ИЗМЕРЕНИЕ СХОДИМОСТИ АДАПТИВНЫХ ФИЛЬТРОВ

Изобретение относится к адаптивным фильтрам, обеспечивающим компенсацию эхо-сигналов. Измерение степени сходимости в устройстве адаптивной фильтрации обеспечивается на основе сравнения степени адаптации, достигнутой в устройстве адаптивной фильтрации, за предварительно определенный период времени с нормирующим значением, накопленным за тот же самый период времени. Может быть использована дополнительная обработка сигнала, которая может быть изменена или отменена, на основе полученного значения степени сходимости. Технический результат: определение момента нахождения адаптивного фильтра в состоянии сходимости на основе индикации сходимости. 2 с. и 24 з.п. ф-лы, 5 ил.

ПОДАВЛЕНИЕ СИГНАЛА ДЛЯ ИССЛЕДОВАНИЯ СПЕКТРА ДРУГОГО СИГНАЛА

Заявленный способ предназначен для подавления выделенного радиочастотного сигнала для исследования спектра по меньшей мере одного другого радиочастотного сигнала. Технический результат заключается в сокращении времени отклика. Способ содержит прием 602 смешанного сигнала 108 аналого-цифровым преобразователем 112. Смешанный сигнал содержит множество отдельных сигналов 104а-104n из различных источников 106а-106n сигналов. Смешанный сигнал оцифровывают 606 аналого-цифровым преобразователем. Генерируют идентичные первый оцифрованный сигнал 116 и второй оцифрованный сигнал 120. Первый оцифрованный сигнал задерживают 608 на заданное время задержки, а второй оцифрованный сигнал обрабатывают 610 в нейроморфическом процессоре 118 для обработки сигналов для выделения выделенного сигнала 122. Заданное время задержки соответствует задержке, встроенной 204 в нейроморфический процессор для обработки сигналов. Фазовую задержку и амплитуду выделенного сигнала регулируют 614 на основании фазовой задержки ...

АДАПТИВНЫЙ КОРРЕКТИРУЮЩИЙ ФИЛЬТР

FIELD: electronic engineering. SUBSTANCE: filter is built up of two partial filters TF1 and TF2 whose discrimination ratios are variable by means of discrimination-ratio control circuit (CORR), so as to form, for example, approximately inverting filter for time-varying transmission channel which enables both non-decimating mode of operation when polling frequency corresponds to rate of characters and decimating one when polling frequency satisfies count theorem by means of ON/OFF operations. Wiring between ratio multipliers 31-41 of circuits of frequency filters TF1, TF2 and partial correlators 80-90 of ratio control circuit (CORR) is invariant relative to mode-of- operation change-over operations. EFFECT: reduced quantity of switches or selector switches. 2 cl, 2 dwg 99 тс ПЧ ГЭ (19) РОССИЙСКОЕ АГЕНТСТВО ПО ПАТЕНТАМ И ТОВАРНЫМ ЗНАКАМ ВИ” 2 125 764 ^^ Сл (51) МПК Н 03 Н 17/06, 21/00 12) ОПИСАНИЕ ИЗОБРЕТЕНИЯ К ПАТЕНТУ РОССИЙСКОЙ ФЕДЕРАЦИИ (21), (22) Заявка: 93045352/09, 11.09.1993 (30) Приоритет: 18.09.1992 ОЕ Р4231309.0 (46) Дата публикации: 27.01.1999 (56) Ссылки: 5068873 А, 26.11.91. Нолль Тобиас Гебхард. Диссертация "Проект архитектуры и схемы цифрового адаптивного корректора для цифровой направленной радиосвязи с локально систолическим Саггу-Зауе-Аггауз в КНОП технологии", Рурский университет, Бохард, 1989, с. 50. ЗЧ 1319295 АЛ, 23.06.87. $4 1388896 АЛ, 15.04.88. ЗЦ 1432730 АЛ, 23.10.88. $Ц 1655309 АЗ, 07.06.91. ЕР 0331963 А2, 13.09.89. 4$ 4417314 АР, 22.11.83. 4$ 4741068 А, 24.05.88. .Р 1-20805 В4, 18.04.89. ЕР 0201281 А2, 12.11.86. ЦЗ 5068873 А, 26.11.91. ЕР 0305708 А2, 08.03.89. (98) Адрес для переписки: 103735 Москва, ул.Ильинка 5/2, Союзпатент патентному поверенному Дудушкину С.В. (71) Заявитель: Сименс АГ (0Е) (72) Изобретатель: Тобиас Нолль (0Е), Эрик Де Ман (ВЕ) (73) Патентообладатель: Сименс АГ (ОЕ) (54) АДАПТИВНЫЙ КОРРЕКТИРУЮЩИЙ ФИЛЬТР (57) Реферат: Изобретение относится к адаптивному корректирующему фильтру с двумя частичными фильтрами (ТЕЛ, ТЕ2), ...

СПОСОБ ОБРАБОТКИ СИГНАЛОВ ВЧ ДИАПАЗОНА КАСКАДОМ АДАПТИВНЫХ ФИЛЬТРОВ РАЗЛИЧНОЙ ИНЕРЦИОННОСТИ С ОБЩЕЙ ОБРАТНОЙ СВЯЗЬЮ ПО РЕШЕНИЮ

Способ обработки сигналов, заключающийся в том, что сигнал обрабатывают широкополосной адаптивной антенной системой (ШААС), функционирующей по алгоритму Винера и критерию минимума среднего квадрата ошибки (МСКО), выход с ШААС поступает на схему принятия решения, отличающийся тем, что выходной сигнал с широкополосной антенной системы обрабатывается дополнительным фильтром, управляемым общим с ШААС сигналом оценки эталонаи независимым сигналом ошибки второго каскада е, формируемым разностьюи; при этом выходной сигнал с ШААС поступает на дополнительный фильтр нерекурсивной структуры с рекурсивным алгоритмом наименьших квадратов; адаптивный алгоритм ШААС, управляемыйи независимым сигналом ошибки первого каскада e, формируемым разностьюи y(k), должен иметь инерционность выше, чем у дополнительного фильтра, в котором интервал стационарности Lравный 250 (символьных посылок) должен быть меньше Lинтервала стационарности широкополосной антенной системы, равного 600 (символьных посылок), т.е. L<L, обеспечивая этим условия различной инерционности каскадов; длина линии задержки дополнительного фильтра должна быть меньше L - длины линии задержки ШААС; равняются 5 и 19 символьным посылкам соответственно; преобразованный из y(k) всигнал только после дополнительного фильтра поступает на схему принятия решения, которая вырабатывает общую оценку сигнала, обеспечивающую объединяющую управляющую связь между ШААС и дополнительным фильтром, при этом сигналы ошибки e(k) и e(k) для ШААС и дополнительного фильтра различны; общая оценка сигнала эталонав схеме принятия решения для системы адаптивных фильтров вырабатывается по единому характеристическому критер РОССИЙСКАЯ ФЕДЕРАЦИЯ (19) RU (11) (51) МПК H03H 9/00 (13) 2014 147 617 A (2006.01) ФЕДЕРАЛЬНАЯ СЛУЖБА ПО ИНТЕЛЛЕКТУАЛЬНОЙ СОБСТВЕННОСТИ (12) ЗАЯВКА НА ИЗОБРЕТЕНИЕ (21)(22) Заявка: 2014147617, 26.11.2014 (71) Заявитель(и): Тумачек Александр Сергеевич (RU) Приоритет(ы): (22) Дата подачи заявки: 26.11.2014 (43) Дата публикации заявки: 20 ...

