Abstract: A variable-precision distributed arithmetic (VPDA) multi-input multi-output (MIMO) equalizer is presented to reduce the size and dynamic power of 112 Gbps dual-polarization quadrature phase-shift-keying (DP-QPSK) coherent optical communication receivers. The VPDA MIMO equalizer compensates for channel dispersion as well as various non-idealities of a time-interleaved successive approximation register (SAR) based analog-to-digital converter (ADC) simultaneously by using a least mean square (LMS) algorithm. As a result, area-hungry analog domain calibration circuits are not required. In addition, the VPDA MIMO equalizer achieves 45% dynamic power reduction over conventional finite impulse response (FIR) equalizers by utilizing the minimum required resolution for the equalization of each dispersed symbol.

Abstract: An image rejection (IR) circuit is configured to receive a complex signal from a radio frequency (RF) mixer, where the complex signal includes an in-phase signal portion and a quadrature signal portion. This IR circuit may include: an in-phase path to remove first mismatch information from the in-phase signal portion and associated with at least one in-phase multi-tap filter; a quadrature path to remove second mismatch information from the quadrature signal portion and associated with at least one quadrature multi-tap filter; and a correlation unit to independently update each of the multiple taps of the in-phase multi-tap filter and the quadrature multi-tap filter according to a priority scheme.

Abstract: A method for calibrating mismatches of an in-phase signal path and a quadrature signal path of a transmitter, including: additionally configuration at least one mixer calibration coefficient at a transmitting part of the transmitter; obtaining at least one mixer testing signal from the transmitting part via loopback for spectrum analysis to derive at least one mixer spectrum analysis result; adjusting the mixer calibration coefficient of the transmitting part according to the mixer spectrum analysis result; and additionally utilizing an in-phase signal path finite impulse response filter and a quadrature signal path finite impulse response filter to calibrate mismatches between a low pass filter of the in-phase signal path of the transmitting part of the transmitter and a low pass filter of the quadrature signal path of the transmitting part of the transmitter. A similar mismatch calibration operation may be applied to a receiver.

Abstract: A receiver applies a calibration method to compensate for skew between input channels. The receiver skew is estimated by observing the coefficients of an adaptive equalizer which adjusts the coefficients based on time-varying properties of the multi-channel input signal. The receiver skew is compensated by programming the phase of the sampling clocks for the different channels. Furthermore, during real-time operation of the receiver, channel diagnostics is performed to automatically estimate differential group delay and/or other channel characteristics based on the equalizer coefficients using a frequency averaging or polarization averaging approach. Framer information can furthermore be utilized to estimate differential group delay that is an integer multiple of the symbol rate. Additionally, a DSP reset may be performed when substantial signal degradation is detected based on the channel diagnostics information.

Abstract: A method of frequency-domain filtering is provided that includes a plurality of filters, the plurality of filters including at least one constrained filter(s) W=I, I and at least one unconstrained filter(s) W=1,K? The method includes cascading the W k=i,K unconstrained filter(s). A single constraint window C is applied to the cascaded W=i,K unconstrained filter(s). The W=1,I constrained filter(s) are cascaded with the constrained cascaded Wk=1,K unconstrained filter(s) to form a resulting filter Wll=C(W 1{circle around (x)} . . . {circle around (x)} W){circle around (x)} W . . . W. The frequency domain representation of the single constraint window C may be based, at least in part, on a time domain representation of a single constraint window C that has been circularly shifted such that the frequency domain representation of the constraint window matches a property of the frequency domain representation of the cascaded W=1,K unconstrained filters.

Abstract: An integrated circuit includes a component calculator configured to compute at least one component value of a highly programmable analog-to-digital converter (ADC) from at least one application parameter, and a mapping module configured to map the component value to a corresponding register setting of the ADC based on at least one process parameter, wherein the integrated circuit produces digital control signals capable of programming the ADC. In a specific embodiment, the component calculator uses an algebraic function of a normalized representation of the application parameter to approximately evaluate at least one normalized ADC coefficient. The component value is further calculated by denormalizing the normalized ADC coefficient. In another specific embodiment, the component calculator uses an algebraic function of the application parameter to calculate the component value.

