44 results about "Filter impulse response" patented technology

A filter compressor for generating compressed subband filter impulse responses from input subband filter impulse responses corresponding to subbands, which include filter impulse response values at filter taps, includes a processor for examining the filter impulse response values from at least two input subband filter input responses to find filter impulse response values having higher values and at least one filter impulse response value having a value being lower than the higher values, and a filter impulse response constructor for constructing the compressed subband filter impulse responses using the filter impulse response values having the higher values, wherein the compressed subband filter impulse responses do not include filter impulse response values corresponding to filter taps of the at least one filter impulse response value having the lower value or include zero-valued values corresponding to filter taps of the at least one filter impulse response value having the lower value.

The present invention provides a method of characterizing a frequency response of a transmission channel between a transceiver and a subscriber unit. The method includes once per predetermined interval of time, the transceiver transmitting a signal including multiple carriers, a plurality of the carriers including training symbols, a plurality of the carriers including information symbols. The subscriber unit generates frequency response estimates at the frequencies of the carriers including training symbols, each interval of time. The frequency response estimates are converted into a time domain response generating an impulse response once per interval of time. The impulse responses are filtered over a plurality of intervals of time. A channel profile is determined from the filtered impulse responses. The channel profile is converted to the frequency domain generating a channel interpolator. The characterized frequency response is generated from the channel interpolator and the frequency response estimates. The filtering can include averaging the impulse responses over a plurality of intervals of time, accumulating the impulse responses over a plurality of intervals of time, or weighted averaging of the impulse responses over a plurality of intervals of time. The weighted averaging can be dependent upon a phase error between the impulse responses, and / or an amplitude error between the impulse responses.

A receiver is operable to detect a synchronisation position for recovering data from a set of received signal samples. The receiver comprises a filter having an impulse response matched to a predetermined characteristic of the received signal and is operable to produce an output signal which is representative of the convolution of the impulse response and the received signal samples. The receiver includes a synchronisation detector operable to detect the synchronisation position from the filter output signal, and a data detector operable to detect and recover data from the set of received signal samples from the synchronisation position provided by the synchronisation detector. The synchronisation detector is operable to represent each of the received signal samples as a positive or negative constant in dependence upon the relative sign of the signal sample, and to represent the samples of the filter impulse response as a positive or negative constant in dependence upon the relative sign of the impulse response samples, and the filter is operable to convolve the impulse response with the received signal samples by logically combining the representation of the received signal samples and the impulse response to produce the output signal. The receiver is therefore provided with a facility for detecting a synchronisation position from an output of the filter matched to a characteristic of the received signal.The output signal of the matched filter is formed by logically combining the filter impulse response and the received signal samples to form a representation of a convolution but with a substantially reduced number of computations.

A method generates constant modulus multi-dimensional modulations for coherent optical communications by first projecting points in a constellation of the code onto a Poincare sphere or its higher-dimensional hyper-sphere. By using meta-heuristic procedures, nonlinear programming and gradient search methods, constellation points in the hyper-sphere are optimized in certain criteria, such as maximizing the minimum Euclidean distance, minimizing the union bound, minimizing the bit-error rate, minimizing the required signal-to-noise ratio, maximizing the nonlinear fiber reach, maximizing the phase noise tolerance, and maximizing the mutual information. Some methods use parametric unitary space-time block codes such as Grassmannian packing, and filter impulse response as well as unitary rotation over adjacent code blocks to generate near-constant modulus waveform, not only at the symbol timing, but also over the entire time.

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.

A receiver is operable to detect a synchronisation position for recovering data from a set of received signal samples. The receiver comprises a filter having an impulse response matched to a predetermined characteristic of the received signal and is operable to produce an output signal which is representative of the convolution of the impulse response and the received signal samples. The receiver includes a synchronisation detector operable to detect the synchronisation position from the filter output signal, and a data detector operable to detect and recover data from the set of received signal samples from the synchronisation position provided by the synchronisation detector. The synchronisation detector is operable to represent each of the received signal samples as a positive or negative constant in dependence upon the relative sign of the signal sample, and to represent the samples of the filter impulse response as a positive or negative constant in dependence upon the relative sign of the impulse response samples, and the filter is operable to convolve the impulse response with the received signal samples by logically combining the representation of the received signal samples and the impulse response to produce the output signal. The receiver is therefore provided with a facility for detecting a synchronisation position from an output of the filter matched to a characteristic of the received signal. The output signal of the matched filter is formed by logically combining the filter impulse response and the received signal samples to form a representation of a convolution but with a substantially reduced number of computations.

