Abstract: A digital front-end circuit is disclosed. In one aspect, the circuit includes a filtering block for filtering received data. The filtering block has a first filter branch for filtering the received data in a first frequency band and a second filter branch for filtering the received data in a selected second frequency band. The second filter branch is in parallel with the first filter branch, is programmable and includes a block for resampling the received data. The front-end circuit also includes a circuit for performing synchronization and spectrum sensing on the received data, which is in connection with the output of the filtering block. The front-end circuit also includes a controller block for controlling the filtering block and the synchronization circuit.

Abstract: Certain embodiments of the invention may include systems and methods for implementing a multirate digital decimating filter for filtering received symbol data. The method may include sampling the received symbol data at a selected sample rate, pre-averaging the sampled received data to provide two samples per symbol; convolving the pre-averaged samples with decimated finite impulse response (FIR) aperture impulse response coefficients to produce detected output samples, convolving the pre-averaged samples with shifted decimated FIR aperture impulse response coefficients to produce zero-crossing transition samples, and adjusting the sample rate based at least in part on averaging the zero-crossing transition samples.

Abstract: Computationally efficient methods and related systems, for use in a test and measurement instrument, such as an oscilloscope, optimize the performance of DFEs used in a high-speed serial data link by identifying optimal DFE tap values for peak-to-peak based criteria. The optimized DFEs comply with the behavior of a model DFE set forth in the PCIE 3.0 specification.

Abstract: The invention relates to an apparatus and a method for detecting and correcting input signals regarding click noise, using a first detector unit and a second correction unit, the detector unit differentiates the input signal and extracts the noise signal and compares the sum of two noise samples with a threshold value and in case this sum exceeds the threshold value a click is observed, while the correction unit corrects the input signal in the phase domain and differentiates it prior to outputting the corrected signal.

Abstract: This equalization device and method, while preventing an expansion of circuit size, enable high-speed detection of a CIR in order to effectively achieve frequency domain equalization even when the phase of the received signal is significantly different from the phase of the known signal, and when the transmission channel has large temporal variations. A reference signal extraction unit (112) extracts both a real part component and an imaginary part component from the portion of the received signal including the known signal. A CIR detection filter unit (120) filters the known signal to generate a first processed signal and a second processed signal, detects real part filter coefficients by updating a first filter coefficient used in filtering the known signal in such a way that the first processed signal converges to the real part component, and detects imaginary part filter coefficients by filtering the known signal in such a way that the second processed signal converges to the imaginary part component.

Abstract: The present invention relates to a method for generating impairment covariances for equalization in a receiver of a wireless communication system, as well as an covariance estimator, a receiver and a wireless communication system associated therewith, where the receiver is equipped with multiple antennas, and an impairment covariance matrix is used to calculate equalization weighting vector for signals transmitted by a transmitter, the method comprising the steps of: calculating a raw impairment covariance estimate between a first antenna and a second antenna of the multiple antennas on each of subcarriers allocated to the transmitter in frequency domain, transforming the raw impairment covariance estimates into time domain, masking the transformed impairment covariance estimates by a triangle window with a width defined by a maximum delay spread, determining a threshold based on the transformed impairment covariance estimates for the subcarriers and thresholding the masked impairment covariance estimates wit

Abstract: One embodiment of the present invention relates to a combined mixer filter circuit. The circuit includes a sampler, a plurality of filter branches, and a coefficient generator. The sampler is configured to provide a sampled signal by sampling a received signal at a specified rate. The plurality of filter branches has selectable filter coefficients. The plurality of filter branches are configured to receive the sampled signal and generate a mixed and filtered output signal without a separate mixer component. The coefficient generator is coupled to the plurality of filter branches. The coefficient generator is configured to assign filter coefficient values to the selectable filter coefficients to yield a selected mixing function for the mixed filtered output signal.

Abstract: An apparatus to process an input signal received via a power line in a power line communication system includes a jammer remover and a jammer detector. The jammer remover removes interference resulting from transmission of electricity and the input signal over the power line from the input signal. The jammer remover generates a jammer-canceled signal using an adaptive filtering procedure. The jammer detector is coupled to the jammer remover to detect the interference.

