Abstract: Aspects of a method and system for unconstrained frequency domain adaptive filtering include one or more circuits that are operable to select one or more time domain coefficients in a current filter partition. A value may be computed for each of the selected one or more time domain coefficients based on a corresponding plurality of frequency domain coefficients. The corresponding plurality of frequency domain coefficients may be adjusted based on the computed values. A subsequent plurality of frequency domain coefficients in a subsequent filter partition may be adjusted based on the computed values. Input signals may be processed in the current filter partition based on the adjusted corresponding plurality of frequency domain coefficients. A time-adjusted version of the input signals may be processed in a subsequent filter partition based on the adjusted subsequent plurality of frequency domain coefficients.

Abstract: The least mean square method for estimation in sparse adaptive networks is based on the Reweighted Zero Attracting Least Mean Square (RZA-LMS) algorithm, providing estimation for each node in the adaptive network. The extra penalty term of the RZA-LMS algorithm is then integrated into the Incremental LMS (ILMS) algorithm. Alternatively, the extra penalty term of the RZA-LMS algorithm may be integrated into the Diffusion LMS (DLMS) algorithm.

Abstract: An adaptive filter is disclosed, having a plurality of computation groups, a plurality of computation circuits, a summation circuit, a slicer circuit, an updating circuit, and a control circuit. Each computation group corresponds to an equalization parameter and has a plurality of memory cells. When the corresponding equalization parameter of a computation group is greater than a predetermined value, the control circuit configures the computation group and the computation circuit to collaboratively generate an output of the computation group. The summation circuit sums up the outputs of the computation groups to produce a filter output. The slicer circuit generates a slicer output according to the filter output. The updating circuit updates the equalization parameters according to the filter output and the slicer output.

Abstract: An exemplary receiver equalizer includes a first decision feedback equalizer (DFE) sampler coupled to a summer, the first DFE to latch an equalized output of the summer. The first branch includes a second DFE sampler coupled to the first DFE sampler, the second DFE to latch an output of the first DFE sampler. The first branch includes a third DFE sampler coupled to the second DFE sampler, the third DFE to latch an output of the second DFE sampler. The summer coupled to the first, second, and third DFE samplers of the first branch, the summer to integrate the output of said DFE samplers, the received signal, and equalized outputs from one or more other branches, wherein the integrating occurs over a plurality of unit intervals (UIs).

Abstract: Improvements in voice signals transmitted within communication systems are obtained by use of adaptive filters, front and rear microphones, noise cancelling systems and other means and methods. Disclosed embodiments include the use of directional microphones, primary inputs, secondary inputs, adaptive weight generators, canceller outputs to improve signal to noise ratios and other communication attributes.

Abstract: Methods and apparatus for provision of equalization effort-balancing of transmit (TX) Finite Impulse Response (FIR) and receive (RX) Linear Equalizer (LE) or RX Decision Feedback Equalizer (DFE) structures in high-speed serial interconnects are described. In some embodiments, data corresponding to a plurality of transmit equalization values and a plurality of receive equalization values for each lane of a link having a plurality of lanes is detected. At least one of the plurality of the transmit equalization values and at least one of the plurality of the receive equalization values are selected for each lane of the plurality of lanes of the link based on detection of saturation in a Decision Feedback Equalizer (DFE) tap of a corresponding lane of the link. Other embodiments are also claimed and/or disclosed.

Abstract: An audio system with at least one audio channel may include a digital audio processor in which at least one digital filter is implemented for each channel. The digital filter of each channel may include: an analysis filter bank configured to receive a broad-band input audio signal and divide the input audio signal into a plurality of sub-bands, a sub-band filter for each sub-band, and a synthesis filter bank configured to receive the filtered sub-band signals and combine them for providing a broad-band output audio signal. A delay is associated with each sub-band signal, the delay of one of the sub-band signals being applied to the broad-band input audio signal upstream of the analysis filter bank and the residual delays being applied to the remaining sub-band signals downstream of the analysis filter bank.

Abstract: A method and a circuit for controlling an equalizer and an equalizing system are disclosed. The method includes providing a first level from a set of levels as a peaking level of the equalizer; equalizing a transmission signal by using the equalizer with the first level to obtain a first signal; providing a second level from the set of levels as the peaking level of the equalizer; equalizing the transmission signal by using the equalizer with the second level to obtain a second signal; determining a first frequency of the first signal; determining a second frequency of the second signal; comparing the first frequency and second frequency to obtain a comparing result; and determining the peaking level of the equalizer for following equalization of the transmission signal in accordance with the comparing result.