Abstract: The present invention concerns a method and associated apparatus for reducing the time required to scan an incoming satellite transmission power spectrum for available signals and to determine the characteristics of those signals. The frequency range of interest is scanned in narrow slices to determine approximate input power within each slice. Center frequencies and symbol rates of individual transponders are then estimated based upon these input power approximations.

Abstract: First and second inputs are received. The first input indicates a frequency offset of a frequency band allocated for signal transmission. The said allocated band is a subband of a total band available for transmission. The second input indicates a bandwidth of the allocated band. One or more filters of a transmitter of a communications system are controlled to operate cumulatively in a lowpass filtering mode, wherein the highest frequency in a pass band in the lowpass filtering mode is less than the highest frequency of the total band available for transmission. A signal is filtered using the filter(s).

Abstract: A digital down converter is disclosed. The digital down converter includes an input for receiving a sampled signal having a frequency band of interest, sampled at a first sampling rate, a commutator structure for distributing a set of real sampled signals for polyphase filtering, a complex band pass polyphase filter associated with the distributed signals, for generating in phase and quadrature filtered components, a baseband notch filter, and a frequency translator for generating in phase and quadrature components of the frequency band of interest at baseband.

Abstract: For clipping a signal with respect to a clipping threshold (Amax), a signal portion above the clipping threshold and a peak (Pk) in the signal portion are detected, a clipping function depending on end times (ts, te) of the signal portion, the amplitude of the detected peak and the clipping threshold is derived, and the signal portion is clipped by multiplying it by the clipping function. The first portion can be enlarged up to the lowest boundary points (t1+1, t2?1) on the first ascending portion and the last descending portion of the detected peak. Spectral leakage in the sidebands out to the frequency band of the clipped signal is eliminated by a filtering device. The filtered dipped signal to the input of the amplifier has peaks as close to the saturation threshold of a power amplifier as possible.

Abstract: Methods and systems for using pitch predictors in speech/audio coders are provided. Techniques for optimal pre- and post-filtering are presented, and a general result that post-filtering is more effective than pre-filtering is derived. A practical paired-zero filter design for the low-rate regime is proposed, and this design is extended to handle frequency-dependent periodicity levels. Further, the methods described provide a general performance measure for a post-filter that only uses information available at the decoder, thereby allowing for the optimization or selection of a post-filter without increasing the rate.

Abstract: Systems and methods are disclosed for employing an equalization technique that improves equalizer input sensitivity and which reduces power consumption. In particular, an equalization architecture is described that includes a continuous-time linear equalizer and a decision feedback equalizer, each with offset cancellation that enables the equalizer to be used at high data rates. In addition, the equalization structure has a power-saving mode for bypassing the decision feedback equalizer. These offset cancellation and power-saving features are enabled and controlled using programmable logic on a programmable device.

Abstract: Techniques for fast filtering for a transceiver are presented. A multidimensional filter processor component (MDFPC) can perform configurations and adaptations of multiple digital filters of a transceiver. The MDFPC can treat multiple, separate filters of a transceiver as a single larger multidimensional filter, and jointly update the multiple filters in a single adaptation operation instead of performing multiple adaptation operations on multiple filters. To facilitate multidimensional filter adaptation, the MDFPC can manage respective cross-correlations associated with the inputs of the filters. The MDFPC can facilitate multidimensional filter adaptation by performing multidimensional filter adaptation in the frequency domain, wherein the adaptation can be performed in parallel for multiple frequency sub-channels.

Abstract: Method and apparatuses are disclosed to substantially compensate for various unwanted interferences and/or distortions within a communications receiver. Each of these apparatuses and methods estimate the various unwanted interferences and/or distortions within the communications receiver. Each of these apparatuses and methods remove the estimates of the various unwanted interferences and/or distortions within the communications receiver from one or more communications signals within the communications receiver to substantially compensate for the various unwanted interferences and/or distortions.

Abstract: Techniques are provided that utilize information on the characteristics of the transmitted signal to improve the accuracy of the I/Q imbalance estimation and correction. These techniques can be used to achieve improved image rejection performance over existing solutions, through the use of signal conditioning, windowing and the use of statistical information on the received signal. A number of techniques aimed at improving the accuracy of the I/Q imbalance estimation/correction are presented, including different modes of I/Q imbalance estimation and I/Q imbalance correction in the presence of a complex IF filter in the transceiver. The processing is performed mostly in the DBB in order to be area/power efficient.