Abstract: A method and system of adaptation of a linear equalizer using a virtual decision feedback equalizer (VDFE) are disclosed. In one embodiment, a method of adjusting a setting of a linear equalizer includes determining a change to a decision feedback equalizer (DFE) tap weight value of a predefined metric according to a data signal and an error signal (e.g., the change may be generated according to an average of a specified plurality of data signals and the error signal); using the change in the DFE tap weight value to algorithmically generate a modification in a linear equalizer setting; and adjusting the linear equalizer setting. The linear equalizer is located in a feed-forward path and/or a feedback path of data transmission. The linear equalizer may be located in a transmitter and/or a receiver. The linear equalizer may be a continuous time linear equalizer and/or a Finite Impulse Response (FIR) linear equalizer.

Abstract: A processor, comprising a first data input configured to receive a stream of samples of a first signal having a spectral space, the stream having a data rate of at least 4 GHz; a second data input configured to receive a stream of samples of a second signal; a multitap correlator, configured to receive the first stream of samples and the second stream of samples, and producing at least one correlation output for each respective sequential sample of the first signal received; and a programmable control configured to alter a relationship of the stream of samples of the first signal and the stream of samples of the second signal, to thereby select, under program control, an alterable correlation output.

Abstract: A receiver and associated method estimates a channel impulse response of the communications signal. A communications signal is received as a burst of transmitted symbols, including a known training sequence. The joint estimation of timing offset and initial channel impulse response is determined based on the cross-correlations of the known transmitted symbols and received communications signal. A constant modulus interference removal iteration is applied to improve the initial channel impulse estimation.

Abstract: According to various embodiments, a method is disclosed that includes receiving an orthogonal frequency-division multiplexing (OFDM) modulated signal at a modulator; filtering the received modulated signal using a plurality of sets of filter coefficients with a linear predictor algorithm; and estimating a channel frequency response based on the filtering.

Abstract: A received signal of interest is processed by determining timing of interference spikes in the received signal of interest. Receivers can determine when certain types of interference spikes are expected to occur, e.g., based on when different users are scheduled to transmit data during an overlapping portion of the same transmission time interval. The interference timing information is used by the receiver to soft scale signal values recovered from the received signal of interest that coincide with the interference spikes separately from remaining ones of the signal values. This way, fast changing interference power can be accurately tracked during periods of known interference spikes while also accurately tracking slower changing interference power during other periods.

Abstract: Embodiments herein include a method of channel estimation in a wireless communication node. The method comprises generating, based on samples of a received signal, initial estimates of a plurality of channel coefficients forming a channel response. The method also entails, for each of the channel coefficients, dynamically calculating a coefficient-specific filter span for that channel coefficient. Notably, the coefficient-specific filter span for any given channel coefficient is calculated according to a closed-form function that minimizes aggregated estimation noise for the channel coefficient and estimation bias attributable to errors in tracking the channel coefficient. The method finally includes adapting filtering of the initial estimates independently for each of the channel coefficients to be performed over the coefficient-specific filter span calculated for that channel coefficient.

Abstract: A communication circuit includes a sampling clock generating circuit generating a sampling clock signal having a frequency that is “m” times greater than a bit rate of the communication data and containing “n” pulses in each bit period of the communication data; and a sampling circuit sampling the communication data based on the sampling clock signal to obtain “n” sets of received data in each bit period of the communication data. The sampling clock generating circuit delays the sampling clock signal when a first one or more of the “n” sets of received data are different from a value of the rest of the “n” sets of received data, and advances the sampling clock signal when a value of a last one or more of the “n” sets of received data is different from a value of the rest of the “n” sets of received data.

Abstract: Apparatuses, methods and systems of synchronizing a receiver to a desired signal are disclosed. One method includes obtaining synchronization information of an interfering signal, and adjusting an out-of-band response of a receiver filter based at least in part on the synchronization information of the interfering signal, wherein the in-band response of the receiver filter is determined by frequency components of the desired signal.

Abstract: A mobile device includes a processing device that converts a digital input signal to an analog output signal and a memory device that stores a plurality of gain tables, each including values associated with processing the digital input signal. The mobile device further includes at least one filter and at least one amplifier. The processing device is configured to select one of the plurality of gain tables based on the signal strength of the digital input signal and apply one or more values in the selected gain table to the filter, the amplifier, or both, to process the digital input signal, the analog output signal, or both. A method includes receiving a digital input signal, identifying a signal strength of the digital input signal, selecting one of a plurality of gain tables, and processing the digital input signal based on the selected gain table.