A Multi-mode Loop Frequency modulator (10) for modulating a carrier signal according to various modulation pulse shapings and data signal to provide a modulated output signal (VCOout), comprises a waveform generator (12) and two multiplexors (14,38) to perform various pulse shaping filtering and or predistord the pulse shaping filter impulse response for either a dual port modulation or for a single port modulation. The waveform generator (12) can also provide interpolation to input frequency deviation and also provide scaling for various reference frequency clock provided by the crystal (30) and the reference divider (28). The low pass filter (22) is used for providing the PLL loop filtering and also as the smoothing filter for the high port DAC (18).

Method to perform link adaptation at the radio interfaces of an enhanced packet data cellular network handling several Modulation and Coding Schemes (MCS) for maximizing data throughput. In a preliminary off-line step the system behaviour, in terms of net throughput of the various available MCSs, is simulated for different C / l conditions. From the simulation two sets of tables are obtained, each table including upgrade and downgrade thresholds expressed in terms of Block Error Rate (BLER). Thresholds correspond to switching points from an MCs to the two available MCSs having the immediate less or more protection. The two sets of tables are referred to higher or lower diversity RF environments and are further specialized for taking into account EGPRS type II hybrid ARQ, namely Incremental Redundancy (IR). During transmission the transmitted blocks are checked for FEC and the results are sent to the network. The network continuously updates BLER using exponential smoothing. In order to achieve the correct time response, in spite of that RLC blocks can be received or not, a reliability filter is provided whose output is used to decide the weight between the new and old measurements to make the BLER filter impulse response exponentially decreasing with time. The IR efficiency is tested for each incoming block and an indicative variable IR status is filtered using the same approach used for BLER. Each actual threshold of BLER to be used in link adaptation is obtained by a linear interpolation between the tabulated threshold without IR and with perfect IR, both weighed with filtered IR status. Filtered BLER is then compared with said interpolated thresholds for testing the incoming of a MCS switching condition. Power control pursues the goal of maintaining constant QoS peak throughput per time slot(Fig 16).

A filter compressor for generating compressed subband filter impulse responses from input subband filter impulse responses corresponding to subbands, which include filter impulse response values at filter taps, includes a processor for examining the filter impulse response values from at least two input subband filter input responses to find filter impulse response values having higher values and at least one filter impulse response value having a value being lower than the higher values, and a filter impulse response constructor for constructing the compressed subband filter impulse responses using the filter impulse response values having the higher values, wherein the compressed subband filter impulse responses do not include filter impulse response values corresponding to filter taps of the at least one filter impulse response value having the lower value or include zero-valued values corresponding to filter taps of the at least one filter impulse response value having the lower value.

A data processing method is provided to enable a calculation based signal analyzer, such as an FFT based spectrum analyzer, to produce results corresponding to a swept spectrum analyzer employing a video bandwidth (VBW) filter. Once a spectrum is produced the frequency axis is replace by a corresponding time axis, so that a time domain filter, such as a video bandwidth (VBW) filter can be applied. In a first example, the filter characteristics are applied by performing an FFT on display spectrum data, which has had its frequency axis replaced by a corresponding time axis, produce frequency domain data, multiplying by the frequency response to produce a filtered version, performing an inverse FFT and replacing the time axis with the original frequency axis to produce a filtered version of the display spectrum data. In another example, a filter impulse response is computed based upon the desired filter characteristics, a convolution integral of the display spectrum data, which has had its frequency axis replaced by a corresponding time axis, and the filter impulse response is conducted, and the time axis is replace by the original frequency axis to produce filtered display spectrum data.