Abstract: Adaptive filtering is used to substantially cancel distortion in radio frequency (RF) signals. Such adaptive filtering can be used in an RF transmitting module to pre-compensate an RF signal with compensation (inverse) distortion to cancel inherent transmission path distortion from the RF signal. Adaptive filtering can also be used in a multi-carrier RF receiving module to cancel from a given carrier signal distortion due to cross talk from adjacent carrier signals. Adaptive filtering in an RF transceiver can be used to cancel from a received RF signal distortion arising from leakage of a transmit signal into the receive path.

Abstract: The adaptive filter for system identification is an adaptive filter that uses an algorithm in the feedback loop that is designed to provide better performance when the unknown system model has sparse input, i.e., when the filter has only a few non-zero coefficients, such as digital TV transmission channels and echo paths. In a first embodiment, the algorithm is the Normalized Least Mean Square (NLMS) algorithm in which the filter coefficients are updated at each iteration according to: w ? ( i + 1 ) = w ? ( i ) + ? ? ( i ) ? e ? ( i ) ? u T ? ( i ) ? u ? ( i ) ? 2 , where the step size ? is varied according to ?(i+1)=??(i)+?|e(i)|.

Abstract: The present inventions are related to systems and methods for data processing, and more particularly to systems and methods for data processing systems with symbol selective scaling interacting with parity forcing.

Abstract: In described embodiments, Linear Phase, Finite Impulse Response, filters incorporate a power complementarity property into a perfect reconstruction filter bank. Non-linear constraints for type A and type B filters are included in the Sequential Quadratic Programming design of the filters. An initial Quadrature Mirror Filter includes perfect reconstruction constraints, which might be optimized through iterative design techniques. Embodiments might be employed in noise reduction applications related to, for example, signal processing of images.

Abstract: Various embodiments of the present invention provide systems and methods for reducing filter sensitivities. As an example, reduced sensitivity filter circuits are discussed that include a digital filter and a filter tap adaptation circuit. The digital filter is operable to filter a received input based at least in part on a plurality of filter taps, and to provide a filtered output. The filter tap adaptation circuit is operable to receive an error value and a weighting control value, and to adaptively calculate at least one of the filter taps using the error value and the weighting control value.

Abstract: Systems and methods are provided for an adjustable filter engine. In particular, an electronic system is provided that can include a focus module, memory, and control circuitry. In some embodiments, the focus module can include an adjustable filter engine and a motor. By using the adjustable filter engine to generate a filter with a large number of filter coefficients, the control circuitry can accommodate a variety of system characteristics. For example, by generating a set of cumulative coefficients and re-arranging the order of the cumulative coefficients, the control circuitry can reduce the bit-width requirements of the adjustable filter engine hardware. For instance, the control circuitry can reduce the number of multipliers required to perform a convolution between an updated filter and one or more input signals. In some embodiments, the updated filter can be generated to reduce oscillations of the motor movement due to a new position request.

Abstract: The system and method for least mean fourth adaptive filtering is a system that uses a general purpose computer or a digital circuit (such as an ASIC, a field-programmable gate array, or a digital signal processor that is programmed to utilize a normalized least mean fourth algorithm. The normalization is performed by dividing a weight vector update term by the fourth power of the norm of the regressor.

Abstract: A method and a system is provided for Coordinate Rotation Digital Computer (CORDIC) based matrix inversion of input digital signal streams from multiple antennas using an bi-directional ring-bus architecture. The bi-directional ring bus includes a first ring bus having signals flow in a clockwise direction, and a second ring bus having signals flow in a counter-clockwise direction. An I/O controller is coupled to the first and the second ring bus, respectively. A plurality of processing elements (PEs), where each of the plurality of PEs is coupled to the first and the second ring bus, respectively, wherein each of the plurality of PEs includes at least one CORDIC core for performing CORDIC iterations on the plurality of input digital stream signals to produce inversed matrix signals.

Abstract: A receiver is configured to receive a sample of an inter-symbol correlated (ISC) signal, the sample corresponding to a time instant when phase and/or amplitude of the ISC signal is a result of correlation among a plurality of symbols of a transmitted symbol sequence. The receiver may linearize the sample of the ISC signal. The receiver may calculate a residual signal value based on the linearized sample of the ISC signal. The receiver may generate an estimate of one or more of said plurality of symbols based on the residual signal value. The linearization may comprise applying an estimate of an inverse of a non-linear model. The non-linear model may be a model of nonlinearity experienced by the ISC signal in a transmitter from which the ISC signal originated, in a channel through which the ISC signal passed en route to the receiver, and/or in a front-end of the receiver.