Abstract: A data signal correction circuit includes a channel characteristic calculator unit configured to calculate a channel characteristic estimate value of a received data signal on the basis of a pilot signal, a path detector unit configured to determine a delay quantity of multipath propagation of the received data signal on the basis of the calculated channel characteristic estimate value, and an adaptive filter configured to receive the delay quantity and the channel characteristic estimate value as input items, adjust an input interval of the channel characteristic estimate value along a carrier frequency axis in accordance with the delay quantity, and perform adaptive equalization on the channel characteristic estimate value inputted to the adaptive filter at the adjusted input interval.

Abstract: One aspect of the present invention includes a radio frequency (RF) receiver having a first mixer and a second mixer. The first mixer may be an I-mixer and the second mixer a Q-mixer for downconverting the received RF signal. An impedance circuit is disposed between the first mixer and the second mixer to decouple the channels. In another aspect of present invention, the RF receiver includes a digital filter having at least one complex coefficient. The digital filter exhibits asymmetrical frequency response, and may be used to compensate the asymmetrical frequency response of another filter in the RF receiver.

Abstract: A signal receiver includes an antenna interface for receiving signals from an antenna, analog signal processing circuitry coupled to the antenna interface for processing the received signals to produce filtered signals, sampling circuitry to sample the filtered signals so as to produce digitized received signals, a digital compensator to receive the digitized received signals and compensate for non-uniform group delay introduced by the analog signal processing circuitry to produce compensated digitized received signals, and a digital processor to process the compensated digitized received signals so as to produce a result.

Abstract: An ADC has ADC channels converting an analog input signal into an digital output signal in a time interleave manner; a channel combiner combining channel digital signals respectively output by the ADC channels and generate the digital output signal; an adaptive filter provided at one of the plurality of ADC channels; and a correction circuit detecting a skew error in the digital output signal, generating a coefficient of the adaptive filter according to the skew error for setting it in the filter. According to the skew error, in a first setting, the correction circuit sets the coefficient such that the adaptive filter phase-shifts to one direction a phase of the channel digital signal and, in a second setting, the correction circuit sets the coefficient such that the adaptive filter phase-shifts to an opposite direction and sets a coefficient with which the skew error is suppressed to a desired level.

Abstract: In a method for interference suppression of a signal composed of symbols, each symbol is transmitted twice in temporal succession as first symbol and second symbol, so that a segment of a symbol corresponds to a segment of the first symbol and a segment of the second symbol. An interference-suppressed signal is composed from the first and second symbols.

Abstract: According to one or more embodiments, multiple transmitters may simultaneously transmit orthogonal acquisition sequences with certain frequencies zeroed out, such that receivers receiving a signal (waveform) may separate the signal into the orthogonal sequences based on which frequencies are zeroed out to perform acquisition processes. For example, each transmitter may simultaneously transmit an orthogonal acquisition sequence with certain tones zeroed out depending upon which transmitter transmits the symbol. A particular receiver may then receive a signal, and filter it to produce a plurality of filtered signals that distinguish orthogonal acquisition sequence symbols based on which tones are zeroed out within the symbols.

Abstract: A receiver and a method for equalizing signals, the method includes: receiving input signals; sampling the input signals to provide oversampled samples; processing the oversampled samples to provide symbol spaced samples and to provide fractionally spaced samples that represent the oversampled samples; calculating taps of a fractionally spaced equalizer based on the symbol spaced samples; feeding the taps to the fractionally spaced equalizer; and filtering the fractionally spaced samples by the fractionally spaced equalizer to provide equalized samples.

Abstract: A signal receiver includes an antenna interface for receiving signals from an antenna, analog signal processing circuitry coupled to the antenna interface for processing the received signals to produce filtered signals, sampling circuitry to sample the filtered signals so as to produce digitized received signals, a digital compensator to receive the digitized received signals and compensate for non-uniform group delay and amplitude distortion introduced by the analog signal processing circuitry to produce compensated digitized received signals, and a digital processor to process the compensated digitized received signals so as to produce a result.

Abstract: An apparatus and a method for estimating a channel in a broadband wireless access system. A receiving apparatus includes a first channel estimation filter having a tap coefficient which is channel-adaptively variable, the first channel estimation filter acquiring a channel impulse response by filtering received pilot symbols, a second channel estimation filter having a fixed tap coefficient, the second channel estimation filter acquiring a channel impulse response by filtering received pilot symbols, and a selector which selects the operation of one of the first and the second channel estimation filters according to channel correlativity.