The present invention provides a method of characterizing a frequency response of a transmission channel between a transceiver and a subscriber unit. The method includes once per predetermined interval of time, the transceiver transmitting a signal including multiple carriers, a plurality of the carriers including training symbols, a plurality of the carriers including information symbols. The subscriber unit generates frequency response estimates at the frequencies of the carriers including training symbols, each interval of time. The frequency response estimates are converted into a time domain response generating an impulse response once per interval of time. The impulse responses are filtered over a plurality of intervals of time. A channel profile is determined from the filtered impulse responses. The channel profile is converted to the frequency domain generating a channel interpolator. The characterized frequency response is generated from the channel interpolator and the frequency response estimates. The filtering can include averaging the impulse responses over a plurality of intervals of time, accumulating the impulse responses over a plurality of intervals of time, or weighted averaging of the impulse responses over a plurality of intervals of time. The weighted averaging can be dependent upon a phase error between the impulse responses, and / or an amplitude error between the impulse responses.

A method generates constant modulus multi-dimensional modulations for coherent optical communications by first projecting points in a constellation of the code onto a Poincare sphere or its higher-dimensional hyper-sphere. By using meta-heuristic procedures, nonlinear programming and gradient search methods, constellation points in the hyper-sphere are optimized in certain criteria, such as maximizing the minimum Euclidean distance, minimizing the union bound, minimizing the bit-error rate, minimizing the required signal-to-noise ratio, maximizing the nonlinear fiber reach, maximizing the phase noise tolerance, and maximizing the mutual information. Some methods use parametric unitary space-time block codes such as Grassmannian packing, and filter impulse response as well as unitary rotation over adjacent code blocks to generate near-constant modulus waveform, not only at the symbol timing, but also over the entire time.

A method for the compensation of frequency-response mismatch errors in M-channel time-interleaved ADCs. The compensation is done through an M-periodic time-varying filter hn(k)=hn mod M(k) (2), or, equivalently, a set of M time-invariant filters hn(k), n=0, 1, . . . , M−1. The overall compensation system is constructed by determining the M filter impulse responses hn(k) through M separate matrix inversions, where the size of the matrices equals the filter impulse response length. Also, a compensated M-channel time-interleaved ADC based on and performing the method.

The invention provides a channel estimation method and device for an MIMO (multiple input multiple output) OFDM (orthogonal frequency division multiplexing) system. The method comprises the following steps of: determining the channel response of a multipath channel between a receiving antenna and a transmitting antenna based on the signal received by the receiving antenna; performing inverse Fourier transform on the determined channel response to obtain the impulse response of the multipath channel; performing iterative time-variable filtering on the impulse response of the multipath channel; and performing Fourier transform on the filtered impulse response. The invention solves the problem of lower estimation precision resulted from that the factors are not considered in the relevant technologies, thus improving the channel estimation precision.

A method of operation of a communication system includes: generating a filter impulse response and a filter time-domain data with a shortening filter; generating a filter frequency response based on the filter impulse response with a filter frequency response calculator; generating a filter frequency-domain data based on the filter time-domain data with a first process unit; and generating a raw channel impulse response with a filter frequency removal unit for removing the filter frequency response from the filter frequency-domain data.

A system and method and non-transitory computer readable medium method for filtering signals representative of microseismic events with an infinite impulse response (IIR) Wiener filter which precludes the need for statistics or prior knowledge of the signal of interest. The second-order statistics of the noise and the noisy data are extracted from the recorded data only. The criteria used to optimize the filter impulse response is the minimization of the mean square error. The IIR Wiener filter was tested on synthetic and field data sets and found to be effective in denoising microseismic data with low SNR (−2 dB).

A data processing method is provided to enable a calculation based signal analyzer, such as an FFT based spectrum analyzer, to produce results corresponding to a swept spectrum analyzer employing a video bandwidth (VBW) filter. Once a spectrum is produced the frequency axis is replace by a corresponding time axis, so that a time domain filter, such as a video bandwidth (VBW) filter can be applied. In a first example, the filter characteristics are applied by performing an FFT on display spectrum data, which has had its frequency axis replaced by a corresponding time axis, produce frequency domain data, multiplying by the frequency response to produce a filtered version, performing an inverse FFT and replacing the time axis with the original frequency axis to produce a filtered version of the display spectrum data. In another example, a filter impulse response is computed based upon the desired filter characteristics, a convolution integral of the display spectrum data, which has had its frequency axis replaced by a corresponding time axis, and the filter impulse response is conducted, and the time axis is replace by the original frequency axis to produce filtered display spectrum data.