Schnelles Konvergenzsystem eines adaptiven Filters zur Erzeugung eines zeitabhängigen Signals zur Kompensation eines primären Signals

ADAPTIVE AUDIOSYSTEME UND TONWIEDERGABESYSTEME

AKTIVES DIGITALES ANNULATIONSSYSTEM "VIRTUELLE ERDE"

Kombinierte und vereinfachte Multiplexierung in Zusammenhang mit Zitteranalog-Digitalwandler

Vorrichtung und Verfahren zur adaptiven I/Q-Asymmetriekompensation

Vorrichtung, umfassend: einen begrenzten Impulsantwort-(FIR-)Filter (320), der dazu konfiguriert ist, eine Version eines Digitalsignals zu filtern, um ein Kompensationssignal zu erzeugen, wobei der FIR-Filter X Realzahlfilterkoeffizienten und Y Imaginärzahlkoeffizienten aufweist, wobei Y kleiner als X ist, wobei das Digitalsignal eine digitale Darstellung einer demodulierten gleichphasigen und einer Quadraturphasenkomponente eines Funkfrequenz-(RF-)Signals umfasst; und einen Addierer (330), der dazu konfiguriert ist, eine andere Version des Digitalsignals und eine Version des Kompensationssignals zu summieren, um ein symmetrisches Digitalsignal zu erzeugen, das im Verhältnis zum Digitalsignal ein verbessertes Spiegelfrequenz-Unterdrückungsverhältnis aufweist; wobei eine der Versionen des Digitalsignals oder die Version des Kompensationssignals jeweils eine komplexe Konjugierte des Digitalsignals oder eine komplexe Konjugierte des Kompensationssignals ist, und die übrigen nicht die komplexen ...

System zur Geisterbildkompensierung in Videosignalen

Programmierbare Schaltung zur Realisierung eines digitalen Filters

Eine Temperatur- und Spannungskompenzationsschaltung

Computing least mean square algorithm for adaptive filter

The method involves using an N-tap filter and a modified least-mean-square algorithm, which is computed in a first computational step with a first, wide bit length in the N taps. In a second step a lower number of bits is selected for the further steps of the process, depending on the results of the first step.

Digitales adaptierbares Filter zur besseren Messgenauigkeit in einem elektronischen Instrument

Method and apparatus for reduction of unwanted feedback

Image data storage

An image data value storage system is described comprising a plurality of tiled memories 24 which are fed with a single set of read addresses from a read address generator 34. The data stored within each of the tiled memories is offset relative to each other by the action of a delay unit 46 upon writing of the data into the memory units. Thus, a single set of read addresses accesses a larger contiguous array of image data values 10 than is individually addressed within each of the memory units. ...

Method and apparatus for updating precoder coefficients in a data communication transmitter

In a data communication system having a channel with time-varying impairments and further having a transmitter with a precoder, a method and apparatus for updating the precoder coefficients. The method and apparatus is based on the arrangement of a duplicate precoder and correction filter in the receiver, where the correction filter generates incremental coefficients. The incremental coefficients are sent via a back channel to the transmitter and added to the current values of the precoder coefficients. The method and apparatus provide DFE-like performance and do not have the delay problems associated with DFEs.

Method and apparatus for attenuating an unwanted signal in a mix of signals

Apparatus for attenuating an unwanted signal, eg. a speech signal, in a mix of signals, eg. a broadcast audio signal, includes an adaptive cancellation circuit 30 having an adaptive transversal filter 32 controlled in accordance with the circuit output. The apparatus is provided with a delay measurement system 10 for measuring the delay between the unwanted and mixed signals and for controlling the appropriate one of two variable delays 21 and 22 so as to approximately co-time the unwanted signal and the contribution of the unwanted signal which is in the mixed signal. The adaptive filter can thus be made significantly smaller. ...

Method and apparatus for attenuating an unwanted signal in a mix of signals

RADAR RECEIVER FOR A RADAR HAVING A DIGITAL BEAM FORMING ANTENNA

Method for channel estimation using individual adaptation

The process of the present invention computes the update (403) to the least mean square channel estimator taps. The update coefficient for each tap is computed independently based on the power of each tap (402) relative to the main tap power. ...

Method for distributing a digital audio signal

A method for distributing a digital audio signal comprises transmitting timing information in a continuous channel (`the timing channel') that is synchronous to an audio clock (master clock) at a source. The timing channel includes information for both clock synchronization and sample synchronization. Audio sample data is transmitted in a separate channel (the data channel) that is asynchronous to the timing channel. The data channel may be optimized for data related parameters, such as bandwidth and robustness. The timing channel may be optimized for minimum clock jitter or errors in clock timing. The method is suitable for use in distributing stereo or surround sound signals to loudspeakers. A slave device receiving the timing channel may be equipped with a low bandwidth filter to filter out high frequency jitter, so that the jitter of the recovered slave clock is of the same order as the jitter in the master clock.

Method for distributing a digital audio signal

Ambient noise reduction system

Improvements in or relating to adaptive filters

Loop gain normalization is employed in adaptive filters to control weighting of the filter characteristic updates in order to converge properly to a desired filter characteristic. Filter stability and rapid high quality convergence is realized for a variety of received or inputted signals by employing both long term and fast attack estimates of a prescribed input signal characteristic to normalize the update gain. In one embodiment, both long term and fast attack input signal power estimates are generated and one of the two estimate values is selected to normalize the update gain. Specifically, the fast attack estimate is modified by a predetermined value and, then, the larger of the long term estimate and modified fast attack estimate is selected to normalize the update gain.

ADAPTIVE FILTER

AN ADAPTIVE FIR FILTER HAVING RESTRICTED COEFFICIENT RANGES

Adaptive filters and equalisers

PROCEDURE AND DEVICE FOR THE BLIND IDENTIFICATION OF FOURTH ORDER OF A UNTERBESTIMMTEN SOURCE MIXTURE

BASIC FREQUENCY EXTRACTION WITH ADAPTIVE FILTER

SELECTIVE ACTIVE SYSTEM FOR THE SUPPRESSION OF REPEATING FEATURES

DIGITAL FILTERS.

ADAPTIVE ALGORITHM WITH VARIABLE BLOCK LENGTH FOR INTOXICATION-ROBUST ACOUSTIC ECHO COMPENSATION

DIGITALER VIRTUELLER ERDE-ALGORITHMUS UNTER VERWENDUNG VON MEHRFACHEN INTERAKTIONEN

ADAPTIVE CONDITION SPACE SIGNAL SEPARATION, UNTERSCHEIDUNGSUND R�CKGEWINNUNGSARCHITEKTUREN AND YOUR ADJUSTMENTS FOR USE IN DYNAMIC ENVIRONMENTS

ACTIVE DIGITAL ANNULATIONSSYSTEM ßVIRTUELLE ERDEß

CIRCUIT FOR CONTROLLING TAP GAINS AT RAPID SPEED IN AN ADAPTIVE FILTER

ELECTRONIC FILTERS, HEARING AIDS AND METHODS

BLIND SIGNAL SEPARATION

ADAPTIVE TRANSVERSAL FILTER

ADAPTIVE FILTER INCLUDING CONTROLLED TAP GAIN COEFFICIENT DRIFT

Adaptive filters and equalisers

Fast reacquisition catv downstream adaptive equalizer

MSE VARIABLE STEP ADAPTIVE FILTER

ADVANCED PERIODIC SIGNAL ENHANCEMENT

An enhancement system improves the perceptual quality of a processed speech. The system includes a delay unit that delays a signal received through a discrete input. A spectral modifier linked to the delay unit is programmed to substantially flatten the spectral character of a background noise. An adaptive filter linked to the spectral modifier adapts filter characteristics to match a response of a non-delayed signal. A programmable filter is linked to the delay unit. The programmable filter has a transfer function functionally related to a transfer function of the adaptive filter.

