1582 results about "Pulse response" patented technology

In an OFDM system, a transmitter broadcasts a first TDM pilot on a first set of subbands followed by a second TDM pilot on a second set of subbands in each frame. The subbands in each set are selected from among N total subbands such that (1) an OFDM symbol for the first TDM pilot contains at least S₁ identical pilot-1 sequences of length L₁ and (2) an OFDM symbol for the second TDM pilot contains at least S₂ identical pilot-2 sequences of length L₂, where L₂>L₁, S₁·L₁=N, and S₂·L₂=N. The transmitter may also broadcast an FDM pilot. A receiver processes the first TDM pilot to obtain frame timing (e.g., by performing correlation between different pilot-1 sequences) and further processes the second TDM pilot to obtain symbol timing (e.g., by detecting for the start of a channel impulse response estimate derived from the second TDM pilot).

A MIMO OFDM system includes a plurality of space-time encoders for encoding respective data blocks with independent space-time codes. The transformed data block signals are transmitted by a plurality of transmit antennas and received by a plurality of receive antennas. The received data is pre-whitened prior to maximum likelihood detection. In one embodiment, successive interference cancellation can be used to improve system performance. Channel parameter estimation can be enhanced by weighting the channel impulse response estimates based upon a deviation from average.

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.

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 novel and useful whitening matched filter (WMF) for use in a communications receiver. The WMF is constructed by cascading a matched filter and a noise-whitening filter. The response of the matched filter is derived from the time reversed complex conjugate of the channel impulse response. The whitening filter is derived by extracting the minimum phase portion of the mixed phase channel impulse response using homomorphic deconvolution. The whitening filter is implemented using either an FIR or IIR filter adapted to process the data received before and after the training sequence using a minimum phase system in a direction in time opposite to that of the direction of corresponding data sample processing performed by the equalizer.

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.

Method and apparatus for OFDM synchronization and channel estimation. In a temporal embodiment, received embedded system pilot symbols are inverse Fourier transformed at expected index locations and correlated with computed complex conjugates of inverse Fourier transforms of pilot symbols for providing a correlation function for the channel impulse response. In a frequency domain embodiment, embedded system pilot symbols are augmented with pilot-spaced inferred guard band symbols, multiplied by scaled complex conjugates of computed pilot systems, and inverse Fourier transformed into the channel impulse response. Time and frequency are synchronized in feedback loops from information in the channel impulse response. The channel impulse response is filtered, interpolated, and then Fourier transformed for determining channel estimates for equalization.

Multiple Transmit Multiple Receive Orthogonal Frequency Division Multiplexing (‘OFDM’) comprising generating bit streams and corresponding sets of N frequency domain carrier amplitudes ({tilde over (s)}(kN+j), 0≦j≦N−1) modulated as OFDM symbols subsequently to be transmitted from a transmitter, where k is the OFDM symbol number and j indicates the corresponding OFDM carrier number. Affix information is inserted at the transmitter into guard intervals between consecutive time domain OFDM symbols and are used at the receiver to estimate the Channel Impulse Response (H_lm) of the transmission channels, the estimated Channel Impulse Response (Ĥ_lm) being used to demodulate the bit streams in the signals received at the receiver. The affix information is known to the receiver, as well as to the transmitter, and is mathematically equivalent to a vector (c_D) that is common to the time domain OFDM symbols multiplied by at least first weighting factors (α_k) that are different for one time domain OFDM symbol (k) than for another and second weighting factors (w_i(k)) that enable one of the transmit antenna means (i) to be distinguished from another.

A speech processing apparatus which, in the process of performing echo canceling by using a pseudo acoustic echo signal, continuously uses an impulse response used for the previous frame as an impulse response to generate the pseudo acoustic echo signal when a voice is contained in the microphone input signal, and which uses a newly updated impulse response when a voice is not contained in the microphone input signal.

