490 results about "Infinite impulse response" patented technology

A cardiac rhythm management system includes an atrial pacing preference (APP) filter for promoting atrial pacing. The APP filter includes an infinite impulse response (IIR) or other filter that controls the timing of delivery of atrial pacing pulses. The atrial pacing pulses are delivered at an APP-indicated pacing rate that is typically at a small amount above the intrinsic atrial heart rate. For sensed beats, the APP indicated rate is increased until it becomes slightly faster than the intrinsic atrial heart rate. The APP-indicated pacing rate is then gradually decreased to search for the underlying intrinsic atrial heart rate. Then, after a sensed atrial beat, the APP filter again increases the pacing rate until it becomes faster than the intrinsic atrial rate by a small amount. As a result, most atrial heart beats are paced, rather than sensed. This decreases the likelihood of the occurrence of an atrial tachyarrhythima, such as atrial fibrillation. The decreased likelihood of atrial tachyarrhythmia, in turn, decreases the likelihood of inducing a ventricular arrhythmia, either as a result of the atrial tachyarrhythmia, or as the result of delivering a defibrillation shock to treat the atrial tachyarrhythmia.

A vibration data collection system performs an integration or differentiation process on incoming digitized vibration data in real time. The system uses a digital Infinite Impulse Response (IIR) filter running at the input data rate to provide the integration or differentiation function. With this approach, the system reduces hardware complexity and data storage requirements. Also, the system provides the ability to directly integrate or differentiate stored time waveforms without resorting to FFT processing methods.

Multiple Steiner codes are transmitted as bursts from multiple base stations (182, 184, 186) having one or more transmit elements (174, 176, 178, 180), with successive bursts providing an extended training sequence for use in channel estimation at an addressed unit (172), such as a mobile handset. Accurate channel estimation is possible through the use of Wiener frequency domain MMSE deconvolution (518) combined with frequency domain spatial decoupling matrices, with quasi-orthogonal pseudo-noise sequences (502, 504, 520, 522) allocated to base stations and their antenna elements. The use of Steiner codes to supplement Wiener frequency domain MMSE deconvolution and frequency domain spatial decoupling results in the possibility of allocating only a single training sequence to each base station provided that the training sequence is of sufficient length to encompass all multiple time-translated channel impulse responses (H). Estimates may be refined iteratively by minimising the MS error of demodulated pilot symbols. Estimates may also be refined by removing taps from the impulse response which are insignificant based on a relatively long-term power-delay profile for the channel.

A versatile signal processor. The inventive signal processor includes a plurality of filters which are selectively interconnected to provide a variety of digital signal processing functions. In the illustrative embodiment, each filter is adapted to multiply input data by a coefficient. Further, each filter includes adders for accumulating the products. The coefficients are provided by an external processor which configures the general purpose filter to a particular function, such as a general purpose filter, a Hilbert filter, a finite impulse response filter, an equalizer, a beamforming network, a convolver, a correlator, or an application specific integrated circuit by way of example. When interconnected in accordance with the teachings provided herein, these circuits may be used to provide a digital receiver.

A clock reconstruction mechanism for synchronous TDM communications traffic transported over asynchronous networks such as Ethernet networks. The invention is applicable to edge switches in Metropolitan Area Networks (MANs) that transport legacy TDM traffic using a Circuit Emulation Services (CES) module whereby TDM traffic is encapsulated and transported across the Ethernet network where it is de-encapsulated and clocked out to the destination. The mechanism encapsulates the input TDM data stream into Ethernet packets and inserts a network timestamp within the packet. At the destination CES, a local timestamp is generated for each received packet as it is received. The network timestamp is extracted and input along with the local timestamp to a Digital Time Locked Loop (DTLL) which is operative to accurately reconstruct the original transmit TDM clock. The filter in the DTLL performs a Least Squares Regression (LSR) algorithm and Infinite Impulse Response (IIR) filter algorithm to generate a clock control signal for adjusting the clock generated.