Abstract: A reconfigurable digital signal filter processing unit for use in a communication device is provided. The reconfigurable filters processor can implement different filter topologies to adapt to a range or wireless technology characteristics. The reconfigurable filter processor comprises a plurality of filter blocks whose inputs can be selected based on the desired configuration of the filter. Each filter block applies a transfer function to a received signal to achieve a desired filtering function.

Abstract: In order to reduce or eliminate aliasing in a process control network, filtering of a measurement signal may be set based on the module execution rate in a process control system. A Nyquist frequency for the module may be determined based on the module execution rate where the Nyquist frequency may be twice the execution rate. Filtering after an analog to digital convertor may be set based on the module execution rate. In the analog to digital convertor, digital filtering after the converter may be set based on the module execution rate and the frequency content of the analog signal may be attenuated by a filter at and above the Nyquist frequency for the module execution rate.

Abstract: Various embodiments of the present invention provide systems and methods for data processing. As an example, a data processing circuit is discussed that includes: a noise predictive filter circuit, a mean calculation circuit, a summation circuit, and a filter tap adaptation circuit. The noise predictive filter circuit is operable to perform a noise predictive filtering process on a data input based upon a selector input to yield a noise predictive output. The selector input is derived from a data detector output. The mean calculation circuit is operable to average two or more instances of the noise predictive output to yield a mean output. The summation circuit is operable to subtract the mean output from the noise predictive output to yield a sum output. The filter tap adaptation circuit is operable to adaptively calculate a filter tap based at least in part on a value derived from the data input and a value derived from the sum output.

Abstract: A system and method for processing a signal with a filter employing FIR and/or IIR elements. The required controller function is decomposed into primary FIR and/or IIR elements and a compensation filter is provided to address the latency in the primary elements, which would result in undesired operation of the filter. Several configurations of suitable filters are discussed, including multi-rate filters and filters with reduced power requirements.

Abstract: Systems and methods are provided for an adjustable filter engine. In particular, an electronic system is provided that can include a focus module, memory, and control circuitry. In some embodiments, the focus module can include an adjustable filter engine and a motor. By using the adjustable filter engine to generate a filter with a large number of filter coefficients, the control circuitry can accommodate a variety of system characteristics. For example, by generating a set of cumulative coefficients and re-arranging the order of the cumulative coefficients, the control circuitry can reduce the bit-width requirements of the adjustable filter engine hardware. For instance, the control circuitry can reduce the number of multipliers required to perform a convolution between an updated filter and one or more input signals. In some embodiments, the updated filter can be generated to reduce oscillations of the motor movement due to a new position request.

Abstract: A method and apparatus for receiving data in high-speed applications wherein an analog-to-digital converter (ADC) samples a received signal and a data decoder implemented with a tree search algorithm detects the bits of the sampled data for timing recovery. In some embodiments, a Viterbi detector is implemented to provide accurate bit detection for data output while tree search detected data is used to determine the optimal sampling phase for the ADC. In some embodiments, after the phase acquisition stage of timing recovery has completed, the tree search decoder may decrease the rate of data detection to maintain phase tracking.

Abstract: Disclosed are a receiving equalization device and a method thereof, the receiving equalization device including a subtracter to output a first output signal, an eye monitor block to obtain a sampling timing by using the output first signal, and a slicer to generate a sampling signal by sampling the first output signal based on the sampling timing, and to return the generated sampling signal to the subtracter via a feedback filter or an algorithm determining block.

Abstract: Physiological signals are denoised. In accordance with an example embodiment, a denoised physiological signal is generated from an input signal including a desired physiological signal and noise. The input signal is decomposed from a first domain into subcomponents in a second domain of higher dimension than the first domain. Target subcomponents of the input signal that are associated with the desired physiological signal are identified, based upon the spatial distribution of the subcomponents. A denoised physiological signal is constructed in the first domain from at least one of the identified target subcomponents.

Abstract: A testing system tests an audio connection between an audio source and a loudspeaker. The system includes a loudspeaker that converts a reference signal into a sound. An adaptive filter processes the reference signal to minimize an error signal. A decision circuit analyzes the error signal and the received signal to determine signal correlation. When the signals are not correlated, a defect is detected.