Abstract: A radio frequency (RF) front end having group delay mismatch reduction is provided. One embodiment provides a first feed forward path and a second feed forward path. The second feed forward path is electrically in parallel with the first feed forward. The second feed forward path has a first signal path and a second signal path. The first and second signal paths are arranged to be electrically in parallel. The first signal path has a digital filter. The second signal path has a tunable analog filter. The tunable analog filter operates to reduce a delay associated with the second feed forward path as compared with a delay associated with the first feed forward path.

Abstract: For digital data transmitted using a vector signaling encoding, a rank-order equalizer cancels various channel noise such as inter-symbol interference. Further, rank-order units may be cascaded to achieve improved equalization over successive sample vector signals in a rank-order equalizer. Multiple rank-order equalizers further operate in parallel in a feed forward mode or in series in a feedback mode to provide a continuous vector signaling stream equalization.

Abstract: A channel equalization scheme is provided. A linear equalizer using a continuous-time linear equalization and a decision feedback equalizer using a discrete-time decision feedback equalization are integrated together from a hybrid receiver equalizer. The continuous-time linear equalization scheme and the discrete-time decision feedback equalization scheme are blended using a joint adaptation algorithm to form an equalization scheme for inter-symbol interference cancellation in the hybrid receiver equalizer. The hybrid receiver equalizer controls crosstalk while maintaining signal bandwidth and linearity of a signal by the high-order high frequency roll-off of the linear equalizer used. Using this configuration, the hybrid receiver equalizer eliminates the need for adaptive bandwidth controller used in conventional low-pass receiver equalization schemes. The hybrid receiver equalizer can be used in receivers for dual-speed simultaneous transmission on the same physical link.

Abstract: An apparatus relates generally to crest factor reduction. In this apparatus, a finite impulse response filter provides a first cancellation pulse and a second cancellation pulse. A first adder is coupled to receive an input signal and the first cancellation pulse to provide a first difference signal. A peak engine is coupled to receive the first difference signal to provide a cancellation pulse value responsive to the first difference signal. The finite impulse response filter is coupled to receive the cancellation pulse value to provide each of the first cancellation pulse and the second cancellation pulse. A delay is coupled to receive the input signal to provide a delayed input signal. A second adder is coupled to receive the delayed input signal and the second cancellation pulse to provide a second difference signal. The second difference signal is a crest factor reduced version of the delayed input signal.

Abstract: Embodiments of the invention comprise a continuous-time equalizer for reducing ISI in data received from a communication channel, and methods and circuitry for tuning or calibrating that equalizer. Selected coefficients for a transfer function of the equalizer circuit are fixed, while other coefficients are tuned by an adaptive algorithm. The adaptive algorithm minimizes errors associated with the tunable coefficients based on one or more training signals sent by the transmitter and received by the equalizer circuit at the receiver. The training signals allow for a variety of error terms to be calculated, from which the tunable coefficients are updated so as to iteratively minimize the error terms and simultaneously tune the equalizer to more accurately compensate for the degrading effects of the channel.

Abstract: A multichannel radio receiver may include a radio frequency (RF) subsystem and a digital subsystem. The RF subsystem may be configured to provide analog information associated with a radio band to an analog to digital converter (ADC). The ADC samples the analog input and sends digital output to the digital subsystem. The digital subsystem may be configured with one or more channelizers and one or more decoders. A channelizer within the digital subsystem may filter and re-sample the digital output to result in a channel plan having a desired bandwidth and a desired sample rate. The sample rate may be selected for compatibility with a decoder. The decoder may have design specifications based in part on a modulation scheme to be decoded. The design specifications may indicate the desired sample rate to be provided by the channelizer.