Systems and methods for orbital angular momentum (OAM)-based multidimensional signaling with multiple-input-multiple-output (MIMO). The wireless communication is achieved using a transmitter including a plurality of orthogonal pulse shapers of a multidimensional modulator, the plurality of orthogonal pulse shapers generating a multidimensional modulated signal by imposing one or more data sequences on corresponding mutually orthogonal filter impulse responses derived from multiple orthogonal basis functions including Slepian sequences. An OAM antenna array including OAM antenna elements is used such that each of the OAM antenna elements includes an azimuthal phase shifter for shifting an azimuthal phase term of a wavefront generated by an antenna element such that the OAM antenna elements impose the multidimensional modulated signal on pre-determined OAM modes of a carrier signal corresponding to the azimuthal phase term.

The present invention provides a method of characterizing a frequency response of a transmission channel between a transceiver and a subscriber unit. The method includes once per predetermined interval of time, the transceiver transmitting a signal including multiple carriers, a plurality of the carriers including training symbols, a plurality of the carriers including information symbols. The subscriber unit generates frequency response estimates at the frequencies of the carriers including training symbols, each interval of time. The frequency response estimates are converted into a time domain response generating an impulse response once per interval of time. The impulse responses are filtered over a plurality of intervals of time. A channel profile is determined from the filtered impulse responses. The channel profile is converted to the frequency domain generating a channel interpolator. The characterized frequency response is generated from the channel interpolator and the frequency response estimates. The filtering can include averaging the impulse responses over a plurality of intervals of time, accumulating the impulse responses over a plurality of intervals of time, or weighted averaging of the impulse responses over a plurality of intervals of time. The weighted averaging can be dependent upon a phase error between the impulse responses, and / or an amplitude error between the impulse responses.

A numerical string consisting of a ratio of “−1, m, −1” or “1, m, 1” is subjected to a predetermined moving average calculation n times. A numerical string thus obtained is used as filter coefficients of a basic filter and at least one basic filter is combined in an arbitrary way for cascade connection, thereby calculating the filter coefficients of the digital filter to be obtained. This significantly reduces the number of taps and the number of multipliers used as compared to the conventional FIR filter. Moreover, by using the numerical strings “−1, m, −1” and “1, m, 1” so that the filter impulse response becomes a finite-base function, it is possible to obtain a preferable frequency characteristic having no discretization error and having a great attenuation amount out of band.

A filter adaptation unit suitable for producing a set of filter coefficients is provided including an error characterization unit for characterizing the error in a filter's impulse response. The error characterization unit generates a set of error characterization data elements associated to a newly generated set of filter coefficents. A selection unit then makes a selection between one of the newly generated set of filter coefficients and an existing set of filter coefficients at least in part on the basis of their respective sets of error characterization data elements. The selected set of filter coefficients is then released in a format suitable for use by an adaptive filter.

A method and apparatus for generating a reflection filter. The method comprises the steps of determining an averaged step response of an acquired signal and generating a spectrum response from the averaged step response. A time at which a reflection in the averaged step response begins is determined and information in the averaged step response after the determined time when the reflection begins is replaced with an ideal flat line response to generate a required step response. A spectrum response is generated from the required step response and each frequency point of the spectrum response corresponding to the required step response is divided by each frequency point of the spectrum response corresponding to the averaged step response to generate a spectrum of the reflection filter. The result of the dividing step is processed to generate a reflection filter impulse response.

A system and method for signal peak reduction in a spread spectrum communication system of the type including a filter for limiting signal bandwidth of symbols transmitted from the system. A signal peak reduction unit is provided before the filter that includes a filter predictor that predicts the effect of the filter on input symbols by using filter coefficient values corresponding to the filter impulse response function. Input symbols that are predicted to cause the output signal to exceed a predetermined peak limit value are adjusted. Several examples of suitable algorithms for calculating the necessary peak reduction to be applied to the input symbols are disclosed. The peak reduction unit provides adjusted symbols to the filter for processing and communication system output.

