155 results about "Equaliser" patented technology

A system, method, apparatus and computer code for communicating a training sequence for initializing an equalizer in a digital receiver are provided including receiving a digital signal containing data to be broadcast from a digital RF transmitter and inserting the training sequence into the digital signal deterministically such that a predetermined sequence of symbols are communicated to the receiver.

A plurality of computer nodes communicate using seemingly random Internet Protocol source and destination addresses. Data packets matching criteria defined by a moving window of valid addresses are accepted for further processing, while those that do not meet the criteria are quickly rejected. Improvements to the basic design include (1) a load balancer that distributes packets across different transmission paths according to transmission path quality; (2) a DNS proxy server that transparently creates a virtual private network in response to a domain name inquiry; (3) a large-to-small link bandwidth management feature that prevents denial-of-service attacks at system chokepoints; (4) a traffic limiter that regulates incoming packets by limiting the rate at which a transmitter can be synchronized with a receiver; and (5) a signaling synchronizer, that allows a large number of nodes to communicate with a central node by partitioning the communication function between two separate entities.

Graphic equalizer as a cascade of equalization filters with the gain of each individual filter compensated for the gain leakage from other filters. A linear approximation allows individual filter gains to be set to give desired frequency responses.

Tap coefficients for fractionally-spaced equalizers are updated iteratively using error statistics from an input bit stream and an output bit stream, such as from a forward error correction circuit. The process continues until the errors converge to a sufficiently small number. Knowledge of the error patterns are used apriori to adaptively change the tap coefficients in a feedforward filter.

A method of and apparatus for optical signal dispersion compensation uses tunable optical (ODC) and electronic (EDC) dispersion compensation. The ODC provides significant first-order dispersion compensation, but leaves significant high-order dispersion and transmissivity ripple effects. The EDC is used to provide compensation for the high-order dispersion and transmissivity ripple from the ODC.

A decision feedback equalizer (DFE) architecture uses feedback samples that are over-sampled with respect to the symbol rate. On-baud feedback samples are quantized with a slicer, while off-baud samples are linear, IIR samples. Both forward and feedback filters are fractionally-spaced, but adapted only at the baud instances.

An improved method and arrangement for altering musical dynamics of a sound S included in a sound signal is disclosed. The altering of the musical dynamics is performed by filtering and amplification of the sound signal. The filtering is performed by the use of a parametric equalizer, the parametric equalizer having a first gain G1 and a resonance frequency f, being related to a pitch frequency fp of said sound S. The amplification is performed by an amplifier amplifying the sound signal with a second gain G2, the second gain G2 being dependent on the first gain G1.

Constrained tap weights of a decision feedback equalizer are determined according to the channel impulse response of a channel and a constraint function. The constraint function is differentiable and is an approximation of a non-differentiable tap weight constraint function. The tap weight constraint function may have a constraint value M that is a function of a mean squared error of the estimated mean squared error at the output of the decision feedback equalizer.

Equaliser (4) or phase plug, for electro-acoustic transducers (1), having at least a central hole (40), and or a first series of openings (41), or slots, in the from of concentric rings, coaxial to an axis (4a) of symmetry of the phase plug (4). Said phase plug is characterised in that it has at least a second series of radial slots (42) positioned externally to said hole (40) or to said first series of slots (41). In the phase plug (4) has a third series of radial slots (43) interposed between said first and second series (41;42).

A vector decoder for detecting symbols transmitted along a communication channel is disclosed, as is a method for decoding symbols. The decoder includes at least two pairs of decision feedback equalizers and noise predictors operating on a plurality M of vectors of symbols being detected, a path storage unit, a metric determiner a sorter and a path builder. The vectors are ordered in accordance with quality levels and each noise predictor estimates the noise associated with its associated vector. The path storage unit stores M of the ordered vectors, each vector being of length L. The metric determiner is connected to the output of all of the decision feedback equalizers and the noise predictors and produces a performance metric for each of Q branches extending from each vector. Each branch is associated with one of Q possible symbol values. The sorter is connected to the output of the metric determiner and sorts data, formed of a) the symbol values and b) the vectors which they extend, in accordance with their performance metrics. The sorter also orders, from the M*Q performance metrics, the data having the M best performance metrics. The path builder is connected to the output of the sorter and associates the ordered data with the appropriate ones of the previously stored ordered vectors.