Abstract: Method for identifying within an incident signal at least one frequency communication channel, the incident signal being received on a wireless apparatus, comprising determining a power spectral density (PSD) of that incident signal and identifying the at least one channel from the PSD. The PSD determination comprises storing temporal samples of the incident signal received during a predetermined duration and performing a first Discrete Fourier Transform (DFT) processing on the stored samples.

Abstract: A method for DC offset cancellation includes defining, in a range of possible gain values for operating a direct conversion receiver, multiple sub-ranges of the possible gain values. Multiple DC offset correction values for the respective sub-ranges are stored in a memory. Upon detecting at the receiver that a gain of the receiver has changed from a first sub-range to a second sub-range, DC offset cancellation is initiated based on a DC offset correction value stored for the second sub-range and on a condition relating to past operation in the second sub-range.

Abstract: In an embodiment of the present invention, a feedback technique is used to track a reference signal with a DFE summing node common mode voltage. For example, in an embodiment implemented in CML, the feedback signal shifts both differential signals (e.g., the summing node common voltage and the reference voltage) by the same amount. In such an embodiment, the feedback technique preferably changes the reference common mode but not its differential mode.

Abstract: A receiver (or transceiver) is selectable between a Gaussian mode and a non-Gaussian mode. In the non-Gaussian mode, a transformation block applies a non-linear transformation to signal samples to convert non-Gaussian noise in the signal samples to Gaussian or approximately Gaussian noise. In the Gaussian mode, the transformation block is bypassed. Samples are equalized using an equalizer configured to operate with a Gaussian or approximately Gaussian channel.

Abstract: This invention discloses a method and an apparatus for reducing false alarm rate in preamble detection due to discontinuities between successive symbols in a received signal. The invention is usable in, e.g., a LTE compliant system. In one embodiment, the apparatus includes a filter for filtering the received signal to yield a filtered signal, and a correlator for correlating the filtered signal with a predetermined preamble. The apparatus further comprises a first multiplier for modifying the received signal before it is filtered, or a second multiplier for modifying the filtered signal before it is correlated, or both. The first multiplier multiplies the received signal with a first time-domain window function configured to substantially smooth discontinuities at symbol boundaries in the received signal. The second multiplier multiplies the filtered signal with a second time-domain window function configured to substantially suppress spikes present in the filtered signal around the symbol boundaries.

Abstract: In general, the subject matter described in this disclosure can be embodied in methods, systems, and program products for adapting data rate. The method includes receiving a data transmission at a data rate, identifying a first quantity of packets that were not adequately received, and using the first quantity to increase a packet loss level. A second quantity of improper bits in those packets that were adequately received at the data rate are identified, and the second quantity is used to increase a bit error level. As a result of having determined that the first value does not satisfy a first criterion for reducing the rate of the data transmission and that the second value does satisfy a second criterion for increasing the rate of the data transmission, an instruction is sent for causing the sending device to increase the rate of the data transmission.

Abstract: In order to solve a problem of achieving distortion compensation with high accuracy, a digital filter device includes a first distortion compensation filter unit for conducting distortion compensation of first waveform distortion included in an inputted signal through digital signal processing, a first filter coefficient setting unit for setting a filter coefficient of the first distortion compensation filter unit, a second distortion compensation filter unit for compensating second waveform distortion included in a signal outputted from the first distortion compensation filter unit, and a second filter coefficient setting unit for setting a filter coefficient of the second distortion compensation filter unit based on the filter coefficient set by the first filter coefficient setting unit.

Abstract: Filter coefficients are generated by testing a wireless communication system using single-tone signals. While using the filter coefficients in a filter module, signal imbalance caused by a local oscillator or analog elements in a wireless communication system can be eliminated, so as to prevent the wireless communication system from being affected by the noises.

Abstract: Communication devices and associated methods for reducing I/Q impairments in signals used by the communication devices are described. A transmitter device may perform filtering (or matrix multiplication) on digital I and Q signals to pre-correct them before converting them into analog I and Q signals. The pre-correction may pre-compensate for I/Q impairments which have not been introduced yet, but which will subsequently be introduced during digital-to-analog conversion, I/Q modulation, or other processing that occurs to produce a transmission signal from the original digital I and Q signals. A receiver device may receive a transmission signal, produce digital I and Q signals from the received signal, and perform filtering on the digital I and Q signals to correct I/Q impairments at a plurality of frequency offsets.