Abstract: A method and system that remove an unwanted signal and its harmonics from an input signal in a computationally efficient manner are disclosed. Embodiments include processing the FFT matrix to selectively zero-out rows of the matrix before multiplying the matrix with the Inverse FFT (IFFT) matrix. The resulting product (which is a sparse matrix) is then used to generate coefficients for a linear Finite Impulse Response (FIR) filter to process the input. The filtered output signal has the unwanted signal and its harmonics removed with minimal effect on a desired signal. The method produces a stable, physically realizable filter, requiring fewer computations than current methods.

Abstract: A signal processing apparatus includes a signal processing unit configured to carry out signal processing on a single-carrier signal and a multi-carrier signal by making use of a plurality of common filters shared by the single-carrier signal and the multi-carrier signal.

Abstract: A finite impulse response filter (FIR) for processing a communication channel. The FIR comprises a delay line, a tap processor and a summer. The delay line has “N” taps to successive portions of the communication channel. The delay line shifts the successive portions of the communication channel once in each symbol processing interval. The tap processor subjects each of the “N” taps to a first scaling utilizing first scaling coefficients associated with filtering the current symbol interval and further subjects at least one of the “N” taps to a second scaling by a second scaling coefficient associated with filtering the prior symbol interval. The summer generates in each symbol interval a filtered output comprising a sum of the “N” scaled taps from the first scaling in the prior symbol interval and the second scaling of the at least one tap in the current symbol interval, thereby increasing an order of the FIR without corresponding increase in an order of the delay line.

Abstract: Various embodiments of the present invention provide systems and methods for data processing. As an example, a data processing circuit is disclosed that includes: a noise predictive filter circuit, a scaling factor adaptation circuit, and a scaling factor application circuit. The noise predictive filter circuit is operable to perform a noise predictive filtering process on a data input based on a filter tap to yield a noise filtered output. The scaling factor adaptation circuit is operable to calculate a scaling factor based at least in part on a derivative of the noise filtered output. The scaling factor application circuit is operable to apply the scaling factor to scale the noise filtered output.

Abstract: An adaptive notch filter adaptively modifies one or more notch frequencies. The adaptive notch filter includes a transfer function equivalent to coupling j stages (where j is a natural number equal to or greater than 1) of adaptive notch filter units each having a transfer function Hf(z) expressed by a predetermined equation and having a different notch frequency.

Abstract: A signal detection apparatus detects the frequency of an input signal without using a PLL. The detection apparatus includes a first and a second orthogonalizer, a phase difference calculator and an integrator, to control the variable coefficient a1 of a band-pass filter. Information e[k]=M·sin(?) representing the phase difference ? between the input data x[k] and the output data y[k] is calculated with the first and second orthogonalizers and the phase difference calculator. The sign of e[k] is inverted and a predetermined integral calculation is performed with the integrator, and the calculated integral value is set as the coefficient a1 of the band-pass filter. Every time input data x[k] is input, the coefficient a1 is changed by reducing it when e[k]>0 and increasing it when e[k]<0. Thus, the frequency of the output signal of the band-pass filter is matched to the input signal.

Abstract: Adaptive processing of an input signal is achieved by offline analysis, with inline processing comprising an adaptive filter. The method comprises passing the input signal through an adaptive time domain filter to produce an output signal. The input signal and/or output signal is used as an offline analysis signal. The analysis signal is transformed into a transform domain (eg frequency domain) to produce a transformed analysis signal. The transformed analysis signal is analyzed, for example by ADRO, to produce a plurality of desired gains each corresponding to a respective transform domain sub-band. A time domain filter characteristic is synthesized to at least approach the desired gains. The adaptive filter is updated with the synthesized filter characteristic. Minimum phase adaptive filter techniques are found to possess particular benefits in this scheme.

Abstract: Methods and systems for multi-input IIR filters with error feedback are disclosed. By using multiple-inputs to generate multiple outputs during each iteration, a multi-input IIR filter in accordance with the present invention has greatly increased throughput. Furthermore, the addition of a multi-variable error feedback unit in accordance with the present invention in a multiple-input IIR filter can greatly increase the accuracy of the multi-variable IIR Filter.