Multiple Steiner codes are transmitted as bursts (s11, S12, . . . S33, 560, 524) from multiple base stations (182, 184, 186) having one or more transmit elements (174, 176, 178, 180), with successive bursts providing an extended training sequence for use in channel estimation at an addressed unit (172), such as a mobile handset. Accurate channel estimation is possible through the use of Wiener frequency domain MMSE deconvolution (518) combined with frequency domain spatial decoupling matrices, with quasi-orthogonal pseudo-noise sequences (502, 504, 520, 522) allocated to base stations and their antenna elements. The use of Steiner codes to supplement Wiener frequency domain MMSE deconvolution and frequency domain spatial decoupling results in the possibility of allocating only a single training sequence to each base station provided that the training sequence is of sufficient length to encompass all multiple time-translated channel impulse responses (H).

Channel estimation techniques are provided for a receiver of a wireless communication system using orthogonal frequency division multiplexing (OFDM), including legacy 802.11a and various extended rate systems. Training signals are received from one or more receive antennas. An estimated channel impulse response is computed from the received training signals by reference to a training sequence. The estimated channel impulse response is truncated in the time domain based on channel power and noise data. Channel response tracking techniques may also be implemented to correct for variations in channel response over the transmission time of a packet.

According to one embodiment of the invention, a programmable finite impulse response (FIR) filter is implemented with differential isolation circuits to isolate parasitic capacitance from attenuating an output signal at both a first and second differential output terminals of the FIR filter. The FIR includes a summing circuit that provides operational advantages to the FIR filter.

Generally, the present invention determines and uses spectral peak information, which preferably augments feature vectors and creates augmented feature vectors. The augmented feature vectors decrease errors in pattern recognition, increase noise immunity for wide-band noise, and reduce reliance on noisy formant features. Illustratively, one way of determining spectral peak information is to split pattern data into a number of frequency ranges and determine spectral peak information for each of the frequency ranges. This allows single peak selection. All of the spectral peak information is then used to augment a feature vector. Another way of determining spectral peak information is to use an adaptive Infinite Impulse Response filter to provide this information. Additionally, the present invention can determine and use incremental information. The incremental information is relatively easy to calculate and helps to determine if additional or changed features are worthwhile. The incremental information is preferably determined by determining a difference between mutual information (between the feature vector and the classes to be disambiguated) for new or changed feature vectors and mutual information for old feature vectors.

A method for determining the reverse link data Rate Limit for mobile stations active on the reverse link of a High Data Rate system is disclosed. In the ideal case, the Rate Limit is based on only the number of mobile stations located in a common sector that are actually active on the reverse link. Currently, the Rate Limit is determined from the total number of mobile stations in a common sector where the total includes mobiles that are transmitting and receiving. Thus, the current method includes mobile stations that are active on the forward link and may not be active on the reverse link. In this invention, a more optimum method of estimating the reverse link loading is obtained from calculations which includes only the mobile stations which are active on the reverse link. An estimate of the reverse link loading of the mobile stations in a common cell is obtained by adding together the data rates of the data sent from each mobile in a common sector during a common frame. This aggregate rate of data during the frame is filtered to minimize irregularities by using the moving average of an infinite impulse response filter and then normalized. The normalized result is a percentage of the maximum achievable aggregate reverse link rate. The final result is compared with a set of threshold values to obtain the maximum Rate Limit that is then set for each mobile station.

Initial values of the tap weights for the taps of a linear equalizer are determined based on a channel impulse response of a channel so that the values corresponding to the weights of the equalizer taps achieve optimum initialization of the equalizer. These values are determined through use of a nested summation where the number of summations is dependent upon the number of multi-paths characterizing the channel.

A method and apparatus for extracting and changing the reverberant content of an input signal is provided. The method of estimating an impulse response for a reverberant system includes sampling a signal from a reverberant system, having an impulse response, into a sequence of blocks, for each block of the sequence, determining a magnitude in dependence upon a magnitude of a previous block, forming an estimate of the impulse response using determined magnitudes as parameters for a finite impulse response filter. The method of creating a multi-channel signal using a signal with fewer channels from a reverberant system includes sampling the signal from a reverberant system, having an impulse response, into a sequence of blocks, for each block of the sequence, determining a magnitude in dependence upon a magnitude of a previous block, forming an estimate of the impulse response using determined magnitudes as parameters for a finite impulse response filter, applying an inverse of the estimated impulse response to the signal to derive at least one of a direct signal component and a reverberant signal component, and recombining the direct signal component and the reverberant signal component to create a multi-channel signal.