ADAPTIVE FILTER UPDATE GAIN NORMALIZATION

ADAPTIVE FILTER UPDATE GAIN NORMALIZATION Update gain normalization is employed in adaptive filters to control weighting of the filter impulse response updates in order to converge properly to a desired impulse response. Singing, i.e., oscillating, of the filter is overcome by adjusting the update gain when an incoming signal power estimate used to normalize the gain exceeds a prescribed threshold value. In one example, the normalized gain is adjusted to be a fixed value for power estimate values which exceed the threshold. In accordance with another aspect of the invention, a single normalized gain value is used to adjust the update gain in two adaptive filters employed as echo cancelers in a bidirectional voice frequency repeater.

DIGITAL FILTERS

A transversal finite impulse response, adaptive digital filter comprises three stages, one of which is a memory stage which stores input samples X and coefficients, another of which is an arithmetic stage which forms convolution products and accumulates the product to provide an output, and the third of which forms update values for updating the coefficients according to a predetermined algorithm. The stages are coupled by buses and data transfer between the stages via the buses is controlled by a control unit. The filter operates so that convolution products are formed during a first part of each sample period and update values are formed during a second part of each sample period for use in the subsequent sample period.

INVERSE CONTROL SYSTEM

... ! An inverse control system is disclosed, which comprises FIR filters provided between transmitting elements at n (n = 2, 3, ...) input points of a linear FIR system and a common signal source, for an inverse control such as to provide desired impulse responses between the signal source and m (n > m) output points of the linear FIR system. A j-th (j = 1, 2, ..., n) one of the FIR filters has a number Lj of taps which satisfies the relationships represented by for all i = 1, 2, ..., m and j = 1, 2, ..., n where wij is the number of discrete signals representing the impulse response gij(k) between the j-th output point and i-output point and Pi is the number of discrete signals representing the desired impulse response ri(k) between the signal source and i-th output point. The j-th FIR filters has a filter coefficient hj(k) satisfying the relationship for all i = 1, 2, ..., m where ? is a discrete convolution.

NOISE CANCELING APPARATUS

... 66446-414 In order to eliminate background noise infiltrating into an input audio signal, a noise-canceling apparatus includes: first and second acoustic pickups for primarily picking up the audio signal and noise, respectively; a filter receiving the output of the second acoustic pickup; a subtracter for subtracting the output of the filter, when the audio signal is silent, from the output of the first acoustic pickup; and a coefficient determination for determining coefficients of the filter so as to make the filter generate a noise signal corresponding to a signal generated by passing the noise through a transmission path having a transmission frequency characteristic from the noise source to the first acoustic pickup. The coefficients of the filter are determined on the basis of mutual-correlation coefficients between the outputs of the first and second pickups and self-correlation coefficients of the output of the second pickup, thereby improving the processing efficiency.

ADAPTIVE FILTERING SYSTEM

TRANSMISSION SYSTEM COMPRISING AT LEAST A CODER

A transmission system is described comprising at least a coder (101) for coding a signal (10a). The system furthermore comprises at least a transmitter (102), at least a receiver (103) and at least a decoder (104). At least for a coder (101) is at least provided an adaptive prediction filter (10A, 10B) by means of which speech or audio signals are coded while their bit rates are reduced. The filter coefficients for an adaptive prediction filter (10A, 10B) are recursively determined accordin g to the Levinson-Durbin Recursion in fixed point arithmetic.

GHOST CANCELLING SYSTEM AND CONTROL METHOD THEREOF

A ghost cancelling system which distinguishes whether ghosts have been substantially cancelled from a video signal for finding an equilibrium condition. The system calculates a specified value which depends on the noise signal contained in the video signal. The tap coefficients which are equal or greater than the specified value are then detected, and the system corrects only those tap coefficients after the equilibrium condition exists.

Adaptively Controlled Filter

MULTIPORTED REGISTER FILE FOR COEFFICIENT USE IN FILTERS

Multiported register files for use in storing coefficients in adaptive FIR filte rs. incorporate computational ability, e.g., the ability to perform computation on c oefficient values or derivatives thereof, or to control the operations performed thereon. F or example, a multiported register file may incorporate an overflow/underflow detec tion and/or saturation unit. Also, the multiported register file may incorporate a sp ecial encoder to speed up the multiplication process, e.g., the so-called "Booth" enco der. Likewise, the multiported register file may incorporate a converter for changing the representation of the coefficients, e.g., a two's complement to sign-magnitude c onverter. All computation performed in the multiported register file is performed outside of the critical path of the filtering or of the coefficient updating. Using such improv ed multiported register files, adaptive FIR filters can be constructed which operat e faster, and with lower power consumption.

SIGNAL PROCESSOR FOR REDUCING UNDESIRABLE SIGNAL CONTENT

A signal processor for reducing undesirable signal content reduces the undesirable signal content by exaggerating the undesirable signal content and then using this exaggerated undesirable signal and adaptive filter means to estimate the undesirable content in the signal and then substantially removing it from the signal. The signal processor includes a signal mapping means for exaggerating the undesirable signal content; and an adapti ve filter means for reducing the undesirable signal content using the exaggerated undesirabl e signal content.

METHODS AND APPARATUS FOR BLIND SIGNAL SEPARATION

A set of generalized architectures, frameworks, algorithms, and devices for separating, discriminating, and recovering original signal sources by processing a set of received mixtures and functions of said signals based on processing of the received, measured, recorded or otherwise stored signals or functions thereof. There are multiple criteria that can be used alone or in conjunction with other criteria for achieving the separation and recovery of the original signal content from the signal mixtures. The system of the invention enables the adaptive blind separation and recovery of several unknown signals mixed together in changing interference environments with very minimal assumption on the original signals. The system of this invention has practical applications to non-multiplexed media sharing, adaptive interferer rejection, acoustic sensors, acoustic diagnostics, medical diagnostics and instrumentation, speech, voice, language recognition and processing, wired and wireless modulated ...

SCHALTUNGSANORDNUNG IN EINER UEBERTRAGUNGSANLAGE ZUR AUTOMATISCHEN ENTZERRUNG.

Compensating noise of constant envelope curve transmission signal subject to fading - using Viterbi detector to provide estimated symbols used by coefft. processor to provide transversal filter coeffts.

The noise compensation method involves the use of a Kalman filter for estimation of the noise. A viterbi detector (3) detects the estimated symbols using a max. probability principle. A coefft. processor (6) uses the estimated symbols and the demodulated signal to determine the long term static noise characteristic for calculating the filter characteristics of transversal filters in the path of the received signal. The received signal is fed to a delay element (5) in front of the coefft. processor (6), the latter using the long term static characteristic of the delayed signal and the reception signal multiplied with the estimated symbols. USE/ADVANTAGE - Optimum compensation of fading including a priori unknown fading spectrum.

Switchable of not dezimierendes/dezimierendes adaptive Entzerrerfilter.

