filter published in 1987"

A theory of multirate filter banks

Abstract: Multirate filter banks produce multiple output signals by filtering and subsampling a single input signal, or conversely, generate a single output by upsampling and interpolating multiple inputs. Two of their main applications are subband coders for speech processing and transmultiplexers for telecommunications. Below, we derive a theoretical framework for the analysis, synthesis, and computational complexity of multirate filter banks. The use of matrix notation leads to basic results derived from properties of linear algebra. Using rank and determinant of filter matrices, it is shown how to obtain aliasing/ crosstalk-free reconstruction, and when perfect reconstruction is possible. The synthesis of filters for filter banks is also explored, three design methods are presented, and finally, the computational complexity is considered.

Theoretical analysis of the max/Median filter

Abstract: Median filtering has been used successfully for extracting features from noisy one-dimensional signals; however, the extension of the one-dimensional case to higher dimensions has not always yielded satisfactory results. Although noise suppression is obtained, too much signal distortion is introduced and many features of interest are lost. In this paper, we introduce a multidimensional filter based on a combination of one-dimensional median estimates. It is shown that threshold decomposition holds for this class of filters, making the deterministic analysis simpler. Invariant signals to the filter, called root signals, consist of very low resolution features making this filter much more attractive than conventional median filters.

Truncated subband coding of images

Abstract: Image data is analyzed in a number of iterated analysis procedures, using two-dimensional quadrature mirror filters to separate low-pass spatial filter response component and three differently oriented high-pass spatial filter response components, which filter response components are decimated in both dimensions. The high-pass filter response components are coded as is. The low-pass filter response component is coded as is only in the last iteration; in the earlier analysis procedures the low-pass filter response component provides the input data for the secceeding analysis procedure.

Optimum time domain signal transmission and source location in a waveguide

Abstract: The research on transmission and filter operations has two purposes. The first purpose is to reduce channel distortion and the second is to locate a source. It uses impulse responses of transmissions between a source and a receiver in a waveguide. A time reversal of the impulse response gives the matched filter for the transmission. The matched filter is an optimum filter for the reduction of transmission distortion. A numerical example demonstrates distortion reduction. Since the impulse response between a source and receiver is a function of waveguide structure, source position, and receiver position, one can use the impulse response to determine source position by passing the received signal through a set of reference or theoretical impulse response functions for trial source positions. The ambiguity of source locations depends on waveguide structure and the complexity of the impulse response. Basically ambiguity decreases as complexity increases. A plane‐layered waveguide requires at least three receivers that are not in a line. The signal processing consists of match filtering each receiver for a trial source location and then cross correlating the trial filter outputs. The pairwise cross correlations have a maximum when the trial location is the same as the source position. The cross correlation at match also gives an estimate of the autocorrelation of the source. The source can radiate impulsive transients or continuous random signals.

Polyphase filter matrix for rational sampling rate conversions

Abstract: The polyphase filter array has been used for efficient implementations of filters with integer sampling rate conversions. [1] The filter in the high sampling rate side is decomposed into its polyphase filters which can be moved to the lower sampling rate side without changing their functions. For FIR filters the computational complexity is reduced by a factor equal to the sampling rate ratio. A rational (L/M) sampling rate conversion system realized with a 1-to-L interpolator followed by an M-to-1 decimator has three sampling rates F, LF and (L/M)F involved. By using the polyphase filter array a filter operating at the sampling rate of LF can be implemented in either the input side or the output side with lower sampling rates. The polyphase filter matrix structure will operate at the sampling rate of F/M, which does not show in the above model and is lower than any one of those three rates. For FIR filters the computational complexity is reduced by a factor of LM compared to the direct realization of the integral filter or by a factor of M (or L) compared to the polyphase filter array realization while the system input-output relation is maintained.

Simplified analysis of the LMS adaptive filter using a transfer function approximation

Abstract: A simple approximation to the behavior of the LMS adaptive filter as a discrete transfer function is developed. This representation is a valid description for both deterministic inputs and for the expected results with random inputs (including correlated inputs). The results are shown to be exact for some classes of input including periodic signals. One result of this analysis is the demonstration that the LMS filter can produce results which are biased from the least-squares solution under the combined conditions of a nonzero mean primary and a correlated reference input.