An apparatus including a bang-bang clock data recovery module and a decision feedback equalizer. The decision feedback equalizer is coupled with the bang-bang clock data recovery module. The apparatus is configured to reduce an effect on a settling point of the bang-bang clock and data recovery module due to coupling between the bang-bang clock and data recovery module and the decision feedback equalizer.

An equalizer comprises: an FIR; a trellis decoder coupled to the FIR; a mapper coupled to the trellis decoder; and a decision feedback equalizer coupled to the mapper. The decision feedback equalizer receives the mapped and scaled output of the trellis decoder as input and an error signal is generated by subtracting an output of the decision feedback equalizer from the input to the trellis decoder.

The invention discloses a clock recovery and adaptive equalizer combination device and method in coherent light communication, relates to the technical field of coherent light communication, and adopts a new equalizer structure and algorithm to realize clock recovery and adaptive equalization at the same time. Specifically, a two-stage real number equalization structure and a second-stage adaptivecoefficient are adopted to calculate a timing error so as to realize clock recovery, polarization demultiplexing and dispersion compensation. According to the structure, an independent clock recoverymodule is omitted, compared with a traditional structure, the calculation complexity is reduced, the clock recovery accuracy is improved, and the tolerance to IQ data imbalance of a receiving end isimproved. The two-stage equalization of the structure adopts real number equalization, so that the IQ two paths of data are not combined together any more in the equalization process, and the problemthat the system performance is greatly influenced under the condition that IQ has time delay can be effectively solved.

An adaptive equalizer comprises a decision device; a decision feedback equalizer coupled to the decision device; an FIR filter coupled to the decision device; and a trellis decoder coupled to the decision device, adapted to provide a reliability output and a decoded output. An error signal is generated by subtracting an output of the decision feedback equalizer from an output of the decision device, the error signal being used to update coefficients of the taps of the FIR filter and the decision feedback equalizer. A magnitude of the change to the coefficients is selected based at least in part the reliability output of the trellis decoder.

The invention provides a self-adaptive equalization method, a device and an equalizer, which relate to the field of communication. The self-adaptive equalization method comprises steps: frame positioning information about polarization state data outputted by the equalizer and a tap coefficient updated after blind equalization processing are acquired; according to the frame positioning information and performance information of the equalizer, a current operating state of the equalizer is determined; according to the current operating state of the equalizer and the tap coefficient updated after blind equalization processing, the tap coefficient or a leakage factor in blind equalization processing are correspondingly adjusted; and according to the adjusted tap coefficient or the leakage factor, self-adaptive convergence is carried out until the equalizer operates stably in a stable state. The solution of the disclosure solves the problem in an existing technical solution that the efficient and stable operation of a self-adaptive equalizer cannot be realized during a convergence process.

A wafer level burn-in method for static-random access memory. The SRAM memory has a plurality of word lines and a plurality of bit lines. The SRAM memory also has pull up circuits and equalizer circuits connected to various bit lines. All the word lines are switched on for testing any leakage in the gate dielectric layer. A high potential is applied to a bit line of every bit line pairs and a low potential is applied to the other bit line of the bit line pairs. The pull-up circuits and the equalizer circuits are shut down. The current at a steady state is used to judge the normality of an SRAM chip.

A method for selecting which tap coefficients of a programmable finite-impulse-response (FIR) equalizer to fix is disclosed. In one embodiment, such a method includes performing an initial calibration to determine an initial value for each tap coefficient of a FIR equalizer. These initial values may be used to produce a first waveform. The method then performs an operation on the first waveform to produce a second waveform comprising multiple lobes. The second waveform is then analyzed to determine one or more lobes of the second waveform that have the largest area. The method then fixes coefficients of one or more taps that are closest to the lobe or lobes having the largest area. A corresponding apparatus and computer program product are also disclosed.

A method for adaptive selection of floating taps in a decision feedback equalizer including the steps of (A) determining values for a predefined metric for tap positions within a range covered by a decision feedback equalizer (DFE) and (B) setting one or more floating taps of the DFE to tap positions based upon the values of the predefined metric.