Folding sequence adaptive equalizer

Linear filter equalizer

FILTERS HAS UNILATERAL SUB-BANDS

Dans un procédé pour traiter un signal d'entrée le signal d'entrée est décomposé en une pluralité de sous-bandes à l'aide d'un banc de filtres à sous-bandes unilatérales à valeurs complexes. Les spectres de fréquence unilatéraux des sous-bandes résultantes rendent négligeable le repliement aux taux voisins de deux fois le taux de sous-échantillonnage critique.

Method for calculating weighting coefficients for a digitizer analytical for signal processing, and digitizer analytical.

DEVICE OF LOCALIZATION Of a POINT OF REFLEXION OF SIGNAL ON a LINE OF TRANSMISSION

SYSTEM OF SEPARATION OF NONSTATIONARY SOURCES

Le système (10) de séparation de sources traite des signaux d'entrée formés par des mélanges de signaux primaires issus des sources et estime les signaux primaires de manière à ce que les estimations ne divergent pas même en présence de signaux d'entrée non stationnaires.Le système comprend un premier sous-ensemble (12) de séparation qui délivre des premières estimations des signaux primaires, un second sous-ensemble (14) qui détermine adaptativement les coefficients de séparation, un troisième sousensemble (15) qui normalise les premières estimations et délivre des premières estimations normalisées servant au calcul des coefficients de séparation.Un module de sortie (17) permet de délivrer des estimations ayant entre elles un même rapport de proportionnalité que celui existant entre les signaux primaires. Un module de sélection (19) évite à une estimation d'être dupliquée sur plusieurs sorties dans le cas où certains signaux primaires sont absents ou très faibles.Référence : figure 8Applications ...

DEVICE OF LOCALIZATION Of a POINT OF REFLEXION OF SIGNAL ON a LINE OF TRANSMISSION

SELF ADAPTIVE FILTER AND CONTROL CIRCUIT THEREFOR

Récepteur radar, notamment pour radar ayant une antenne à formation de faisceau par le calcul

Ce récepteur radar, notamment pour radar ayant une antenne à formation de faisceau par le calcul, comporte essentiellement des moyens de filtrage adapté 7, 11 ayant une bande passante B, des moyens 12 de détection d'amplitude et de phase, des moyens 8 de codage analogique-numérique, et est tel que les moyens 8 de codage analogique-numérique opèrent sur les signaux à fréquence intermédiaire, à une fréquence très supérieure à la fréquence intermédiaire. Il peut être réalisé entièrement en technologie Arséniure de Gallium.

오디오 신호 처리 방법 및 장치

... 본 발명은 오디오 신호를 효과적으로 재생하기 위한 신호 처리 방법 및 장치에 관한 것으로서, 더욱 상세하게는 멀티채널 혹은 멀티오브젝트 오디오 신호를 스테레오로 재생하기 위한 바이노럴 렌더링을 낮은 연산량으로 구현하기 위한 오디오 신호 처리 방법 및 장치에 관한 것이다. 이를 위해 본 발명은, 멀티채널 또는 멀티오브젝트 신호를 포함하는 멀티 오디오 신호를 수신하는 단계; 상기 멀티 오디오 신호의 필터링을 위한 절단된 서브밴드 필터 계수들을 수신하는 단계, 상기 절단된 서브밴드 필터 계수는 상기 멀티 오디오 신호의 바이노럴 필터링을 위한 BRIR(Binaural Room Impulse Response) 필터 계수로부터 획득된 서브밴드 필터 계수의 적어도 일 부분이며, 상기 절단된 서브밴드 필터 계수의 길이는 해당 서브밴드 필터 계수에서 추출된 특성 정보를 적어도 부분적으로 이용하여 획득된 필터 오더 정보에 기초하여 결정됨; 및 상기 멀티 오디오 신호의 각 서브밴드 신호에 대응하는 상기 절단된 서브밴드 필터 계수를 이용하여 상기 서브밴드 신호를 필터링 하는 단계; 를 포함하는 것을 특징으로 하는 오디오 신호 처리 방법 및 이를 이용한 오디오 신호 처리 장치를 제공한다.

system for determined application of and Method thereof

GAUGING CONVERGENCE OF ADAPTIVE FILTERS

DEVICE AND PROCESS FOR FILTERING OF AN ENTRANCE SIGNAL

SYSTEM OF FILTER ADAPTATIVO

ADAPTIVE DIGITAL FILTER, SIGNAL PROCESSING METHOD, FM RECEIVER, AND PROGRAM

A complex signal having one signal in a real part and the other signal in an imaginary part, the two signal being generated from one real signal and having phases shifted by 90 degrees from each other, is inputted into an input terminal (301). By a convolution calculation of the input signal and the real signal filter coefficient, a filter unit generates an output signal of a complex signal and outputs it to an output terminal (302). A coefficient control unit is formed by a common section (318) and separate sections (3190 to 319N-1) controls an envelope target value according to the input signal and updates the filter coefficient so that the envelope value obtained from the output signal approaches the envelope target value.

ADAPTIVE COEFFICIENT SIGNAL GENERATOR FOR ADAPTIVE SIGNAL EQUALIZERS WITH FRACTIONALLY-SPACED FEEDBACK

An adaptive coefficient signal generator for use in an adaptive signal equalizer with fractionally-spaced feedback. The signals representing the feedback tap coefficients are generated in conjunction with a timing interpolation parameter such that the fractionally-spaced feedback circuitry dynamically emulates symbol-spaced feedback circuitry.

ADAPTIVE TECHNIQUES FOR AUTOMATIC FREQUENCY DETERMINATION AND MEASUREMENT

Various implementations of a technique are presented for capitalizing on signal correlation in a manner for providing usable frequency information. In a specific embodiment, a filter (51) is adapted to a signal component of an input signal using a variable generated by an adapter (58). The variable has a prescribed relationship to the frequency of the signal component. The variable is used to weight a combination of signal values derived from past signal history to form a prediction of a future signal. An adder (56) combines the new signal value with the prediction to form an error signal which is used by the filter (51) and the adapter (58). When the filter converges, a frequency indicator (59) using the variable provides an output signal indicative of the frequency of the signal component.

SIGNAL SEPARATION METHOD AND DEVICE FOR NON-LINEAR MIXING OF UNKNOWN SIGNALS

Selon l'invention sont déterminés des paramètres d'un système technique avec lequel des signaux de sortie constitués d'une pluralité de signaux d'entrée superposés, statistiquement indépendants les uns des autres peuvent être déterminés. Les paramètres sont déterminés de telle sorte que l'indépendance statistique des signaux de sortie est maximisée.

Parallel decimator adaptive filter and method for all-rate gigabit-per-second modems

Parallel adaptive filters and filtering methods that enable processing of an input signal in a circuit that has an clock speed many times slower than the input rate of the input signal that is processed. The present invention extends the use of a polyphase decimator structure to processes a data stream requiring a low pass filtered bandlimited (low-rate) output that is used for high-rate output structures. The filters and methods break an input data stream into parallel paths that efficiently produce a bandlimited (decimated, low-rate) filtered output. Each of the parallel paths is processed at a decimated rate to provide a filtered output signals corresponding to a filtered version of the input signal ...