A low power programmable digital filter adapted for use with a telecommunications system transceiver. The digital filter includes a first finite impulse response filter section for receiving an input signal and having a first transfer function. An infinite impulse response filter section is connected to the first finite impulse response filter section and has a second transfer function. A second finite impulse response filter section is connected to the infinite impulse response filter section and outputs a filtered output signal in response the receipt of the input signal by the programmable digital filter. The second finite impulse response filter section has a third transfer function. A programmable coefficient is included in the first, second, and / or the third transfer function. In a specific embodiment, the first transfer function has a first programmable coefficient. The second transfer function has a second programmable coefficient and the third transfer function has a third programmable coefficient. The programmable digital filter further includes a processor for providing a control signal. A memory provides the first, second, and / or third programmable coefficients in response to the control signal. A high-pass filter section provides input to the first finite impulse response filter section. A multiplexer selectively bypasses the high-pass filter in response to a bypass control signal from the processor. The processor generates the bypass control signal in response to bias occurring in the input signal. In the illustrative embodiment, the first finite impulse response filter section includes a first jammer filter, a second jammer filter, and a third jammer filter for removing telecommunications jammer signals in the input signal. The first, second, and third jammer filters have first, second, and third jammer filter transfer functions with the first programmable coefficient, a fourth programmable coefficient, and a fifth programmable coefficient, respectively. The infinite impulse response filter section includes a first equalization filter and a second equalization filter.

A novel gear shifting mechanism operative to adjust the loop gain of a phase locked loop (PLL) circuit in a continuous and reversible manner. The loop gain can be increased to widen the bandwidth of the loop and can also be decreased to narrow the loop bandwidth. The mechanism incorporates an α gear shift circuit, a p gear shift circuit and an optional IIR gear shift circuit. The α gear shift circuit comprises a infinite impulse response (IIR) filtering which enables hitless operation of the PLL loop at the occurrence of gear shift events. The α gear shift circuit comprises an accumulator whose output is multiplied by the gain value ρ. The invention enables multiple gear shifts in either positive or negative direction to be achieved by configuring the loop gain variables α and ρ which may be accomplished in software.

A cardiac rhythm management system is capable of treating irregular ventricular heart contractions, such as during atrial tachyarrhythmias such as atrial fibrillation. A first indicated pacing interval is computed based at least partially on a most recent V—V interval duration between ventricular beats and a previous value of the first indicated pacing interval. Pacing therapy is provided based on either the first indicated pacing interval or also based on a second indicated pacing interval, such as a sensor-indicated pacing interval. A weighted averager such as an infinite impulse response (IIR) filter adjusts the first indicated pacing interval for sensed beats and differently adjusts the first indicated pacing interval for paced beats. The system regularizes ventricular rhythms by pacing the ventricle, but inhibits pacing when the ventricular rhythms are stable.

A linearizer and method. In a most general embodiment, the inventive linearizer includes a characterizer coupled to an input to and an output from said circuit for generating a set of coefficients and a predistortion engine responsive to said coefficients for predistorting a signal input to said circuit such that said circuit generates a linearized output in response thereto. In a specific application, the circuit is a power amplifier into which a series of pulses are sent during an linearizer initialization mode of operation. In a specific implementation, the characterizer analyzes finite impulse responses of the amplifier in- response to the initialization pulses and calculates the coefficients for the feedback compensation filter in response thereto. In the preferred embodiment, the impulse responses are averaged with respect to a threshold to provide combined responses. In the illustrative embodiment, the combined responses are Fast Fourier Transformed, reciprocated and then inverse transformed. The data during normal operation is fed back to the data capture, corrected for distortion in the feedback path from the output of the amplifier, converted to basedband, synchronized and used to provide the coefficients for the predistortion linearization engine. As a result, in the best mode, each of the coefficients used in the predistortion linearization engine can be computed by solving the matrix equation HW=S for W, where W is a vector of the weights, S is a vector of predistortion linearization engine outputs, and H is a matrix of PA return path inputs as taught herein.

A network receiver is configured for receiving a modulated carrier signal representing a data frame from a network transmitter via a network medium, the receiver includes an adaptive equalizer with a finite impulse response filter for filtering the received signal. The filter utilizes a plurality filter coefficients. A cache stores a plurality of sets of coefficients for use by the filter and the equalizer selects a set of coefficients for receiving the data frame.