Abstract: A receiver may be operable to generate estimates of transmitted symbols using a sequence estimation process that may incorporate a non-linear model. The non-linear model may be adapted by the receiver based on particular communication information that may be indicative of non-linearity experienced by the transmitted symbols. The receiver may generate a reconstructed signal from the estimates of the transmitted symbols. The receiver may adapt the non-linear model based on values of an error signal generated from the reconstructed signal, and the values of the error signal may be generated from a portion of the generated estimates that may correspond to known symbols and/or information symbols. The values of the error signal corresponding to the known symbols may be given more weight in an adaptation algorithm, and the values of the error signal corresponding to the information symbols may be given less weight in the adaptation algorithm.

Abstract: A propagation channel estimation unit estimates a propagation channel estimation value. A symbol replica generation unit generates a symbol replica that is modulated symbol of a demodulated information. A signal extraction unit extracts, in an arbitrary time duration, each subcarrier component of a received signal in which an interference signal was removed, the extraction being made based on the propagation channel estimation value and the symbol replica. A demodulator demodulates a signal of each subcarrier component in the received signal, the demodulation being made based on signal during the time duration extracted by the signal extraction unit.

Abstract: Systems, methods, and devices enable spectrum management by identifying, classifying, and cataloging signals of interest based on radio frequency measurements. In an embodiment, signals and the parameters of the signals may be identified and indications of available frequencies may be presented to a user. In another embodiment, the protocols of signals may also be identified. In a further embodiment, the modulation of signals, data types carried by the signals, and estimated signal origins may be identified.

Abstract: The first and second outputs of a signal generation system are coupled to the first and second inputs of a signal digitizing system via respective electrical conductors. A controller directs the generation system to generate a first calibration signal, and the digitizing system responsively captures a first set of vector samples. The conductors are then reconfigured so they connect the first and second outputs of the generation system respectively to the second and first inputs of the digitization system. The controller then directs the generation system to generate a second calibration signal, and the digitizing system responsively captures a second set of vector samples. The controller or other processing agent computes gain and/or phase impairments using the first and second vector sample sets. Digital filter parameters may be computed based on the computed impairment(s), and used to correct the impairment(s) of the generation system and/or the digitizing system.

Abstract: A digital filter-decimator-tuner is configured to receive a complex signal input xk and output a complex USB signal yuk and a complex LSB signal ylk. It includes a USB processing path coupled to receive xk and output yuk, the USB processing path including a USB FIR filter configured to receive a portion of the xk signal and output a first USB intermediate filtered signal, a decimator configured to decimate the first USB intermediate filtered signal and output a second USB intermediate signal, a USB tuner configured to receive the second USB intermediate signal and a USB tuning signal and output a third USB intermediate signal, and a USB equalization filter configured to receive the third USB intermediate signal, and output yuk; and a parallel LSB processing path coupled to receive xk and output ylk. The USB and LSB processing paths may be implemented by the same hardware in one embodiment.

Abstract: A system for synchronizing IQ demodulators includes IQ demodulators, phase-controlling devices, a reference signal, a phase detector, and a control device. The phase-controlling devices are each associated with one of the IQ demodulators for outputting an output signal to its associated IQ demodulator having a phase controlled by the associated phase-controlling device. The phase detector is in communication with the output signals for determining whether the phase of any of the output signals is out-of-phase with a reference phase of the reference signal. The control-device is in communication with the phase-controlling devices programmed, internally or externally, to send a control signal to the associated phase-controlling device for any of the output signals which are out-of-phase with the reference phase of the reference signal so that the associated phase-controlling device synchronizes the phase of the output signal to being in-phase with the reference phase of the reference signal.

Abstract: In order to reduce interference in an audio signal during a call on a mobile communication device, a plurality of transforms of the audio signal is performed, each transform containing phase information and amplitude information of corresponding samples of the audio signal. The results of the transforms are then averaged in order to generating a compensation signal that can be subtracted from the audio signal.