Adaptive radio transceiver with a local oscillator

An exemplary embodiment of the present invention described and shown in the specification and drawings is a transceiver with a receiver, a transmitter, a local oscillator (LO) generator, a controller, and a self-testing unit. All of these components can be packaged for integration into a single IC including components such as filters and inductors. The controller for adaptive programming and calibration of the receiver, transmitter and LO generator. The self-testing unit generates is used to determine the gain, frequency characteristics, selectivity, noise floor, and distortion behavior of the receiver, transmitter and LO generator. It is emphasized that this abstract is provided to comply with the rules requiring an abstract which will allow a searcher or other reader to quickly ascertain the subject matter of the technical disclosure. It is submitted with the understanding that it will not be used to interpret or limit the scope or the meaning of the claims.

Adaptive canceller filter module

An adaptive canceller filter module having signal sensors and signal filters in a circuit for use with filtered-x algorithms to adapt the coefficients of one of said filters to minimize the measure of the error.

Electronic filters, signal conversion apparatus, hearing aids and methods

An electronic filter for filtering an electrical signal. Signal processing circuitry therein includes a logarithmic filter having a series of filter stages with inputs and outputs in cascade and respective circuits as GOVERNMENT SUPPORT This invention was made with U.S. Government support under Veterans Administration Contract VA KV 674P857 and National Aeronautics and Space Administration (NASA) Research Grant No. NAG10-0040. The U.S. Government has certain rights in this invention.

Vectorial adaptive filtering apparatus with convergence rate independent of signal parameters

An adaptive signal processor, used for discriminating between a desired signal and several undesired signals, includes an arrangement of sensors, which samples the signals and forms a set of parallel electrical outputs each of which is proportional to a different linear combination of the desired and interfering signals, a set of complex weights, each of which multiplies a corresponding one of the electrical outputs thus forming a set of weighted signals, a coherent summing junction which coherently sums the weighted signals, thus producing a final output of the processor and a set of improved control loops which sample the parallel electrical outputs and the final output and generate from them a set of control signals to control the complex weights so as to drive them to optimal, steady states, and thereby to suppress the interfering signals and to preserve the desired signal at the final output of the processor. The control loops generate the control signals for the weights in such a ...

Adaptive filter and learning method thereof

The present invention provides an adaptive filter and a learning method therefor to eliminate real output signals generated by input signals and an environment signal response. The adaptive filter receives the input signals from an input signal source and generates estimated output signals according to an filter coefficients and the input signals. The first step of the learning method is storing the input signals and the real output signals within a past time period. Then, a predictive input signal and a predictive output signal are generated according to the input signals and the real output signals in the memories, respectively. Finally, the filter coefficients is updated according to the predictive input signal, the predictive output signal, one of the input, signals and one of the real output signals, causing the estimated signals to approximate the real output signals.

Adaptive equalizer and designing method thereof

Following arrangement of an adaptive equalizer with a direct filter structure according to the least mean square error architecture, look ahead conversion of modifying a tap coefficient of the next cycle utilizing the tap coefficient of a predetermined preceding cycle is carried out and then a retiming process of adjusting the timing of tap coefficients and signals is carried out to arrange delay elements, whereby a transposition filter is realized. A high-speed adaptive equalizer is provided that can have the critical path reduced without increasing the hardware amount and that is superior in expansionability.

Adaptive equalization with group delay

Methods, apparatuses, and systems are presented for performing adaptive equalization involving receiving a signal originating from a channel associated with inter-symbol interference, filtering the signal using a filter having a plurality of adjustable tap weights to produce a filtered signal, and adaptively updating each of the plurality of adjustable tap weights to a new value to reduce effects of inter-symbol interference, wherein each of the plurality of adjustable tap weights is adaptively updated to take into account a constraint relating to a measure of error in the filtered signal and a constraint relating to group delay associated with the filter. Each of the plurality of adjustable tap weights may be adaptively updated to drive group delay associated with the filter toward a target group delay.

Variable block size adaptation algorithm for noise-robust acoustic echo cancellation

An apparatus that automatically adjusts the adaptation block size for a least-mean square (LMS) adaptive filter depending on the input signal-to-noise ratio (SNR) is disclosed. The apparatus monitors the instantaneous SNR and continually adjusts the block size to provide high noise immunity, thereby increasing the convergence speed of the filter and decreasing the asymptotic mean-square error. An exemplary embodiment of the present invention is presented in the context of acoustic echo cancellation, though it is noted that the adaptive filter of the present invention is useful in any environment in which the noise characteristics are subject to change.

Adaptive filter

An adaptive filter and method that proportionately adjusts individual tap gain distributors such that the individually adjusted tap gains are not necessarily equal to one another and that the average of tap gains remains substantially constant. The filter employs a proportionate normalized least-means-squares (PNLMS) method of adaptation that imparts improved convergence speed over prior art adaptive filters that utilize a normalized least-means-squares (NLMS) method of adaptation, without affecting the adaptation quality of the filter.

DECISION FEEDBACK EQUALIZER WITH DYNAMIC FEEDBACK CONTROL

A decision feedback equalizer with dynamic feedback control for use in an adaptive signal equalizer. Timing within the decision feedback loop is dynamically controlled to optimize recovery of the data signal by the output signal slicer. The dynamic timing is controlled by a signal formed as a combination of feedback and feedforward signals. The feedback signal is an error signal related to a difference between pre-slicer and post-slicer signals. The feedforward signal is formed by differentiating and delaying the incoming data signal.

System and method for signal limiting

Methods and systems for processing a signal with a corresponding noise profile are disclosed. Aspects of the method may comprise analyzing spectral content of the noise profile. At least one noise harmonic within the signal may be filtered based on said analyzed spectral content. The filtered signal may be limited. The noise profile may comprise a phase noise profile. The signal may comprise at least one of a sinusoidal signal and a noise signal. At least one filter coefficient that is used to filter said at least one noise harmonic may be determined. The filtering may comprise low pass filtering and the limiting may comprise hard-limiting the filtered signal. The signal may be modulated prior to the filtering. The signal may be downconverted prior to the modulating. At least one signal component of the signal may be downconverted.

Adaptive filter pitch extraction

An enhancement system extracts pitch from a processed speech signal. The system estimates the pitch of voiced speech by deriving filter coefficients of an adaptive filter and using the obtained filter coefficients to derive pitch. The pitch estimation may be enhanced by using various techniques to condition the input speech signal, such as spectral modification of the background noise and the speech signal, and/or reduction of the tonal noise from the speech signal.

Short Link Efficient Interconnect Circuitry

Systems and methods for electronic devices including two or more semiconductor devices coupled via an interconnect. The interconnect includes multiple lanes each having a link between the first and second semiconductor devices. One or more lanes of the multiple lanes each include clock and data recovery circuitry to perform full clock and data recovery. One or more other lanes of the multiple lanes each do not include clock and data recovery circuitry and instead includes a phase adjustment and clock multiplier circuit that is slave to clock and data recovery circuitry of the one or more lanes.

AMBIENT NOISE REDUCTION SYSTEM

An adaptive, feed-forward, ambient noise-reduction system includes a reference microphone for generating first electrical signals representing incoming ambient noise, and a connection path including a circuit for inverting these signals and applying them to a loudspeaker directed into the ear of a user. The system also includes an error microphone for generating second electrical signals representative of sound (including that generated by the loudspeaker in response to the inverted first electrical signals) approaching the user's ear. An adaptive electronic filter is provided in the connection path, together with a controller for automatically adjusting one or more characteristics of the filter in response to the first and second electrical signals. The system is configured to constrain the operation of the adaptive filter such that it always conforms to one of a predetermined family of filter responses, thereby restricting the filter to operation within a predetermined and limited set ...

