814 results about "Finite impulse response" patented technology

A system for providing an accurate prediction of a signal-to-interference noise ratio is described. The system includes a first circuit for receiving a signal transmitted across a channel via an external transmitter. A second circuit generates a sequence of estimates of signal-to-interference noise ratio based on the received signal. A third circuit determines a relationship between elements of the sequence of estimates. A fourth circuit employs the relationship to provide a signal-to-interference noise ratio prediction for a subsequently received signal. In the illustrative embodiment, the inventive system further includes a circuit for generating a data rate request message based on the signal-to-noise ratio prediction. A special transmitter transmits the data rate request message to the external transmitter. In the specific embodiment, the relationship between elements of the sequence of estimates is based on an average of the elements of the sequence of estimates. The third circuit includes a bank of filters for computing the average. The bank of filters includes finite impulse response filters. Coefficients of the transfer functions associated with each filter in the bank of filters are tailored for different fading environments. The different fading environments include different Rayleigh fading environments, one environment associated with a rapidly moving system, a second environment associated with a slow moving system, and a third system associated with a system moving at a medium velocity. A selection circuit is connected to each of the filter banks and selects an output from one of the filters in the filter bank. The selected output is associated with a filter having a transfer function most suitable to a current fading environment.

An equalization system and method. In a most general sense, the inventive equalization system includes first and second filters for filtering an in-phase component of a received signal in accordance with first and second sets of coefficients, respectively. The system includes third and fourth filters for filtering a quadrature component of the input signal in accordance with third and fourth sets of coefficients, respectively. The outputs of the first and third filters are subtracted to provide an equalized in-phase output signal and the outputs of the second and fourth filters are added to provide an equalized quadrature output signal. In the illustrative embodiment the filters are finite impulse response filters and the coefficients are provided by a microprocessor. In accordance with the present teachings, the filters are implemented in a general purpose filter. The delay elements of the filters are calculated in accordance with a mean square error algorithm. Accordingly, the coefficients are the product of the correlation between inputs to the delay elements and a cross correlation between the inputs and a set of values representative of a desired response.

Methods and apparatus for signal processing are disclosed. A discrete time domain input signal x_m(t) may be produced from an array of microphones M₀ . . . M_M. A listening direction may be determined for the microphone array. The listening direction is used in a semi-blind source separation to select the finite impulse response filter coefficients b₀, b₁ . . . , b_N to separate out different sound sources from input signal x_m(t). One or more fractional delays may optionally be applied to selected input signals x_m(t) other than an input signal x₀(t) from a reference microphone M₀. Each fractional delay may be selected to optimize a signal to noise ratio of a discrete time domain output signal y(t) from the microphone array. The fractional delays may be selected to such that a signal from the reference microphone M₀ is first in time relative to signals from the other microphone(s) of the array. A fractional time delay Δ may optionally be introduced into an output signal y(t) so that: y(t+Δ)=x(t+Δ)*b₀+x(t−1+Δ)*b₁+x(t−2+Δ)*b₂+ . . . +x(t−N+Δ)b_N, where Δ is between zero and ±1.

A method and spread spectrum transceiver for demodulating a spread spectrum signal is disclosed. A spread spectrum phase shift keyed (PSK) modulated information signal is received within a demodulator of a spread spectrum receiver on a signal channel. The information signal includes a sequence of data symbols formed from a plurality of high rate mode chips. A precursor portion of the signal channel is Viterbi detected. A multi-state trellis is formed having a predetermined number of states. A post-cursor portion of the signal channel is feedback equalized with a finite impulse response filter having feedback taps operatively connected to a chip detection circuit that tracks high rate mode chips and a carrier loop circuit for phase and frequency tracking. The information signal is despread within a spread spectrum code function correlator.

An iterative method (400) and apparatus (200) for a receiver for reducing interference in a desired signal in a GSM communication system uses a finite-impulse-response filter combined with alternate quadrature component output selection for alternate linear equalization are disclosed The method includes inputting a burst of data of a received waveform including interference, training an alternate linear output filter with a midamble of known quadrature phase, providing an estimate of the desired signal by operating on the received waveform with the finite-impulse-response filter, generating log likelihood ratio estimates for a plurality of bits in the burst of data, selecting bits from the burst of data base upon a predetermined condition, and re-training the alternate linear output filter to provide a second improved estimate of the desired signal.