The present universal wearable computing device relates to a hearing assistance system, device, method, and apparatus that provide a discreet approach to user hearing assistance, without relying on a conventional hearing aid. The hearing assistance system and the requisite electronics may be incorporated into frames that also function as eyeglasses with earphone(s) that may be connected to the frame to assist user hearing. An earphone may be configured with minimal electronics, such that a power source enable sound transmissions to the ear, is provided by a connection to the frame of the eyeglasses. In another example, the earphone is configured without any electronics and sound is transmitted to the user / listener's ear(s) via a psychoacoustic system. The sound quality of the transmissions to the earphones may be optimized using a tuning / equalizer application operating from a computing device, such as an app on a mobile device. The tuning / equalizer application can be used by the user / listener to optimize volume input levels to the earphone(s).

In one embodiment, an apparatus has an equalizer, a tap position locator, and a tap weight updater. The equalizer has a plurality of floating taps. The tap position locator generates metrics for a set of possible tap positions of the equalizer. Each possible tap position corresponds to a different tap weight, and the metrics are generated without updating the tap weights for all of the possible tap positions in the set. Further, the tap position locator selects a subset of possible tap positions from the set based on the metrics. The tap weight updater updates a subset of the tap weights corresponding to the selected subset of possible tap positions, and applies the updated subset of tap weights to the plurality of floating taps.

The invention relates to the technical field of automatic control, and provides a motor excitation signal processing method and device. The method comprises the steps: carrying out the displacement prediction according to an original excitation signal outputted to a motor, and obtaining the original displacement signal of the motor; obtaining the maximum displacement threshold of the motor, and calculating an equalization displacement signal according to the maximum displacement threshold and the original displacement signal; and processing the equalization displacement signal through a second-order equalizer to obtain an equalization excitation signal for driving motor. The displacement of the motor can be within the maximum displacement threshold, and the problems of motor part damage and touch effect reduction caused by overlarge displacement of the motor are solved.

An optical medium reproduction apparatus for optically reproducing an optical medium where a plurality of tracks are formed, including: a detection unit for splitting a cross section of a beam returned from the optical medium into a plurality of regions and for forming respective detection signals of a plurality of the regions; a multiple input adaptive equalizer having a plurality of adaptive equalizer units, wherein the respective detection signals of a plurality of the regions are inputted into the plurality of adaptive equalizer units, and the outputs of the plurality of adaptive equalizer units are computed to from equalization signals; a binarization unit for binarizing the equalization signals to provide binary data; and an equalization error computing unit for determining an equalization error from equalization target signals provided based on the binary data from the binarization unit and the equalization signals outputted from the multiple input adaptive equalizer, and providing the adaptive equalizer units with the equalization error as control signals for adaptive equalization.

The invention discloses a complex channel equalizer design method based on a complex value forward neural network, and relates to the field of artificial intelligence and communication. The complex channel equalizer design method based on the complex value forward neural network comprises the steps: employing a selectable search direction training method to train the complex value forward neural network, and enabling a target function value to be reduced as much as possible after each time of training, wherein the trained complex value forward neural network serves as a channel equalizer and is used in a digital communication system. The complex channel equalizer design method based on the complex value forward neural network has the beneficial effects that in the training process, a groupof direction factors are introduced to construct a plurality of search directions, the optimal training direction is selected from the search directions by adopting a breadth-first search strategy, and the optimal search direction is selected from the search directions. The high-efficiency training of the complex number forward neural network is completed, so that the design of the complex channel equalizer is realized, and excellent performance is obtained.

An approach is provided for customizing output settings for media playback. An equalizer or media player determines one or more segments of a media track and associates the one or more segments with one or more corresponding output settings. Playback of the media track at a device is then based, at least in part, on the one or more output settings.

A system and method for providing a normalized security list including a first module configured to generate a first normalized security list of user identifications within a network and a second module configured to generate a second normalized security list of user identifications within the network. The system and method may also include an equalizer module configured to compare the first normalized security list with the second normalized security list, equalize the first normalized security list based on the second normalized security list, and equalize the second normalized security list based on the first security list. The system and method may also include a processing module configured to perform an audit of user identifications within the network by processing the first equalized normalized security list and the second equalized normalized security list and generating audit results based on the processing.