Abstract: Described embodiments recover timing and data information from a signal received via a communication channel. An analog-to-digital converter (ADC) operating at a baud rate of the communication channel generates an actual ADC value corresponding to each bit sample of the received signal. A fast symbol estimation module estimates, based on the actual ADC value, a bit value corresponding to each bit sample. The fast symbol estimation module operates at a digital clock rate. The estimated bit values are provided to a timing recovery module. An ADC reconstruction module, based on a first number of pre-cursor estimated bit values, an estimated cursor bit value, and a second number of post-cursor estimated bit values, generates a reconstructed ADC value corresponding to each bit sample. Based on the reconstructed ADC values, the estimated bit values, and the actual ADC values, a corrected bit value is generated for each bit sample.

Abstract: Systems and methods are provided for whitening noise of a received signal vector in a multiple-input multiple-output (MIMO) transmission or storage system. The whitening filter may be designed to whiten an interference component of the received signal vector, where the interference component is derived from modeling transmitter and receiver imperfections as a first coupling between MIMO transmitter outputs and a second coupling between MIMO receiver. The whitening filter may be computed based on the covariance matrix of the interference component.

Abstract: Techniques are provided which may be implemented using various methods and/or apparatuses in a device comprising a receiver to scan a spectral band of a received signal comprising a desired signal contribution to determine whether signal data associated with at least a sub-band of the spectral band further comprises at least one undesired signal contribution. In response to determining that the signal data comprises at least one undesired signal contribution, the mobile station may initiate at least one notch filter to affect the undesired signal contribution in subsequent signal data associated with the received signal.

Abstract: A device for obtaining a RCC signal and related methods are described herein improves the reliability of the RCC signal reception and demodulation. In one aspect, a device configured to obtain a RCC signal includes: a receiving circuit to receive an analog AM RCC signal and to process said analog AM RCC signal to generate a digital AM RCC signal; and a demodulation circuit in connection with said receiving circuit, to demodulate said digital AM RCC signal to generate the RCC signal. In another aspect, a method for obtaining a RCC signal includes: processing a received analog AM RCC signal to generate a digital AM RCC signal; and demodulating said digital AM RCC signal to generate said RCC signal. Since the digital processing method is more reliable than the analog processing method, the reliability of RCC signal reception and demodulation are improved.

Abstract: Disclosed are a broadcast signal transmitter, a broadcast signal receiver, and a method for transceiving a broadcast signal in the broadcast signal transmitter/receiver. A method for transmitting a broadcast signal comprises the following steps: signaling in-band signaling information to at least one of a service component physical layer pipe (PLP) including at least one service component of a broadcast service, a first information PLP including first service information applied to one broadcast service and a second information PLP including second service information applied commonly to a variety of broadcast services; performing the FEC encoding on data included in each PLP; performing time-interleaving on the FEC encoded data; generating a transmission frame including the time-interleaved data; and modulating the transmission frame and transmitting a broadcast signal including the modulated transmission frame.

Abstract: Systems and methods for demodulating a plurality of contiguous channels contained within a bandlimited portion of a radio-frequency (RF) input signal are provided. In an embodiment, the bandlimited portion of the RF input signal is down-converted to baseband. After down-conversion, the bandlimited portion overlaps at baseband with a mirror image of the bandlimited portion. The plurality of contiguous channels within the down-converted signal similarly overlap at baseband and subsequently occupy a bandwidth substantially equal to half that required before down-converting. Image rejection is performed in the digital domain to recover each of the plurality of overlapping channels.

Abstract: A method for canceling adjacent channel interference includes: receiving signal of a first channel, where an interference signal of a second channel remains in the signal of the first channel, and the second channel and the first channel are adjacent channels; receiving a cancellation signal provided by the second channel for canceling the interference signal; filtering the received cancellation signal; adjusting a frequency of the filtered cancellation signal; multiplying the cancellation signal with the adjusted frequency by a gain factor to obtain a signal to be cancelled; and subtracting the signal to be cancelled from the signal of the first channel, and canceling the interference signal to obtain a signal to be decoded. According to the technical solutions provided in the embodiments of the present invention, interference signals of adjacent channels are cancelled, and the signal to noise ratio is improved.