A method and apparatus of compensating for compact digital domain chromatic dispersion. The distortion of an optical signal due to chromatic dispersion is compensated by a digital signal processing in the electrical domain, either prior to the optical transmitter or following the receiver. The circular coefficient approximation and sub-band processing reduce the amount of computations to execute a given level of chromatic dispersion compensation compared to a direct finite impulse response filter implementation.

The invention provides a microphone array voice enhancement device with a sound source direction tracking function and a method for the microphone array voice enhancement device. The microphone array voice enhancement device relates to voice signal processing. A microphone array, an adjustable parallel beam former group module, a fixed parameter FIR (Finite Impulse Response) filter module, a fixed parameter signal blocking module, an adaptive noise canceller module and a sound source direction update module are aranged in the devide. The method comprises the following steps of: initializing, forming an adjustable beam, fixing parameter filtering, blocking a signal, carrying out adaptive noise cancelling, and updating a sound source direction. The invention provides an adjustable parallel beam former group combined with a sidelobe canceller structure to achieve real-time tracking of a target sound source direction, embeds the sound source direction tracking function directly to a generalized sidelobe canceller structure, can achieve sound source direction tracking and voice enhancement, thereby overcoming sensibility of an algorithm performance on a DOA (Direction of Arrival) estimation error.

A highly linear analog-to-digital (ADC) conversion system has an analog front-end device in cascade with a standard ADC converter, and a tunable digital non-linear equalizer. The equalizer corrects the quantization distortion, deviations from ideal response, and additive noises generated by the analog front-end device and ADC converter. The equalizer is formed by three main parts: Generate Function Streams Unit, Finite Impulse Response FIR filters and a summer. The equalizer receives the unequalized output from the ADC converter and generates a plurality of monomial streams in a systolic fashion. Each of the monomial streams is passed through a corresponding linear finite impulse response FIR filter. A convolution sum of all outputs from the FIR filters produces a unique equalized output with the non-linear distortion reduced to a satisfactory level. The FIR filter coefficients are determined by an Identity Equalizer Coefficient Unit, and a Test Signal Generator with different types of test signals. The FIR filter coefficients are set to minimize an error function.

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.

Disclosed is a method and system that adapts coefficients of taps of a Finite Impulse Response (FIR) filter to increase elimination of Inter-Symbol Interference (ISI) introduced into a digital communications signal due to distortion characteristics caused by a real-world communications channel. In the communications system there is a Finite Impulse Response (FIR) filter. The FIR filter has at least one pre and/or post cursor tap that removes pre and/or post cursor ISI from the signal, respectively. The pre/post cursor taps each have pre/post cursor coefficients, respectively, that adjusts the effect of the pre/post cursor portion of the FIR filter. The FIR filtered signal is transmitted over the channel which distorts the signal due to the changing and/or static distortion characteristics of the channel. The channel distorted signal is received at a receiver that may pass the channel distorted signal through a quantifier/decision system (e.g., a slicer) as the quantifier input signal to quantify the quantifier input signal to one of multiple digital values. The channel distorted signal may be further adjusted by summing the channel distorted signal with the output of a Decision Feedback Equalizer (DFE) filter to create a DFE corrected signal which then becomes the quantifier input signal. An error signal is determined by finding the difference between the scaled quantifier decision and the quantifier input signal. The pre/post cursor coefficient values that adjust the effects of the pre/post cursor taps of the FIR filter are updated as a function of the error signal and at least two quantifier decision values, and update coefficient values, may be sent over a communications back-channel to the FIR filter.

Methods systems and system components for optimizing contrast resolution of an imaging or sensing system utilizing multiple channels of broadband data associated with an array of transducers. Channels or data are filtered by passing the channels of data through finite impulse response (FIR) filters on each channel. The filters each have multiple taps having tap weights pre-calculated as a function of distance of the array from an object that energy is being transmitted to or reflected from. The weights are pre-computed through a deterministic equation based on an a priori system model.

A Finite Impulse Response (FIR) filter is implemented in software on a general purpose processor in a manner which reduces the number of memory accesses as compared to conventional methods. In particular, an efficient implementation for a general purpose processor having a substantial number of registers includes inner and outer loop code which together make memory accesses and KN multiply-accumulates, where L1 is the number of output vector elements computed during each pass through the outer loop and where L2 is the number of taps per output vector element computed during each pass through the inner loop. The efficient implementation exploits L1+2L2 general purpose registers. For an embodiment in which L1=L2=8, inner and outer loop code make memory accesses, which for filter implementations with large numbers of taps, approaches a 4x reduction in the number of memory accesses as compared to conventional methods.