The invention discloses a super Nyquist system symbol estimation method combined with equalization and interference cancellation. The super Nyquist system symbol estimation method combined with equalization and interference cancellation comprises the following steps: obtaining an input symbol; training an equalizer; judging whether the equalizer training number is less than an equalizer thresholdor not; calculating equalized symbol; calculating an interference cancellation factor of the super Nyquist system; performing iteration for the first time to eliminate inter-symbol interference; performing iteration to eliminate the inter-symbol interference; judging whether the number of iterations is less than an estimation threshold or not; and obtaining the final estimated symbol. The super Nyquist system symbol estimation method combined with equalization and interference cancellation can be used to eliminate the inter-symbol interference in the super Nyquist system, thereby estimating the transmitted symbols; compared with methods in the prior art, the invention can make the super Nyquist system have better bit error rate performance even in the case of more serious inter-symbol interference; and the symbol estimation of the invention is low in complexity and more practical.

A method for estimating at least one optical link parameter using a Viterbi equalizer (1) generating averaged values (M) of signal amplitudes of a distorted optical signal transmitted through an optical link (2) for a set of decided bit patterns (a, b, c, d, . . . ), the method comprising the following steps: forming a first parameter set (S) characteristic of a distorted signal sequence using the averaged values (M(a), M(b), M(d), M(g)) for a given sequence (abdg) of bit patterns (a, b, c, d, . . . ), comparing the first parameter set (S) to a plurality of reference parameter sets (R1, R2, . . . ) characteristic for reference signal sequences of the same sequence (abdg) of bit patterns (a, b, c, d, . . . ), each of the reference parameter sets (R1, R2, . . . ) having a known value of the at least one optical link parameter, and selecting the reference parameter set (R) with the closest correlation to the first parameter set (S), the known value of the at least one optical link parameter of the selected reference parameter set (R) being used as an estimate for the at least one optical link parameter. A computer program product comprising a software or a hardware implementing the method.

The embodiment of the invention discloses an earphone equalizer adjusting method and device, a server and a medium. The method comprises the steps of obtaining current user portrait data of a currentuser; inputting the current user portrait data into a pre-trained target music preference prediction model to obtain a music preference evaluation value of at least one music type; and obtaining a target equalizer parameter according to the music preference evaluation value of the music type, so that the earphone adjusts the equalizer parameter of the earphone based on the target equalizer parameter. According to the technical scheme of the embodiment of the invention, personalized customization of the earphone equalizer parameters for the user is realized, so that the user experience is improved.

The present invention provides a variable gain amplifier and a continuous time linear equalizer, the variable gain amplifier comprising: a gain control voltage generation circuit for generating a gaincontrol voltage according to a differential input reference voltage; and the signal pat connected to the gain control voltage generation circuit an used for controlling the voltage gain of the signalpath according to the gain control voltage and realizing differential amplification on the differential input signal according to the voltage gain to generate a differential output signal. Accordingto the variable gain amplifier, the problem that the voltage gain of an existing variable gain amplifier is easily subjected to power supply voltage, chip temperature and CMOS is solved.

The invention discloses an a self-adaptive equalizer compensation method of vehicle-mounted sound equipment and belongs to the technical field of signal processing systems. The self-adaptive equalizercompensation method of the vehicle-mounted sound equipment comprises the following steps in sequence: after a white noise signal is played, a microphone receives a test signal propagated in a vehicle; carrying out smoothing processing on the test signal to form a secondary test signal; calculating a compensation gain signal of the secondary test signal relative to the white noise signal, smoothing the compensation gain signal to obtain a secondary compensation gain signal, setting a graphic equalizer, and inputting a source audio signal to the graphic equalizer to superpose the secondary compensation gain signal on the source audio signal to obtain a fitting response signal. High flexibility is achieved when the sound signals are corrected, the propagation loss of the sound signals in the air can be well compensated for, distortion is avoided, and the method can be suitable for the conditions of uniform and non-uniform frequency and can also be migrated into other sound systems in aclosed space to be used.