The invention belongs to the field of wireless communication channel estimation, particularly relates to a compressed sensing wireless communication channel estimation method based on sparsity self-adapting, which includes the ssteps: (1) collecting demodulated receiving signals and calculating channel response of a pilot frequency position; (2) constructing a measurement matrix phi required by signal reconstruction; (3) calculating an association degree vector and sequencing elements of the vector; (4) calculating second difference vector of a novel association degree vector after sequencing and setting a threshold value I for judging sparsity of signals; (5) estimating sparsity S of channel impulse response; (6) comparing the threshold value I with the last element of a vector D sequentially, and a coefficient value corresponding to the first element larger than the threshold value is the estimated sparsity S of the signals; and (7) reconstructing the signals. The channel estimation method breaks a bottleneck of a traditional compressed sensing algorithm that the sparsity of the signals must be known, and signal reconstruction of sparsity self-adapting is achieved.

Signal conversion of an input signal can be achieved by processing portions of the signal through plural parallel paths, which collectively approximate a desired infinite impulse response (IIR) filter, either alone or implemented with other signal processing functions. In one aspect, each of the paths can perform filtering, noise-shaping and / or quantization on a respective portion of the input signal to provide a corresponding representation of the respective portion of the input signal, for example, a coarser representation at a higher data rate. The corresponding representations from the parallel paths can be aggregated and further processed in a desired manner, such as conversion to an analog signal.

A polyphase channelizer converts an intermediate frequency wideband signal into a complex signal that is sampled by parallel analog-to-digital converters having a bank of samplers respectively clocked by staggered clocking signals for respective converters for feeding I and Q quadrature samples to a polyphase filter bank of finite impulse response filters driving a fast Fourier transform processor for providing channelized digital signal outputs. The parallel analog-to-digital converters can operate at lower speeds but are parallel connected for effectively operating at required higher speeds. The channelizer channelizes the wideband signal using a polyphase clock for enabling high speed sampling and converting through low speed parallel analog to digital converters.

A symbol sequence contained in a received signal comprising a cyclic convolution of a Walsh code multiplexed signal and a channel impulse response of a multipath channel is detected using Walsh Hadamard domain equalization techniques. The method comprises converting the received signal and the channel impulse response of the multipath channel from the time domain to the WHT domain, and determining the symbol sequence based on equalizing the received signal in the WHT domain using WHT spectra of the channel impulse response to remove inter-symbol interference from the received signal due to cross-correlation between Walsh codes.

The present invention is particularly applied to serial concatenated coding and serial concatenated trellis coded modulation. In second encoding 107, which is inner coding, a sequence that is not encoded or that is encoded so as to produce a finite impulse response and a sequence that is encoded so as to produce an infinite impulse response are output. In interleaving 106 before the second encoding 107, the sequences are permuted so as not to be mixed with each other.

A reference timing signal apparatus with a phase-locked loop has a computer algorithm which adaptively models the multiple frequencies of an oscillator following a training period. The oscillation frequency of the oscillator is controlled in response to a phase detector output. The computer algorithm processes the control signal applied to the oscillator. The computer algorithm updates the characteristics of the model relating to the aging and temperature of the oscillator, using for example, a Kalman filter as an adaptive filter. By the algorithm, the subsequent model predicts the future frequency state of the oscillator on which it was trained. The predicted frequency of the model functions as a reference to correct the frequency of the oscillator in the event that no input reference timing signal is available. In a case of using pre-processing infinite impulse response filters (IIRFs) before the adaptive processor, the time delay caused by the filters are compensated after the adaptive processor. Without pre-processing IIRFs, aging and temperature update rates are adaptively controlled by dynamically changing the rates depending upon the loop condition to achieve a wider tracking bandwidth. With the model updating algorithm, oscillators of low stability performance may be used as cellular base station reference oscillator.