Abstract: A method of equalizing signals from a plurality of balanced transmission line cables having different lengths includes providing a first cable having a first length and a second cable having a second length, the first cable coupled to a variable resistor. A first signal is transmitted along the first cable to the variable resistor such that the first signal is attenuated to assume a first frequency domain characteristic. A second signal is transmitted along the second cable such that the second signal is attenuated to assume a second frequency domain characteristic. A voltage of the first signal is divided in the variable resistor such that the first signal assumes substantially the second frequency domain characteristic. The first signal having the second frequency domain characteristic is outputted.

Abstract: The present invention relates to method for detection of symbols, comprising: receiving at least one communication signal y transmitted over a radio channel H, and receiving a priori LLRs of said channel inputs x; computing soft symbol estimates ? and variance D of said channel inputs x based on LLRs of said channel inputs x; filtering and interference cancelling said received signal y by using a first filter and filtered soft symbol estimates ? so as to obtain a filtered and interference cancelled received signal; estimating MSE matrix B of said filtered and interference cancelled received signal; computing an optimal matrix Gopt for a BCJR algorithm based on said MSE estimate B; further filtering said filtered and interference cancelled received signal by using a second filter so as to obtain an input signal r to the BCJR algorithm; and feeding said input signal r into the BCJR algorithm.

Abstract: Indication of an amount of processing performed in detection and removal of ingress noise may be provided. A frequency domain representation of a narrowband region of a digital input signal may be received. The received frequency domain representation of the narrowband region may be compared with a predetermined threshold. Results from the comparison of the received frequency domain representation of the narrowband region with the predetermined threshold may be aggregated. Based on the aggregated results, an indication of an amount of processing performed by an ingress exciser in removing the ingress noise may be provided.

Abstract: A method and apparatus for tracking and mitigating phase noise in a communication receiver are disclosed. The phase noise tracking and mitigation apparatus operates in a recursive manner and includes a quantizer for determining data symbols from noise-compensated input values, a phase noise estimator for determining raw phase noise values from the data symbols and a first sequence of uncompensated input values, an error concealment module for removing unreliable samples from the raw phase noise values, a filter operable to filter the raw phase noise values to produce filtered phase noise values, and a phase-noise compensator for determining noise-compensated output values dependent upon the filtered phase noise values and the first sequence of uncompensated input values. Filter coefficients, and initial noise-compensated input values are determined from one or more second sequences of uncompensated input values corresponding to known symbols corrupted by phase noise.

Abstract: A received plurality of signals may be filtered to select an in-band signal and/or an out-of-band. A signal strength of the selected signal(s) may be measured. A resolution of an analog-to-digital converter may be controlled based on the measured signal strength(s). The selected in-band signal may be converted to a digital representation via the analog-to-digital converter. The resolution may be decreased when the strength of the in-band signal is higher, and increased when the strength of the in-band signal is lower. The resolution may be increased when the strength of the out-of-band signal is higher, and decreased when the strength of the out-of-band signal is lower. A signal-to-noise ratio and/or dynamic range of the selected signal(s) may be determined based on the measured signal strength(s), and may be utilized to adjust the resolution of the analog-to-digital converter.

Abstract: A transmitter used in a communication system includes a raised cosine filter for transmit pulse shaping. A receiver in the communication system, designed to receive and demodulate transmissions from the transmitter, includes a root-raised cosine filter for receive pulse shaping. The use of a raised cosine filter in the transmitter enables reduction of peak-to-average ratio (PAR) of the output of a power amplifier used in the transmitter, enabling the power amplifier to be implemented to have relatively higher power efficiency than otherwise. In an embodiment, the transmitter and receiver employ ?/2-shift binary phase-shift keying (?/2 BPSK), and the raised cosine filter in the transmitter is implemented to have a roll-off factor of 0.5 and a total length of four symbol periods. In an embodiment, the root-raised cosine filter is implemented to have a roll-of factor of 0.2 and a length of four symbol periods.

Abstract: Techniques for performing channel estimation in an orthogonal frequency domain multiplexing (OFDM) communications system include receiving a plurality of reference signals on a plurality of subcarriers, performing channel estimation to obtain a plurality of raw channel estimates based on the received plurality of reference signals, determining a number of raw channel estimates to be used for a channel estimate refinement and calculating a refined channel estimate using the number of raw channel estimates from the plurality of raw channel estimates.