Abstract: Physiological signals are denoised. In accordance with an example embodiment, a denoised physiological signal is generated from an input signal including a desired physiological signal and noise. The input signal is decomposed from a first domain into subcomponents in a second domain of higher dimension than the first domain. Target subcomponents of the input signal that are associated with the desired physiological signal are identified, based upon the spatial distribution of the subcomponents. A denoised physiological signal is constructed in the first domain from at least one of the identified target subcomponents.

Abstract: A digital signal processing system comprising: an input terminal to receive an input signal that includes a distorted component and an undistorted component, the input signal having a sampling rate of R; and an adaptive self-linearization module coupled to the input terminal, to perform self-linearization based at least in part on the input signal to obtain an output signal that is substantially undistorted, wherein: the adaptive self-linearization module is to generate a replica distortion signal that is substantially similar to the distorted component, the generation being based at least in part on a target component having a sampling rate of R/L, L being an integer greater than 1; the adaptive self-linearization module includes a first digital signal processor (DSP) that is adapted to obtain a filter transfer function that approximates a system distortion transfer function, and a second DSP that is configured using configuration parameters of the first DSP.

Abstract: An adaptive hybrid system is coupled to a loop for adjusting trans-hybrid loss. The system comprises a fixed portion comprising a first receiver transfer function block and a first hybrid transfer function block. The fixed portion is configured to receive a far-end signal and mitigate frequency dependent attenuation experienced by the far-end signal. The system also comprises a variable portion comprising a second receiver transfer function block and a second hybrid transfer function block configured to subtract a transmit echo from the received far-end signal.

Abstract: A read channel of a magnetic data storage device includes a filter to provide equalization of the signal being detected from the magnetic media. The filter utilizes coefficients for the filter response. The filter coefficients may drift sideways over time. The drift is detected and a correction is implemented by imposing a leakage on the coefficients to re-center the filter response. The leakage sign differs depending on the direction of drift detected.

Abstract: Aspects of a method and system for unconstrained frequency domain adaptive filtering include one or more circuits that are operable to select one or more time domain coefficients in a current filter partition. A value may be computed for each of the selected one or more time domain coefficients based on a corresponding plurality of frequency domain coefficients. The corresponding plurality of frequency domain coefficients may be adjusted based on the computed values. A subsequent plurality of frequency domain coefficients in a subsequent filter partition may be adjusted based on the computed values. Input signals may be processed in the current filter partition based on the adjusted corresponding plurality of frequency domain coefficients. A time-adjusted version of the input signals may be processed in a subsequent filter partition based on the adjusted subsequent plurality of frequency domain coefficients.

Abstract: An apparatus and method for implementing an equalizer which combines the benefits of a decision feedback equalizer (DFE) with a maximum-a-posterori (MAP) equalizer (or a maximum likelihood sequence estimator, MLSE) to provide an equalization device with significantly lower complexity than a full-state MAP device, but which still provides improved performance over a conventional DFE. The equalizer architecture includes two DFE-like structures, followed by a MAP equalizer. The first DFE forms tentative symbol decisions. The second DFE is used thereafter to truncate the channel response to a desired memory of L1 symbols, which is less than the total delay spread of L symbols of the channel. The MAP equalizer operates over a channel with memory of L1 symbols (where L1<=L), and therefore the overall complexity of the equalizer is significantly reduced.

Abstract: A system includes a first powered apparatus having a first analog signal with a fundamental frequency; and a second apparatus providing load diagnostics or power quality assessment of the first apparatus from a second digital signal. The second apparatus includes an input of the first analog signal, an output of the second digital signal, a processor, an adaptive filter executed by the processor, a digital-to-analog converter, and an analog-to-digital converter. The adaptive filter routine outputs a third digital signal as a function of the second digital signal and plural adaptive weights. The digital-to-analog converter inputs the third digital signal and outputs a fourth analog signal representative of an estimate of a fundamental frequency component of the first analog signal. The analog-to-digital converter inputs a difference between the first and fourth analog signals, and outputs the second digital signal representative of the first analog signal with the fundamental frequency component removed.

Abstract: An equalization filter is provided for solving the problem in which there is a limited range in which compensated for distortion of a transmission signal can be made. Measuring instrument 104 measures a distortion quantity which characterizes distortion of the transmission signal. Comparator 105a generates a differential signal which indicates the difference between the transmission signal and a compensation signal. Delay device 105b delays the differential signal based on the distortion quantity measured by measurement instrument 104 and generates the compensation signal.