Method and apparatus for digital up-down conversion using infinite impulse response filter

A method and an apparatus for digital up-down conversion using an Infinite Impulse Response (IIR) filter are provided. The method for digital up-down conversion for frequency conversion in a mobile communication system using plural frequency converts, includes IIR-filtering, by a magnitude response IIR filter having the same magnitude response as in Finite Impulse Response (FIR) filtering, an input signal and a stable filter coefficient calculated according to a Levinson polynomial; and receiving, by the magnitude response IIR filter, the IIR filtered signal, and performing IIR filtering by a phase compensation IIR filter having a filter coefficient compensating for a non-linear phase to a linear phase.

Method and apparatus for reducing aliasing in cascaded filter banks

A method and apparatus are disclosed for reducing aliasing between neighboring subbands in cascaded filter banks. An alias reduction filter bank is included to reduce the aliasing components between different subbands. Generally, the magnitude response and phase of the alias reduction filter bank is similar to the magnitude response of the synthesis filter bank of the first stage filter bank. The alias reduction filter bank filters and adds the signals from a set of M2 subbands from the M1 subbands of the first stage analysis filter bank. A higher frequency resolution is obtained after the alias reduction stage by a following analysis filter bank. The signals of these subbands are first fed into an alias reduction filter bank to reduce the aliasing. If the first stage filter bank is a modulated uniform filter bank with M1 bands, and the stage for alias reduction has M2 bands, then to obtain alias cancellation, the alias reduction filter bank has to have a similar frequency response as the ...

Optimal covariance filtering

A multirate optimal filtering technique allows for higher sampling rates with the same processing time throughput or greater throughput for a fixed sampling rate. in many instances, the measurement sampling rate dictated by the system dynamics is much higher than that required for time varying gain considerations. Thus, the optimal gain remains fairly constant for several consecutive measurements. Under such circumstances, the covariance and optimal gain computations can be processed at a slower rate than the filter measurement computations when the optimal gains are appropriately adjusted for use by the higher processing rate measurement equations. This increases the processing throughput for any given sampling rate or allows increased measurement sampling for a fixed throughput time.

Method of low pass filtering and arrangement to implement the method

The invention relates to a method of low pass filtering a signal and an arrangement to implement the method. An improvement in signal smoothing without simultaneously rounding the edges of the useful signal can be achieved by providing for the signal an adaptive filtering arrangement by means of which the averaging interval is determined as a function of an edge steepness of the signal. ...

Scaled signal processing elements for reduced filter tap noise

An adaptive transversal filter having tap weights Wj which are products of corresponding tap coefficients C j and tap gains M j is provided. A filter control loop controls all of the tap coefficients C j such that an error signal derived from the filter output is minimized. One or more tap control loops controls a tap gain M k such that the corresponding tap coefficient C k satisfies a predetermined control condition. For example, |C k | can be maximized subject to a constraint |C k | C max , where C max is a predetermined maximum coefficient value. In this manner, the effect of quantization noise on the coefficients C j can be reduced. Multiple tap control loops can be employed, one for each tap. Alternatively, a single tap control loop can be used to control multiple taps by time interleaving.

FOLDING SEQUENTIAL ADAPTIVE EQUALIZER

A folding adaptive equalizer is provided. The equalizer comprises an equalizer core and an automatic gain control loop. The equalizing transfer function of the equalizer core is modulated by one or more gain control signals generated by the automatic gain control loop and by a folding signal generated by the automatic gain control loop. When the folding signal is inactive, an increase in the gain control signals produces an increase in the high-frequency, high-bandwidth gain of the transfer function of the equalizer core. When the folding signal is active, further gain can be applied by decreasing the gain control signals, which produces a frequency-shift in the transfer function of the equalizer core toward lower bandwidth and an increase in the high-frequency, low-bandwidth gain of the transfer function of the equalizer core. 1. An equalizer , comprising:a gain control loop that produces a folding signal and at least one gain control signal; andan equalizer core coupled to an input signal from a transmission medium that applies a high-bandwidth transfer function and a low-bandwidth transfer function to the input signal to produce an output signal,wherein:the high-bandwidth transfer function has high-bandwidth gain proportional to the at least one gain control signal;when the folding signal is at a first level, the low-bandwidth transfer function has no gain; andwhen the folding signal is at a second level, the low-bandwidth transfer function has low-bandwidth gain inversely proportional to the at least one gain control signal.2. The equalizer of claim 1 , wherein each transfer function has frequency-dependent gain.3. The equalizer of claim 1 , wherein the total gain of the transfer functions applied by the equalizer core approximates the inverse of the losses incurred in the transmission of the input signal through the transmission medium.4. The equalizer of claim 1 , wherein the folding signal is at a first level when the amount of low-frequency gain required to ...

DYNAMIC BIAS CONTROL

Systems and methods for controlling a power amplifier includes combining a digital modulated data signal with a digital bias signal to generate a combined digital signal, the digital bias signal generated based on an envelope for the modulated data signal; converting, by a digital-to-analog converter, the combined digital signal into a combined analog signal, the combined analog signal comprising an analog modulated data signal and an analog envelope bias signal; and separating the analog modulated data signal and the analog bias signal onto separate signal paths, wherein the converting is performed using a single digital-to-analog converter. 1. A communication system , comprising:a digital tuner operable to receive a radio frequency (RF) signal and operable to output a digital intermediate frequency (IF) signal;a digital combiner operable to receive the digital IF signal and operable to output a biased IF signal;a DAC operable to convert the biased IF signal into an analog IF signal;a power amplifier having a signal input, a control input and an output;a high pass filter operable to high pass filter the analog IF signal, the high pass filtered analog IF signal being coupled to the signal input of the power amplifier; anda low pass filter operable to low pass filter the analog IF signal, the low pass filtered analog IF signal being coupled to the control input of the power amplifier.2. The communication system of claim 1 , wherein the high pass filter is configured to filter out the bias signal from the analog IF signal.3. The communication system of claim 1 , wherein the control input of the power amplifier is operable to control a bias of the power amplifier.4. The communication system of claim 1 , wherein the digital IF signal comprises a data signal modulated on a carrier.5. The communication system of claim 1 , wherein the DAC comprises the digital combiner.6. The communication system of claim 1 , wherein claim 1 , an envelope computation module is configured ...

METHOD AND APPARATUS FOR PROCESSING MULTIMEDIA SIGNALS

The present invention relates to a method and an apparatus for processing a signal, which are used for effectively reproducing a multimedia signal, and more particularly, to a method and an apparatus for processing a signal, which are used for implementing filtering for multimedia signal having a plurality of subbands with a low calculation amount. 1. (canceled)2. A method for processing a multimedia signal , the method comprising:receiving a multimedia signal;receiving a set of filter coefficients for each subband, wherein the set of filter coefficients is truncated frequency-dependently from a set of proto-type subband filter coefficients based on a filter order for a corresponding subband, and wherein the filter order determines a length of the set of filter coefficients for each subband and is determined to be variable in a frequency domain; andfiltering each subband signal of the multimedia signal by using the set of filter coefficients corresponding thereto.3. The method of claim 2 , wherein the filter order is determined based at least part on energy decay time information of the corresponding subband.4. The method of claim 3 , wherein the energy decay time information is obtained from one or more sets of proto-type subband filter coefficients for the corresponding subband.5. The method of claim 3 , wherein the multimedia signal includes an audio signal claim 3 , andwherein the set of proto-type subband filter coefficients is a set of binaural room impulse response (BRIR) filter coefficients in the frequency domain.6. The method of claim 5 , wherein the energy decay time information includes reverberation time information.7. The method of claim 2 , wherein the filter order has a single value for each subband.8. An apparatus for processing a multimedia signal claim 2 , the apparatus comprising: receive a multimedia signal;', 'receive a set of filter coefficients for each subband, wherein the set of filter coefficients is truncated frequency-dependently from a ...