Abstract: An accurate channel frequency response is obtained by processing an extracted number of pilot tones provided at different locations within a received OFDM symbol. This includes filtering the extracted pilot tones with a first window function, converting the thus filtered pilot tones to a first channel impulse response signal that may include a main tap and a plurality of adjacent taps, removing taps whose absolute values or energy levels are below a predetermined level, processing the remaining taps having sufficient absolute values or energy levels into a second channel impulse response signal that is significantly free of noises, converting the second channel impulse response signal to a frequency-domain signal, and filtering the frequency-domain signal with a second window function having an inverse characteristic of that of the first window function to obtain an accurate channel frequency response.

Abstract: Briefly, in accordance with one or more embodiments, a platform may comprise a receiver to receive a signal that includes an error in the received signal due to a noise signal generated in the platform, and a processor configured to calculate a noise vector from a source of the noise signal and to send the noise vector to the receiver, The receiver may include a digital signal processor configured to estimate an error vector based at least in part on the noise vector and to subtract the estimated error vector from the received signal to cancel the noise signal from the received signal. The noise cancelled from the received signal may include platform noise generated by a bus, a memory circuit, a clock, a power supply, a circuit ground or integrated circuit substrate, or input/output circuit of the platform.

Abstract: An outdoor radio communication system comprises a first radio unit, a second radio unit, and a single cable coupling the first radio unit to the second radio unit. Each radio unit includes a downconverter, a radio processor that is communicatively coupled to the downconverter, and a XPIC module. The cable further includes a first twisted-pair of wires for communicatively coupling the first downconverter to the second XPIC module and a second twisted-pair of wires for communicatively coupling the second downconverter to the first XPIC module. The first XPIC module generates a first reference signal using a signal from the second downconverter to cancel cross-polarization interference in an output signal of the first radio processor. Similarly, the second XPIC module generates a second reference signal using a signal from the first downconverter to cancel cross-polarization interference in an output signal of the second radio processor.

Abstract: Methods and systems to configure a receiver based on a channel condition. A system may be implemented to estimate a channel based on PN sequences in a received signal, concurrently equalize a frequency domain representation of the signal with multiple blind adaptive equalizers, and evaluate results of the equalizations to select coefficients of one of the equalizers for further processing. A first equalizer may implement a constant-step-size (CSS) algorithm for a static channel. A second equalizer may implement a variable-step-size (VSS) algorithm for a dynamic channel. Static and dynamic channels may be distinguished based on convergence/divergence of the equalizers, which may be determined from a mean square error estimated from PN sequences in results of the blind equalizations.

Abstract: Systems and methods for power efficient iterative equalization on a channel are provided. An iterative decoder decodes received data from a channel detector using a decoding process. The decoder computes a decision metric based on the decoded data and adjusts the number of iterations of the decoding process based on the decision metric. The adjustment occurs prior to a reliability criterion for the decoded data being satisfied. The decoder may pass control back to the channel detector if the adjusted number of iterations has occurred or if the reliability criterion is satisfied. Adjusting the number of iterations of the decoding process may include increasing the number of iterations from a predetermined number of iterations. The decision metric may be based on syndrome weight or hard decisions. The decision metric may be chosen to reduce average power consumption of the detector, the decoder, or circuitry including the detector and the decoder.

Abstract: Embodiments of the present invention provide a receiver and a receiving method. The receiver comprises: a branch forming unit a plurality of signal branches; each of the signal branches comprising a joint processing unit; and the joint processing unit being used to eliminate an inter-carrier interference of the subcarrier signal in a present signal branch and the subcarrier signals in other signal branches adjacent to the present signal branch. With the embodiments of the present invention, the inter-carrier interference may be eliminated and the performance of the system may be improved; and the sampling rate of the analog digital converter may also be lowered.