Sub-band speech analyzing and synthesizing device

Abstract: In a sub-band speech analyzing and synthesizing apparatus, a low-pass filter comprises a nonrecursive filter. The center frequency and bandwidth of each channel are selected so that the decimated sampling period of that channel is an integer multiple of the period of the modulating signal or the demodulating signal of the channel. Modulation or demodulation is performed simultaneously with the low-pass filtering by the nonrecursive filter.

Multiplierless decimating low-pass filter for a noise-shaping A/D converter

Abstract: A method and means for filtering the quantization noise from the output of a 1-bit analog-to-digital converter (noise-shaping coder) is disclosed. The A/D utilizes oversampling of the analog input signal, and decimation of the digital output signal. The multiplierless low-pass filter is comprised of a coefficient ROM for storing the filter coefficients, a counter for addressing the memory, and a true/complement gate for selectively complementing the output of the memory in response to the 1-bit data stream from the noise-shaping coder. An accumulator sums the selectively complemented output words for all samples, and the accumulated output is then applied to the decimator. A second embodiment is also disclosed which utilizes an overlapping digital filter approach, wherein a plurality of digital multiplierless filters are overlapped to provide an arbitrary length filter capable of producing an arbitrary filter response.

System modeling using an adaptive delay filter

Abstract: In this paper, we present a special technique for modeling an unknown system. This technique requires a type of adaptive filter called an Adaptive Delay Filter [1]-[3]. This filter structure includes variable delay taps in addition to variable gains. The Adaptive Delay Filter is especially applicable to system modeling problems in which the system to be modeled has a sparse impulse response [4]-[7]. Using the standard adaptive filter to model a sparse system could require a very large filter [8], while an Adaptive Delay Filter could model the sparse impulse response with very few elements in the filter since the delay taps spread out to adapt to the unknown system. An analysis of the mean-squared error surface using this technique is included, along with the computer simulation results of the performance in modeling both the delay taps and the gains of an unknown system. A comparison of this technique with the conventional approach using Widrow's LMS algorithm will be addressed.

Tone signal processing device using a digital filter

Abstract: At least two sets of filter coefficients corresponding to different filter characteristics are interpolated by using a control signal for controlling tone color as a parameter of interpolation. Filter coefficients obtained by the interpolation are supplied to a digital filter to determine its filter characteristics and an input tone signal is modified in accordance with the filter characteristics thus determined. Filter characteristics of diverse variation as compared with the number of prepared filter coefficients can thereby be realized. Further, timewise change of filter characteristics can be realized by changing a parameter of interpolation with lapse of time or changing two sets of filter coefficients to be interpolated with lapse of time. Designation of filter coefficients can be made by designating coordinate data of coordinates having at least two axes. In this case, filter coefficients can be changed by changing coordinate data of at least one axis in accordance with tone color control information whereby filter characteristics can be variably controlled.

Parallel implementation of efficient LS algorithms for filtering and prediction

Abstract: A highly concurrent algorithm for LS FIR filtering and prediction is derived in this paper. The resulting reflection coefficients associated with the solution of Toeplitz and near to Toeplitz systems of equations are computed in an order recursive manner. This results in a highly parallel algorithm requiring O(p) computing time and O(p) processors, p being the order of the filter.

Method for enhancing the quality of coded speech

Abstract: A comb filter minimizes framing noise resulting from block encoding of speech. The comb filter has both pitch and coefficients adapted to the speech data. Block boundaries may be centered on filter segments of a fixed duration.

On the synthesis of median filter systems

Abstract: In this paper, we address the problem of median filtering synthesis, that is, the design of a median filter system realization that will satisfy a given set of specifications. In particular, we address the problem of a cascade median filter realization, where the performance of a median filter is achieved through the cascade of several median filters of smaller window size. Because of the nonlinear nature of these filters, a statistical performance criterion is chosen for the system specifications. In order to evaluate the output of the cascade filter, a method is developed which finds the statistics of the roots of median filters, where roots are signals obtained after several median filter passes. Finally, a VLSI implementation for a cascade median filter system is presented.

Noise reduction apparatus for video signal

Abstract: A disclosed noise reduction apparatus comprises a vertical filter for extracting a component of high vertical frequency from an input video signal, a two-dimensional filter for extracting a component of low in vertical frequency and high in horizontal frequency from the input video signal, a detector for detecting a vertical line image portion from output components of the vertical filter and the two-dimensional filter, and a controllable two-dimensional filter which suppresses a component of high vertical frequency at the vertical line image portion according to a detection signal from the detection. By this apparatus, the noise superposed on the vertical line image signal is effectively reduced without deteriorating the vertical resolution.