Abstract: A device implements data reception with edge-based partial response decision feedback equalization. In an example embodiment, the device implements a tap weight adapter circuit that sets the tap weights that are used for adjustment of a received data signal. The tap weight adapter circuit sets the tap weights based on previously determined data values and input from an edge analysis of the received data signal using a set of edge samplers. The edge analysis may include adjusting the sampled data signal by the tap weights determined by the tap weight adapter circuit. A clock generation circuit generates an edge clock signal to control the edge sampling performed by the set of edge samplers. The edge clock signal may be generated as a function of the signals of the edge samplers and prior data values determined by the equalizer.

Abstract: A system and method for receiving a radio frequency signal, comprising a device for digitizing, without prior alteration of frequency, an analog radio frequency representation of each of a plurality of radio frequency signals to produce a respective plurality of digital radio frequency signals having a respective associated radio frequency digital clock, the plurality of digital radio frequency signals having a sufficiently high respective associated clock rate to preserve an information content of an information communication present in the analog radio frequency representation; a switch matrix adapted to concurrently switch the plurality of digital radio frequency signals and associated digital radio frequency clock to ones of a plurality of digital signal processors; and a control adapted to selectively automatically control the concurrent switching of a plurality of digital signals and associated digital clock to the respective plurality of digital signal processors; wherein the digital signal processors

Abstract: In one embodiment, a receiver is provided for use in a multiple-input system that includes a receiving antenna receiving a time-domain signal corresponding to a plurality of signals transmitted from a plurality of transmitting antennas. The receiver includes: (a) a transform unit adapted to transform the time-domain signal into a frequency-domain signal; (b) a channel estimation unit adapted to estimate, based on the frequency-domain signal and a frequency-domain pilot signal, a combined transfer function corresponding to a plurality of transfer functions of respective channels between the plurality of transmitting antennas and the receiving antenna; and (c) a channel separation unit including a plurality of frequency-domain convolution units that separate the combined transfer function into a plurality of estimated channel transfer functions.

Abstract: A method and system for signal reception and processing, and more particularly for reducing the effects of random additive impulse interference is provided.

Abstract: A communication device according to an embodiment of the present invention includes a communication antenna that receives a transmission signal where a spectrum spread signal subjected to a spectrum spread is modulated; an intermediate frequency converting unit that converts the transmission signal received by the communication antenna into an intermediate frequency signal having a predetermined frequency; an analog to digital converting unit that discretizes the intermediate frequency signal and outputs a discretization signal; a noise removing unit that detects a noise other than a normal thermal noise included in the discretization signal and removes the detected noise from the discretization signal; and a demodulating unit that demodulates the spectrum spread signal, based on the discretization signal that is output from the noise removing unit.

Abstract: Provided are a method and an apparatus for encoding and decoding an audio signal. A method for encoding an audio signal includes receiving a transformed audio signal, dividing the transformed audio signal into a plurality of subbands, performing a first sinusoidal pulse coding operation on the subbands, determining a performance region of a second sinusoidal pulse coding operation among the subbands on the basis of coding information of the first sinusoidal pulse coding operation, and performing the second sinusoidal pulse coding operation on the determined performance region, wherein the first sinusoidal pulse coding operation is performed variably according to the coding information. Accordingly, it is possible to further improve the quality of a synthesized signal by considering the sinusoidal pulse coding of a lower layer when encoding or decoding an audio signal in an upper layer by a layered sinusoidal pulse coding scheme.

Abstract: There is provided a peak suppressor including a first peak position detector to detect a first time when amplitude of a transmission signal changes in a convex manner and exceeds a specific level as a first peak position, a second peak position detector to detect a second time that is different by a predetermined time from the first peak position when the amplitude of the transmission signal exceeds the specific level as a second peak position, a suppression signal generator to generate a suppression signal for suppressing the amplitude of the transmission signal to a value that is equal to or lower than the specific level, based on amplitude and phase of the transmission signal corresponding to the first and second peak positions detected by the first and second peak position detectors, and an adder to add the suppression signal generated by the suppression signal generator to the transmission signal.

Abstract: A multi-stage receiver including, in one embodiment, a sequence of processing stages. At least one of the processing stages includes a first processing block, a delay block, and a second processing block. The first processing block is adapted to receive an input signal and generate from the input signal one or more processing parameters. The delay block is adapted to generate a delayed signal. The second processing block is adapted to apply the one or more processing parameters to the delayed signal to generate an output signal. The delay block compensates for one or more processing delays associated with the generation of the one or more processing parameters by the first processing block.