An integrated circuit, e.g. an AC '97 conforming audio codec, includes a digital filter and gain module including multiple channels of gain control and multiple channels of digital filtering. A gain control module includes an overflow check of data samples requiring differing lengths of clamping. Each channel of the digital filter includes a finite impulse response (FIR) filter, and an infinite impulse response (IIR) filter. The digital filtering is implemented largely in hardware independent of the number of channels required and/or independent of the required order of the filtering. Thus, filter channels can be added or additional filtering implemented merely by increasing the clock speed without changing the digital filter design. The FIR filter is capable of being reset each frame to prevent a DC buildup at internal nodes. The IIR filter performs a plurality of 2nd order biquadratic equations in an overall average of as few as four clock cycles per 2nd order biquad. A RAM is used to store the state variables for the 2nd order biquadratic equations. The state variable RAM is reset by controlling the clear input of latches at an input and/or the output of the state variable RAM, and the state variable RAM is addressed by a delta counter which is independent of the particular number of filter channels or filter orders implemented. Test patterns may be inserted between functional modules of an integrated circuit such as the disclosed audio codec by appropriate control of the preset and clear inputs to output latches of the functional modules.

System and method for canceling crosstalk in high-speed communications systems. An embodiment comprises precomputing channel pulse responses for a set of communications channels in a communications backplane, precomputing channel signal responses for the set of communications channels in the backplane with a training sequence, estimating noise coefficients using receiver training and the training sequence, storing noise coefficients for each communications channel in the set of communications channels, and canceling noise in a received transmission of a data sequence using the stored noise coefficients. The regularity of the communications backplane enables the precomputing of various values, which reduces computational requirements. Furthermore, it is often the case that a victim receiver has access to the digital data that is being transmitted by its dominant crosstalking transmitters, thereby simplifying noise cancellation.

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.

In described embodiments, a transceiver includes a baud-rate clock and data recovery (CDR) module with an eye sampler, and an adaptation module for adaptively setting parameters of various circuit elements, such as timing, equalizer and gain elements. Data sampling clock phase of the CDR module is set for sampling at, for example, near the center of a data eye detected by the eye sampler, and the phase of data error sampling latch(es) is skewed by the CDR module with respect to the phase of the data sampling latch. Since the error signal driving the timing adaptation contains the information of the pulse response that the CDR module encounters, the phase of timing error sampling latch(es) of the CDR module is skewed based on maintaining a relative equivalence of input pulse response residual pre-cursor and residual post-cursor with respect to the timing error sampling clock phase.

An integrated circuit, e.g. an Audio Codec (AC) '97 conforming audio codec, includes a digital filter and gain module including multiple channels of gain control and multiple channels of digital filtering. A gain control module includes an overflow check of data samples requiring differing lengths of clamping. Each channel of the digital filter includes a finite impulse response (FIR) filter, and an infinite impulse response (IIR) filter. The digital filtering is implemented largely in hardware independent of the number of channels required and/or independent of the required order of the filtering. Thus, filter channels can be added or additional filtering implemented merely by increasing the clock speed without changing the digital filter design. The FIR filter is capable of being reset each frame to prevent a direct current (DC) buildup at internal nodes. The IIR filter performs a plurality of 2nd order biquadratic equations in an overall average of as few as four clock cycles per 2nd order biquad. A random access memory (RAM) is used to store the state variables for the 2nd order biquadratic equations. The state variable RAM is reset by controlling the clear input of latches at an input and/or the output of the state variable RAM, and the state variable RAM is addressed by a delta counter which is independent of the particular number of filter channels or filter orders implemented. Test patterns may be inserted between functional blocks of an integrated circuit such as the disclosed audio codec by appropriate control of the preset and clear inputs to output latches of the functional blocks.

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.

A radio receiver (100) has an equalizer (500) that operates in the time domain to remove residual interference that is not removed by an IF filter (200) operating in the frequency domain that is caused by an adjacent interfering FM station. The equalizer (500) includes a modified constant modulus algorithm (CMA) to generate a tap update signal from the output of the equalizer (500). The equalizer (500) uses the modified CMA to reduce an amplitude fluctuation of the received signal caused by the adjacent interfering station. The CMA is modified to use an infinite impulse response (IIR) filter (540) to generate the tap update. The IIR filter (540) also speeds up a convergence of the modified CMA to provide better performance.

A wireless communication system includes a transmitter and a receiver. The transmitter includes multiple groups of transmit antennas. Input symbols are generated and then orthogonal space-time block is encoded to produce a data stream for each group of transmit antennas. Each data stream is adaptively linear space encoded to produce an encoded signal for each transmit antenna of each group according to feedback information for the group. The receiver includes a single receive antenna, a module for measuring a phase of a channel impulse response for each transmit antenna. The feedback information is determined independently for each group of transmit antennas from the channel impulse responses. The feedback information for each group of transmit antennas is sent to the transmitter.