An embodiment of the present invention provides an automatic gain control system for a wireless receiver that quickly differentiates desired in-band signals from high power out-of-band signals that overlap into the target band. The system measures power before and after passing a received signal through a pair of finite impulse response filters that largely restrict the signal's power to that which is in-band. By comparing the in-band energy of the received signal after filtering to the total signal energy prior to filtering, it is possible to determine whether a new in-band signal has arrived. The presence of this new in-band signal is then verified by a multi-threshold comparison of the normalized self-correlation to verify the presence of a new, desired in-band signal.

A communication transmitting and receiving system in which the effects of near-end echo and near-end crosstalk signal from the communication medium are mitigated by adaptively reproducing the near-end echo and near-end crosstalk signal, which is then subtracted from the received signal. Filter coefficients for a Finite Impulse Response filter are adaptively generated to reproduce the near-end echo and near-end crosstalk. The filter coefficients are regenerated for the Finite Impulse Response filter in an adaptive correlator at the arrival of each received signal and whereby each new filter coefficient is a weighted sum of a previous coefficient and one received signal multiplied by a time delayed version of one transmitted signal.

A method and apparatus is disclosed for performing blind source separation using convolutive signal decorrelation. For a first embodiment, the method accumulates a length of input signal (mixed signal) that includes a plurality of independent signals from independent signal sources. The invention then divides the length of input signal into a plurality of T-length periods (windows) and performs a discrete Fourier transform (DFT) on the, signal within each T-length period. Thereafter, estimated cross-correlation values are computed using a plurality of the averaged DFT values. A total number of K cross-correlation values are computed, where each of the K values is averaged over N of the T-length periods. Using the cross-correlation values, a gradient descent process computes the coefficients of a finite impulse response (FIR) filter that will effectively separate the source signals within the input signal. A second embodiment of the invention is directed to on-line processing of the input signal—i.e., processing the signal as soon as it arrives with no storage of the signal data. In particular, an on-line gradient algorithm is provided for application to non-stationary signals and having an adaptive step size in the frequency domain based on second derivatives of the cost function. The on-line separation methodology of this embodiment is characterized as multiple adaptive decorrelation.

A signal processing system includes an analog-to-digital delta sigma modulator with a duty cycle modulator and a finite impulse response (FIR) filter in a main loop feedback path of the delta sigma modulator. The duty cycle modulator and FIR filter can provide high performance filtering in the main loop feedback path. To prevent instability in the main loop caused by the duty cycle modulator and FIR filter, the delta sigma modulator also includes a stabilizer loop. Transfer functions of the main loop and the stabilizer loop combine to achieve a target transfer function for the analog-to-digital delta sigma modulator that provides for stable operation of the analog-to-digital delta sigma modulator. In at least one embodiment, the stabilizer loop includes a stabilizer path that provides output data directly to an integrator of the main loop filter.

A DSP-enabled system and method for increasing intelligibility of audio on amplified telephones by providing digital audio processing customizable based on characteristics of hearing loss specific to individual end users are disclosed. The DSP-enabled amplified telephone generally comprises a DSP capable of implementing at least one digital processing mode for processing audio input, a volume control for allowing the user to select a volume control level, and a controller for interfacing between the DSP and the volume control. The DSP is optionally programmable to implement a processing mode selected from multiple processing modes that the DSP is capable of implementing. The mode is selected based upon the user's hearing loss characteristics. The DSP may be further customized by being programmed with frequency response, compression ratio, and/or knee point according to hearing loss characteristics of the user. The DSP may implement digital TILL, BILL, and/or PILL (treble, bass, and programmable increase at low levels, respectively), all of which apply more amplification to softer inputs than to louder inputs, as well as input compression, output compression, and/or finite impulse response (FIR) filter tone control.

A data communications system and methods are disclosed. The system includes a transmitter for conveying a data signal filtered by a finite impulse response (FIR) filter to a receiver via a channel. The receiver equalizes the received data signal using a decision feedback equalizer (DFE) and the FIR. The receiver samples the data signal to determine an error signal and uses the error signal to adapt settings of a pre-cursor tap coefficient of the FIR, one or more post-cursor tap coefficients of the FIR, a phase of the recovered clock, and a coefficient of the DFE. To adapt the settings, the receiver determines the error signal based on an error sample taken from the data signal in a single clock cycle. To determine an error signal, the receiver samples the data signal at a phase estimated to correspond to a peak amplitude of a pulse response of the channel.