Abstract: A receiver architecture for processing spread spectrum signals. The receiver has an RF front end to receive and down convert a broadcast signal to an intermediate frequency carrier. The IF signal is digitized and provided to a processor (which may be a software-driven DSP, an ASIC or other embodiment) for processing. A given IF carrier is removed and the signal is low pass filtered. The signal is provided to a number of channels, each, for example, correspond to a unique transmitter. On each channel the sample rate is reduced to a predetermined fixed rate with timing mismatch compensated. The Doppler frequency shift, as estimated for the channel, is removed succeedingly. A locally generated copy of the spreading code used by the transmitter is applied to the carrier and Doppler removed signal at the predetermined fixed sample rate. The de-spread signal is used to provide estimates of the Doppler shift and for subsequent sample selection.

Abstract: A receiver circuit includes a unit configured to determine filter coefficients based on a sampling time error of a received signal and a phase error of the received signal, as well as a filter configured to filter a signal, which is based on the received signal, based on the filter coefficients.

Abstract: An apparatus and method for reducing FM audio artifacts in a receiver are provided. A direct conversion radio frequency (RF) receiver converts an analog FM signal into a phase shifted digital low IF signal. A digital controller coupled to the analog FM receiver provides adaptive frequency translation for different channel spacing and provides adaptive low IF configuration through the different channel spacing, thereby suppressing audio artifacts.

Abstract: Disclosed is a dual band receiver which includes an analog-to-digital converter configured to convert a dual band analog RF signal into a dual baseband digital signal; and a first signal extractor configured to generate a first path signal and a second path signal from the dual baseband signal and to extract a first baseband signal using a relative sample delay difference between the first and second path signals, wherein the dual baseband signal includes the first baseband signal and a second baseband signal, the first path signal is a signal obtained by sample delay of the dual baseband signal and then down sampling of a resultant signal, and the second path signal is a signal obtained by down sampling of the dual baseband signal without sample delay.

Abstract: A wireless communication device includes a reception processing circuit including a weight coefficient computation circuit that includes a computation circuit to compute a weight coefficient which is used for removing a distortion of a reception signal caused by a multi-path, and which of each of fingers corresponds to each of a specified number of paths among a plurality of paths caused by the multi-path between the device and the opposing device, by iteratively performing a computation including a complex multiplication between a weight coefficient while being iteratively computed and a component of a correlation matrix, and a control circuit to cause the computation circuit to compute complex multiplications between a first (second) component of a pair of components having a complex conjugate relationship and a first (second) weight coefficient while being iteratively computed when the pair of components is present among components used for computing the weight coefficient.

Abstract: A receiver communicating according to a wireless communication protocol standard for filtering a signal received at a single antenna. The receiver includes filter modules, receiver modules, and a summer. The filter modules receive from antennas multipath components of the signal as transmitted to the receiver. The signal includes bits of data. Each of the filter modules: receives corresponding ones of the multipath components of the signal as received at a respective one of the antennas; and according to the wireless communication protocol standard, filters the signal as received at the respective one of the antennas to generate a filtered signal. The receiver modules respectively receive the filtered signals. Each of the receiver modules combines the multipath components in the respective filtered signal to generate an output signal. Each of the output signals includes a respective version of the bits of data. The summer sums the output signals.

Abstract: A system for correcting gain imbalance and phase imbalance between first (IOUT) and second (QOUT) signals which are 90° out of phase, including circuitry for estimating the phase mismatch (?) and gain mismatch (?) between the first signal and the second signal signals in a plurality of frequency bands. An inverse fast Fourier transform is performed on each of a number of arrays of the phase mismatch estimates and the gain mismatch estimates to generate correction filter coefficients (h[N]) for a N tap correction filter. The N tap correction filter filters an uncorrected value of the second signal to generate a corrected value of the second signal.

Abstract: Various embodiments of the present invention provide systems and methods for processing data. As one example, a circuit is disclosed that includes a pre-detector that detects an estimated pattern in a digital input signal, and a summation element that subtracts the estimated pattern from the digital input signal to yield a noise estimate. The circuit further includes a data dependent noise prediction filter that is adaptively tuned to detect a noise pattern, and that filters the noise estimate to provide a filtered noise estimate.

Abstract: A radio receiver comprising: an antenna for receiving a radio frequency signal amplitude modulated with an audio frequency signal; a digitizer for periodically sampling the radio frequency signal and generating a digital reception signal representative of the amplitude of the radio frequency signal; and a demodulator for demodulating the digital reception signal to generate a representation of the audio frequency signal.