The invention discloses a strictly band-limited efficient modulating system based on impact filter forming. An emitting end of the system comprises a molding filter which is used for realizing a strictly band-limited communication system and improving spectrum efficiency of the system signal by reducing signal bandwidth, wherein the molding filter is an impact filter; a receiving end of the system adopts an optimal receiver which takes a pre-stored impact waveform sample sequence obtained by an expanded jumping part of the molding filter as a correlation operation template or matching filter impulse response to respectively perform matched filtering with a MPPSK receiving signal of an existing code element received by the receiving end to demodulate a corresponding sending code element after sliding fixed step length to an impact appearing position of the code elements [1,2, ...., M]; the system sending end sufficiently applies an resonance effect of the impact filter, so that the interference effect between the strictly band-limited system codes is small, the structure is simple, the complexity is low, the frequency efficiency is high and the demodulating performance is good.

Systems and methods for orbital angular momentum (OAM)-based multidimensional signaling with multiple-input-multiple-output (MIMO). The wireless communication is achieved using a transmitter including a plurality of orthogonal pulse shapers of a multidimensional modulator, the plurality of orthogonal pulse shapers generating a multidimensional modulated signal by imposing one or more data sequences on corresponding mutually orthogonal filter impulse responses derived from multiple orthogonal basis functions including Slepian sequences. An OAM antenna array including OAM antenna elements is used such that each of the OAM antenna elements includes an azimuthal phase shifter for shifting an azimuthal phase term of a wavefront generated by an antenna element such that the OAM antenna elements impose the multidimensional modulated signal on pre-determined OAM modes of a carrier signal corresponding to the azimuthal phase term.

A method of operation of a communication system includes: generating a filter impulse response and a filter time-domain data with a shortening filter; generating a filter frequency response based on the filter impulse response with a filter frequency response calculator; generating a filter frequency-domain data based on the filter time-domain data with a first process unit; and generating a raw channel impulse response with a filter frequency removal unit for removing the filter frequency response from the filter frequency-domain data.

Mechanisms for operating a wireless user equipment device, to determine the state of data scheduling activity. Such activity state (and other information such as measurements of signal interference and / or channel condition) may be used to dynamically control the selection of a high-power high-performance receiver vs. a low-power basic receiver. A first mechanism involves transitioning to a high activity state and starting a timer upon each occurrence of scheduled subframe. A transition to the low activity state occurs whenever the timer expires. A second mechanism involves filtering a sequence of binary-valued indicators corresponding respectively to a sequence of time intervals. Each indicator indicates whether a subframe is scheduled during the corresponding time interval. The filter output is compared to a threshold to determine high / low activity state. The temporal width of the filter impulse response may be increased to support fast initial response without harming the accuracy of activity-state determination in the steady state.

According to the consistency of the mathematical model between the shape modification process and the two-dimensional filtering process, the invention discloses a method for predicting and forecasting the results of ion beam polishing. The method firstly determines the removal function of the polishing process, and then determines the The equivalent low-pass filter impulse response function, and then use the surface shape detection device to perform error detection on the optical mirror, and calculate the surface shape residual error according to the detected surface shape error and the filter impulse response function, according to the calculated The surface shape residual can realize the prediction and prediction of the ion beam polishing processing results. On this basis, the variation of each spatial frequency component before and after processing (that is, the PSD spectrum) can also be predicted according to the filter impulse response function. The prediction method of the invention is not only simpler and more operable, but also makes the prediction result more intuitive, reliable and comprehensive.

The invention discloses an SOCP (Second Order Conic Programming) based broadband transmission DBF (Digital Beam Formation) interference design method. The principle though of the method is as follows:determining a uniform linear array including M array elements and broadband digital signals, branching the broadband digital signals to obtain M channels of broadband digital signals; obtaining M FIRs (Filter Impulse Responses); performing FIR filtering on each channel of the broadband digital signals through a corresponding FIR to obtain M channels of broadband digital signals subjected to FIR filtering; performing digital-to-analog conversion, up-conversion, RF filtering and amplification on each channel of broadband digital signals subjected to FIR filtering, and obtaining M channels of broadband signals subjected to amplitude and phase processing; obtaining broadband interference beams after transmitting each channel of broadband signals subjected to amplitude and phase processing tothe air through corresponding array elements, the broadband interference beams being a SOCP based broadband transmission DBF interference design result.