Adaptive filter bank for modeling a thermal system

Embodiments of the disclosure implement an application of an adaptive filter bank that is used to characterize the heat transfer of a volume in a thermal system, to estimate temperature and power consumption, and to improve performance characteristics in applications including optimal temperature control and diagnostics. In some embodiments, the adaptive filter bank is an iterative solution, comprised of a collection of adaptive filters defined to consume incident signals, produce an aggregate reference signal, estimate an error relative to an observed primary signal, and modify thermal coefficients to converge on a solution. For example, the incident signals are comprised of properties related to active, passive, solar irradiance, and unobserved heat transfer. A reference signal is an estimate of a primary signal, related to the rate of heat transfer or temperature change. Thereupon, the thermal coefficients are modified in an adaptive process to include gradient descent, which minimizes estimation error.

CARDIAC ELECTRICAL SIGNAL GROSS MORPHOLOGY-BASED NOISE DETECTION FOR REJECTION OF VENTRICULAR TACHYARRHYTHMIA DETECTION

A medical device system, such as an extra-cardiovascular implantable cardioverter defibrillator ICD, senses R-waves from a first cardiac electrical signal by a first sensing channel and stores a time segment of a second cardiac electrical signal in response to each sensed R-wave. The medical device system determines a morphology parameter correlated to signal noise from time segments of the second cardiac electrical signal, detects a noisy signal segment based on the signal morphology parameter; and withholds detection of a tachyarrhythmia episode in response to detecting a threshold number of noisy signal segments. 1. A medical device system comprising: a first sensing channel configured to receive a first cardiac electrical signal via a first sensing electrode vector coupled to the medical device system and to sense R-waves in response to crossings of a sensing amplitude threshold by the first cardiac electrical signal, and', 'a second sensing channel configured to receive a second cardiac electrical signal via a second sensing electrode vector coupled to the medical device system and different than the first sensing electrode vector;, 'a sensing circuit comprisinga memory; and store a time segment of the second cardiac electrical signal in the memory for each of the plurality of R-waves sensed by the first sensing channel;', determine a morphology parameter correlated to signal noise from the stored time segment; and', 'detect the stored time segment as being a noisy signal segment based on the morphology parameter determined for the respective stored time segment; and, 'for each of a plurality of the stored time segments,'}, 'withhold detection of a tachyarrhythmia episode in response to detecting at least a threshold number of the stored time segments as noisy signal segments., 'a control circuit coupled to the sensing circuit and the memory, the control circuit configured to2. The system of claim 1 , further comprising a notch filter claim 1 , wherein the time ...

COMPACT MODEL NONLINEAR COMPENSATION OF BANDLIMITED RECEIVER SYSTEMS

A nonlinear compensator is provided to include a decomposition circuit and a plurality of filter elements. The decomposition circuit has a nonlinear frequency response characteristic and the decomposition circuit is configured to receive an input signal and decompose the input signal into decomposed signals corresponding to positive and negative frequency signal components of the input signal. Each of the plurality of filter elements is configured to receive at least portions of the decomposed signals and apply filter element characteristics to the decomposed signals with the filter element characteristics that are matched to the nonlinear frequency response of the decomposition circuit. 1. A nonlinear compensator comprisinga decomposition circuit having a nonlinear frequency response characteristic, the decomposition circuit being configured to receive an input signal and decompose the input signal provided thereto into decomposed signals corresponding to positive and negative frequency signal components of the input signal; anda plurality of filter elements each of which is configured to receive at least portions of the decomposed signals and apply filter element characteristics to the decomposed signals with the filter element characteristics being matched to the nonlinear frequency response of the decomposition circuit.2. The nonlinear compensator of claim 1 , wherein the plurality of filter elements is configured to reduce the number of tones to characterize the nonlinear system.3. The nonlinear compensator of claim 1 , wherein the plurality of filters comprises at least one of:at least one nonlinear operation element; orat least one linear operation filter.4. The nonlinear compensator of claim 3 , wherein the at least one nonlinear operation element comprises at least one static nonlinear mathematical operation element.5. The nonlinear compensator of claim 3 , wherein the at least one linear operation filter comprises at least one of:at least one static linear ...

Subspace-Constrained Partial Update Method For High-Dimensional Adaptive Processing Systems

A method is explained for any adaptive processor processing digital signals by adjusting signal weights on digital signal(s) it handles, to optimize adaptation criteria responsive to a functional purpose or externalities (transient, temporary, situational, and even permanent) of that processor. Adaptation criteria for the adaptive algorithm may be any combination of a signal or parameter estimation, and measured quality(ies). This method performs a linear transformation adapting parameters from M to (M1+L) dimensions in each adaptation event, such that M1 weights are updated without constraints and M0=M−M1 weights are forced by soft constraints into an L-dimensional subspace they spanned at the beginning of the adaptation period. The same dimensionality reduction, using the same linear transformation, is applied to the input data. The reduced-dimensionality weights are then adapted using the identical optimization strategy employed by the processor, except with input data that has also been reduced in dimensionality.

SYSTEMS, APPARATUSES AND METHODS FOR ADAPTIVE NOISE REDUCTION

An apparatus includes a sensor module configured for receiving sensed information indicative of a sensed signal. The sensed signal includes a source signal component and a source noise component. The apparatus also includes a reference module configured for reference information indicative of a reference signal. The reference signal also includes a reference noise component. The apparatus also includes a filter module configured as a fixed lag Kalman smoother. The filter module is configured for adaptively filtering the reference signal to generate an estimate of the source noise component. The apparatus also includes a processing module configured for calculating an output signal based on the sensed signal and the estimate of the source noise component. The apparatus also includes an interface module configured for transmitting an indication of the output signal. The filter module is further configured for, based on the output signal, tuning the Kalman smoother. 119.-. (canceled)20. An apparatus , comprising:a memory; and receive sensed information indicative of a sensed signal, the sensed signal including a source signal component and a source noise component;', 'receive reference information indicative of a reference signal, the reference signal including a reference noise component correlated with the source noise component;', 'process, via an adaptive filter configured as a fixed lag Kalman smoother, the reference signal to generate an estimate of the source noise component;', 'determine an output signal indicative of an estimate of the source signal component by adjusting the sensed signal to account for the estimate of the source noise component; and, 'a processor operatively coupled to the memory, the processor configured totune one or more aspects of the fixed lag Kalman smoother based on the output signal.21. The apparatus of claim 20 , wherein the fixed lag Kalman smoother introduces a delay between the output signal and the reference signal such that the ...