Frequency selection circuit for hearing aids

Abstract: A signal processing circuit for hearing aids includes a broadband peak detector for generating a control voltage based upon the sound pressure level of an incoming acoustical signal over its entire frequency spectrum. The control signal is used to determine the cut-off frequency of a voltage controlled adaptive high-pass filter. An amplified electrical signal, corresponding to the acoustical signal, also is provided to the high-pass filter. In setting the cut-off frequency, the control voltage causes the high-pass filter to selectively suppress the low frequency portion of the signal, generating a modified signal in which the noise component is reduced.

Tone signal generation device employing a digital filter

Abstract: An address signal generation circuit generates an address signal which changes at a rate corresponding to tone pitch of a tone to be generated. This address signal consists of an integer section (IAD) and a decimal section (FAD). A tone waveshape data generation circuit (13) generates tone waveshape sampled data in response to the integer section (IAD) of the address signal. A filter coefficient supply circuit (24) can generate data corresponding to filter coefficients of m orders and selects and supplies n filter coefficients (where n < m) in response to the decimal section (FAD) of the address signal. In a relation n = m/d, for example, the filter coefficient supply circuit (24) selects filter coefficients corresponding to n orders which are distant sequentially with an interval of d in response to the value of the decimal section of current address signal. A digital filter operation circuit (21) performs filter operations of m orders with respect to tone waveshape data of n sample points by using these n filter coefficient data and tone waveshape data of n sample points generated by the tone waveshape data generation circuit (13). By this arrangement, despite that the actual filter operation is carried out only with respect to data corresponding to the n orders, a filter operation equivalent to performing filter operations of m orders can be realized.

Real-time self-tuning deconvolution filter and smoother†

Abstract: A new self-tuning deconvolution filter/smoother is presented. The algorithm runs in real time on a TMS32010 microprocessor. The filter/smoother employs a moving average of the innovations sequence to estimate the required signal. An application to speech processing and to the channel equalization problem is described.

Quadratic Tasks And The Ideal Observer

Abstract: The conventional whitening matched filter is linear in the data, even for edge-detection and object-location tasks. We have considered some special cases of the next order, or quadratic, matched filter which is second order in the data. Whereas integrals of NEQ-like quantities determine the performance of the linear filter, integrals of squared NEQ-like quantities determine the performance of the nonlinear filter. In the low contrast limit the NEQ-like quantities are precisely NEQ (noise equivalent quanta), and otherwise can be found by the Karhunen-Loeve transformation. The higher power means that these tasks are more sensitive to the higher frequency response of the hardware than are the linear tasks. Whether the human observer is capable of such quadratic tasks is an interesting open question.

Experimentation with an adaptive noise-cancellation filter

Abstract: The presence of noise in speech has severely adverse effects on speech produced by a low-bit-rate voice terminal. This paper describes an adaptive noise-cancellation filter that was designed to suppress the broad-band, nonstationary, and intense noise often encountered in military tracked vehicles, helicopters, and high-performance aircraft. This adaptive noise-cancellation filter was developed for real-time operation using a TMS32010 microprocessor. The filter was tested under various environmental conditions with a wide range of filter parameters. According to our measurements, it reduces the noise floor by 10 to 15 dB without degrading the voice quality.

Estimation of impulse‐response signals through empirically derived digital matched filter smoothing

Abstract: A real‐time digital filter is described which may be most useful for optimal determination of the magnitude of impulse‐response functions found in pulsed, repetitive experiments of low duty cycle. This filter is based on a matched filter but employs an interference orthogonalization step. This results in a signal magnitude estimate which is independent of coherent interference. The filter updates the signal magnitude estimate upon each repetition of the experimental cycle. Comparisons to signal estimation using gated sampling devices are given.

Microwave multiband filter

Abstract: A multiband microwave filter that can selectively transmit a broadband signal or any discret or combination of predefined frequency bands. The filter has two transmission lines, with an output port at one end of a first one of the transmission lines. A switch selectively transmits an input signal to one of the transmission lines. The filter also has a plurality of narrowband directional filters, each of which is used to transmit signals in a corresonding frequency band from one of the transmission lines to the other. Each directional filter includes at least one diode for enabling and disabling the operation of the direction filter in accordance with a bias voltage applied to the diodes. By controlling the switch and applying appropriate bias voltages to each of the diodes, the microwave filter can selectively transmit a broadband signal or any discret or combination of the frequency bands transmitted by the directional filters.