Abstract: Methods and systems are provided for coarse phase estimation for highly-spectrally efficient communications. An example method may include, equalizing, in a receiver, a received inter-symbol correlated (ISC) signal to generate an equalized ISC signal. A phase adjustment signal may be generated based on an ISC feedback signal. The ISC feedback signal may be generated using a sequence estimation process and a non-linearity function. A phase of the equalized ISC signal may be adjusted using the generated phase adjustment signal, to generate a phase adjusted partial response signal. The phase adjustment signal may be generated based on a phase difference between the equalized ISC signal and the partial response feedback signal. At least one ISC vector may be generated by buffering samples of the phase adjusted ISC signal.

Abstract: An embodiment of a method for control of signal level is disclosed. In such an embodiment, a number for a pre-cursor set, a number for a cursor set, and a number for a post-cursor set are set corresponding to a weighted contribution of a pre-cursor symbol, a weighted contribution of a cursor symbol, and a weighted contribution of a post-cursor symbol, respectively, for the signal level. A number associated with a high-impedance set is determined. The number associated with the high-impedance set is determined by subtracting the number for the pre-cursor set, the number for the cursor set, and the number for the post-cursor set from a total available amount of units. The high-impedance set provides no weighted contribution to the signal level. Data is transmitted using the signal level set responsive to the pre-cursor set, the cursor set, and the post-cursor set.

Abstract: The system and method for least mean fourth adaptive filtering is a system that uses a general purpose computer or a digital circuit (such as an ASIC, a field-programmable gate array, or a digital signal processor that is programmed to utilize a normalized least mean fourth algorithm. The normalization is performed by dividing a weight vector update term by the fourth power of the norm of the regressor.

Abstract: Decision feedback equalization (DFE) circuitry and method for equalizing data signals over a wide range of data rates. By using delayed and controlled versions of the recovered data clock to retime the equalized data signal for feedback via the DFE taps, correct feedback signal timing is maintained and jitter tolerance is increased at high data rates.

Abstract: Apparatus and methods for electronic hardware resource management in video processing are provided. A hybrid filter is controllable to apply either Finite Impulse Response (FIR) filtering or Infinite Impulse Response (IIR) filtering for vertical filtering of a video image during a resizing process. A scale factor by which the video image is to be resized in the resizing process is determined, and the hybrid filter is controlled to apply FIR filtering for the vertical filtering where the determined scale factor satisfies a first condition relative to a threshold value ST or to apply IIR filtering for the vertical filtering where the determined scale factor satisfies a second condition relative to the threshold value ST. The first and second conditions are different and exclusive, such that only one type of filtering is applied. This hybrid filtering approach uses each type of filtering to avoid defects of the other.

Abstract: Disclosed is an improved method and apparatus for estimating and applying a step size value for a least mean squares echo canceller. A power estimate of an excitation signal is compared to a reference power level to determine a shift adjustment. The shift adjustment is added to a reference shift amount to determine a shift amount. The product of an excitation signal and an error signal is then calculated and the product is stored in a memory register comprising a plurality of bits. The bits stored in the memory register are shifted either left or right based upon the shift amount. The shift adjustment may be based in part upon the ratio of the excitation signal power estimate and the reference power level.

Abstract: The MPEG2 Advanced Audio Coder (AAC) standard limits the number of filters used to either one filter for a “short” block or three filters for a “long” block. In cases where the need for additional filters is present but the limit of permissible filters has been reached, the remaining frequency spectra are simply not covered by TNS. Two solutions are proposed to deploy TNS filters in order to get the entire spectrum of the signal into TNS. The first method involves a filter bridging technique and complies with the current AAC standard. The second method involves a filter clustering technique. Although the second method is both more efficient and accurate in capturing the temporal structure of the time signal, it is not AAC standard compliant. Thus, a new syntax for packing filter information derived using the second method for transmission to a receiver is also outlined.

Abstract: A feedback module is defined to receive as input a set of data sample signals and a set of reference sample signals. Each of the data and reference sample signals is generated by sampling a differential signal having been transmitted through a FIR filter. The feedback module is defined to operate a respective post cursor counter for each post cursor of the FIR filter and update the post cursor counters based on the received sets of data and reference sample signals. Also, the feedback module is defined to generate a tap weight adjustment signal for a given tap weight of the FIR filter when a magnitude of a post cursor counter corresponding to the given tap weight is greater than or equal to a threshold value. An adaptation module is defined to adapt a reference voltage used to generate the reference sample signals to a condition of the differential signal.