A recognition system (10) incorporates a filterbank analyser (16) producing successive data vectors of energy values for twenty-six frequency intervals in a speech signal. A unit (18) compensates for spectral distortion in each vector. Compensated vectors undergo a transformation into feature vectors with twelve dimensions and are matched with hidden Markov model states in a computer (24). Each matched model state has a mean value which is an estimate of the speech feature vector. A match inverter (28) produces an estimate of the speech data vector in frequency space by a pseudo-inverse transformation. It includes information which will be lost in a later transformation to frequency space. The estimated data vector is compared with its associated speech signal data vector, and infinite impulse response filters (44) average their difference with others. Averaged difference vectors so produced are used by the unit (18) in compensation of speech signal data vectors.

Asynchronous sample rate conversion for use in a digital audio receiver is disclosed. Different algorithms are applied for the upsampling and downsampling cases. In the upsampling case, the input signal is upsampled and filtered, before the application of a finite impulse response (FIR) filter. In the downsampling case, the input signal is filtered by an FIR filter, and then filtered and downsampled. The FIR coefficients of the fractional delay FIR filter are calculated by evaluation of polynomial expressions over intervals of the filter impulse response, at times corresponding to the input sample points.

The invention discloses a training sequence reconstruction-based channel estimation method and a training sequence reconstruction-based channel estimation system. The method comprises the following steps: acquiring a known channel estimation result; according to the known channel estimation result, constructing linear convolution of former frame transmitted data and a channel and the linear convolution of a transmitted training sequence and a channel impulse response; eliminating inter-block interference of data on a training sequence; obtaining a cyclic convolution, of the training sequence and the channel impulse response, serving as a reconstruction item; according to the reconstruction item, reconstructing the training sequence; and performing channel estimation by utilizing the reconstructed training sequence, and updating a channel estimation result. The method and the system of the invention can ensure that a TDS-OFDM system can also obtain relatively accurate channel estimation when maximum delay extension of the channel exceeds the guard space length of the training sequence, simultaneously improves the accuracy of the channel estimation and improves the spectrum utilization ratio and the mobility performance of the system.

A time-reversal positioning system includes a storage storing first data representing channel impulse responses derived from probe signals sent from a plurality of positions and second data representing coordinates of the positions. A data processor determines a position of a terminal device based on the stored channel impulse responses and a time-reversed signal determined based on a time-reversed version of a channel impulse response that is estimated based on a channel probing signal sent from the terminal device.

A method of operation of a receiver of a mobile station or a base station of a radio access network and corresponding equipment, the method for especial use in providing interference cancellation according to a single antenna interference cancellation (SAIC) algorithm. The method includes a step (22) in which a channel estimator (12) estimates channel impulse response components for use by a SAIC equalizer (14); in estimating the channel impulse response components, the channel estimator (12) minimizes a special cost function (Jch(h)) with respect to the components of the channel impulse response h. In some embodiments, the method uses an iterative least mean squares procedure to minimize the cost function (Jch(h)).

Systems and methods for reducing transmit echo in a transceiver are disclosed. A hybrid echo canceller includes a limited tap length FIR filter to cancel short-term echo, while an interpolated filter is used to cancel the long-tail echo. The limited tap length FIR filter adapts and calculates coefficients for each adapted tap. Taps of the interpolated filter, on the other hand, are adapted but coefficients are calculated for a subset of the taps. Various interpolation schemes may be applied to the calculated coefficients to associate a coefficient with each tap of the interpolated filter. The technique presented produces an effective filter length of N taps with a reduction in computation and signal processing resources. Preferred embodiments of the echo canceller may be construed as methods for reducing transmit echo. A preferred method includes the steps of: bifurcating a finite impulse response filter in response to the conversion rate of the filter tap coefficients; adaptively calculating and applying a filter tap coefficient to each tap of a short term portion of the bifurcated filter; adaptively calculating a subset of the filter tap coefficients of a long tail portion of the bifurcated filter; and applying a interpolation technique to the remaining set of filter tap coefficients of the long tail portion of the bifurcated filter.

Symbol timing for a wireless communication system, such as a multiple-input multiple-output (MIMO) orthogonal frequency division multiplexing (OFDM) wireless LAN system, is determined by summing the powers for an appropriate set of channel impulse responses, integrating this power summation over an appropriate window (e.g., equivalent to the guard interval), and identifying the time at which the maximum integration occurs. Depending on the implementation, symbol timing can be determined for each receiver branch individually or for all receiver branches jointly. In either case, the determined symbol timing(s) can minimize the amount of inter-symbol and inter-channel interferences that are invoked in the system.