Suppression and detection of impulse type interference using adaptive median hybrid filters

Abstract: In this paper, we introduce a new type of nonlinear filters, the Adaptive Median Hybrid (AMH) filters, for the suppression and detection of short duration interferences. In the AMH filters, adaptive filter substructures are used to estimate the current signal value from the future and past signal values. The output of the overall filter is the median of the adaptive filter outputs and the current signal value. This kind of nonlinear filter structure is shown to adapt and preserve rapid changes in signal characteristics well. However, it filters out short duration interferences. By examining the difference between the original and filtered data, interferences can be detected. We introduce two types of AMH filters, the AMH filter with separate adaptive substructures (SAMH) and the AMH filter with coupled substructures (CAMH), which have different convergence properties and implementation. We use both synthetic and real data (speech and electroencephalogram (EEG)) to show the applicability of the proposed filters.

Method and apparatus for estimating signal components of a filter output

Abstract: An adaptive matched filter is described as a replacement for a signal replica filter of a maximum likelihood (ML) demodulator, the filter normally receiving an input data vector having signal and noise components. A symbol-justified input data vector is produced from the input data vector. This vector is then used to generate a weight vector. In a time-domain embodiment, the symbol-justified data vector is multiplied by the weight vector on a symbol-by-symbol basis to estimate the signal components of the output of a filter. In a frequency-domain implementation, multiplication is carried out on a symbol block-by-symbol block basis.

Adaptive scanning line converting circuit for a television receiver

Abstract: A circuit and technique for using only a line memory that has small capacity and selecting a proper interpolation filter according to the correlationship of vertical and slanting directions between two scanning lines of input signals. This system performs the steps of applying interlaced scanning signals to an image signal delaying circuit having delaying elements, generating a plurality of signals according to each delay, filtering in a two dimensional space filter, detecting the correlationship of the vertical and slanting directions between the lines by logic operations of the delay signal, selecting a proper space filter of the two dimensional space filter in response to the detected signal and converting in time axis with the interlaced scanning signals and a signal properly generated through the interpolation filter of a filter selector, thereby economically obtaining a high quality picture image.

Digital circuit for sampling rate variation and signal filtering and method for constructing the circuit

Abstract: A digital circuit for sampling rate variation and signal filtering includes an input, an output, a lattice wave digital filter having a plurality of filter branches connected to the input, the filter branches each having at least two series-connected filter subgroups with basic filter elements each formed of one two port adaptor made up of adders and multipliers and one time-lag device, a device disposed between the at least two filter subgroups for varying the sampling rate and for generating a phase change in a digital system, and an adder connected between filter branches and the output. A method for constructing the circuit is also provided.

Digital FM demodulator using digital quadrature filter

Abstract: A frequency-modulated digital signal demodulator includes a digital quadrature filter having n elements arranged in cascade, each delaying the input signal by a value T E =1/F E , where F E is the sampling frequency, a number of channels arranged in parallel with the cascade arrangement of delay elements, each channel having a multiplier for multiplying by a given coefficient, a first and a second summing circuit for summing the respective output signals of said multipliers, the output signals of the two summing circuits constituting the quadrature output signals of the filter and being referred to as reference and phase-shifted signals. At the output of the filter a circuit for calculating the instantaneous phase φ n of successive signal samples is provided. In order to provide a demodulator with an improved threshold and with a digital quadrature filter having an identical amplitude-frequency response for the quadrature channels and, with a view to reducing the cost, a smaller number of electronic components than that in prior art arrangements is used. The demodulator according to the invention includes, in parallel with the circuit for calculating the instantaenous phase, a circuit for evaluating the signal-to-noise ratio which is provided with a control circuit for modifying the coefficients of the quadrature filter when said ratio drops below a given threshold.

Spread spectrum communications systems and transmitters and receivers therefor

Abstract: In a spread spectrum communications system, a filter is inserted in a transmitter and a receiver, respectively. The filter in the transmitter is an FIR filter and the filter in the receiver has a characteristic inverse to that of the FIR filter.