ELECTRO-OPTICAL FINITE IMPULSE RESPONSE TRANSMIT FILTER

An electro-optical FIR transmit filter comprising a segmented MZM including a plurality of MZM segments, for receiving an input optical traveling wave to be filtered; an electrical field driver, for applying a controlled electrical field required for modulation of each MZM using a control signal which controls the electrical field; delay cells associated with at least one MZM, for aligning the control signal with a travelling optical wave; and at least one electrical xT delay cell representing a filter delay, for electrically adjusting the timing of the control signal. The FIR filter's coefficients are implemented in the optical domain by determining the amount of MZM segments driven by each xT delay cell, with respect to the total number of MZM segments. 1. An electro-optical FIR transmit filter comprising:a. a segmented MZM including a plurality of MZM segments, for receiving an input optical traveling wave to be filtered;b. an electrical field driver connected to each MZM segment, for applying a controlled electrical field required for modulation of each MZM using a control signal;c. a control signal input, for inputting said control signal to control the electrical field, required for optical wave modulation;d. at least one delay cell associated with at least one MZM, for aligning said control signal with a travelling optical wave; ande. at least one electrical xT delay cell representing a filter delay, for electrically adjusting the timing of said control signal,wherein the FIR filter's coefficients are implemented in the optical domain by determining the amount of MZM segments driven by each xT delay cell, with respect to the total number of MZM segments.2. The electro-optical FIR transmit filter of claim 1 , wherein all the electrical field drivers apply an electrical field of a constant magnitude to MZM segments.3. The electro-optical FIR transmit filter of claim 1 , wherein the FIR filter's sampling rate is implemented in the electrical domain by ...

METHOD FOR PROVIDING CONFIGURATION INFORMATION FOR A SYSTEM COMPRISING A PLURALITY OF MOVING OBJECTS

A method for providing configuration information for a system having a plurality of vehicles includes assigning, for each vehicle, a first number of degrees of freedom (DOF); presetting one or more system parameters representing a performance of at least part of the system (SP); successively optimizing each respective vehicle at each station by selecting a DOF for each vehicle from a second number of DOF for the respective vehicle and computing a global SP; assigning each vehicle of each station a DOF resulting from a first system configuration (SC); evaluating the global SP for the first SC; identifying one or more stations having a negative impact (NIS) on the global SP on the first SC; and successively optimizing, for the one or more NIS, each vehicle locally at each respective NIS with regard to the SP with a number of DOF greater than said second number of DOF. 1. A method for providing configuration information for a system comprising a plurality of vehicles , each vehicle moving in time and passing one or more stations in succession , the method to be performed in memory available to one or more processors , the method comprising:assigning, for each vehicle, a first number of degrees of freedom (DOF);presetting one or more system parameters representing a performance of at least part of the system (SP);successively optimizing each respective vehicle at each station by selecting a DOF for each vehicle from a second number of DOF for the respective vehicle, wherein the second number of DOF is smaller than the first number of DOF, and computing a global SP;assigning each vehicle of each station a DOF resulting from a first system configuration (SC);evaluating the global SP for the first SC;identifying one or more stations having a negative impact, (NIS) on the global SP on the first SC; andsuccessively optimizing, for the one or more NIS, each vehicle locally at each respective NIS with regard to the SP with a number of DOF greater than said second number of DOF ...

LEAST MEAN SQUARE METHOD FOR ESTIMATION IN SPARSE ADAPTIVE NETWORKS

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. 1. Field of the InventionThe present invention relates generally to adaptive networks, such as sensor networks, and particularly to a least mean square method for estimation in sparse adaptive networks.2. Description of the Related ArtLeast mean squares (LMS) algorithms are a class of adaptive filters used to mimic a desired filter by finding the filter coefficients that relate to producing the least mean squares of the error signal (i.e., the difference between the desired and the actual signal). The LMS algorithm is a stochastic gradient descent method, in that the filter is only adapted based on the error at the current time.In an adaptive network having N nodes, where the network has a predefined topology, for each node k, the number of neighbors is given by N, including the node k itself. In the normalized (NLMS) algorithm, at each iteration i, the output of the system at each node is given by d(i)=u(i)w+v(i), where u(i) is a known regressor row vector of length M, wis an unknown column vector of length M, and v(i) represents noise. The variable i is a time index. The output and regressor data are used to produce an estimate of the unknown vector, given by w(i). If the estimate at any time instant i of wis denoted by the vector w(i) then the estimation error is given by e(i)=d(i)−u(i)w(i). The NLMS algorithm is defined by the calculation of w(i) through the iterationwhere the superscript “T” represents the transpose of u(i) and “∥ ∥” represents the Euclidean norm. Further, μrepresents a step size, defined in the range 0<μ<2 ...

PRECISION FREQUENCY MONITOR

A precision frequency monitor provides a precision frequency monitor value (PFM) indicative of the precision of the frequency or period of an input reference signal. A first averaging module is responsive to the input reference signal to find an average frequency or period during successive predetermined time periods defining operational cycles. A second averaging module is responsive to an output of the first averaging module to average the output of the first averaging module over N operational cycles, where N is an integer, and output an updated PFM value every N operational cycles. An infinite impulse response (IIR) filter is responsive to the output of the first averaging module to filter the output of the first averaging module to output interim updated PFM values within each sequence of N operational cycles. 1. A precision frequency monitor for providing a precision frequency monitor value (PFM) indicative of the precision of the frequency or period of an input reference signal , comprising:a first averaging module responsive to the input reference signal to find an average frequency or period during successive predetermined time periods defining operational cycles;a second averaging module responsive to an output of the first averaging module to average the output of the first averaging module over N operational cycles, where N is an integer, and output an updated PFM value every N operational cycles; andan infinite impulse response (IIR) filter responsive to the output of the first averaging module to filter the output of the first averaging module to output interim updated PFM values within each sequence of N operational cycles.2. A precision frequency monitor as claimed in claim 1 , wherein the IIR filter is a one pole filter.3. A precision frequency monitor as claimed in claim 1 , wherein the second averaging module comprises an adder with a unit delay register in a feedback loop followed by a divider-by-N.4. A precision frequency monitor as claimed in ...

ESTIMATING SECONDARY PATH PHASE IN ACTIVE NOISE CONTROL

The technology described in this document can be embodied in a computer-implemented method that includes receiving, by one or more processing devices, a first plurality of values representing a set of coefficients of an adaptive filter disposed in an active noise cancellation system. The method also includes accessing one or more estimates of instantaneous phase values associated with a transfer function representing an effect of a secondary path of the active noise cancellation system, and updating the first plurality of values based on the one or more estimates of the instantaneous phase values to generate a set of updated coefficients for the adaptive filter. The method further includes programming the adaptive filter with the set of updated coefficients to affect operation of the adaptive filter. 1. A computer-implemented method comprising:receiving, by one or more processing devices, a first plurality of values representing a set of coefficients of an adaptive filter disposed in an active noise cancellation system;accessing one or more estimates of instantaneous phase values associated with a transfer function representing an effect of a secondary path of the active noise cancellation system;updating the first plurality of values based on the one or more estimates of the instantaneous phase values to generate a set of updated coefficients for the adaptive filter; andprogramming the adaptive filter with the set of updated coefficients to affect operation of the adaptive filter.2. The method of claim 1 , further comprising:generating one or more updated estimates of the instantaneous phase values;updating the first plurality of values based on the one or more updated estimates of the instantaneous phase values to generate a second set of updated coefficients for the adaptive filter; andprogramming the adaptive filter with the second set of updated coefficients to affect operation of the adaptive filter.3. The method of claim 1 , further comprising:receiving a ...

ADAPTIVE CONTINUOUS-TIME FILTER ADJUSTMENT DEVICE

A device includes a controller and an adaptive continuous-time filter that includes a control input and a first array of elements. The controller generates a digital word responsive to a time constant and compares a select bit of the digital word to a corresponding reference word to generate a control bit. The controller includes a duplicate array of elements, and applies the control bit to an adjustable element of the duplicate array of elements to modify the time constant. The controller provides the output word to the control input of the adaptive continuous-time filter to generate a filter response that accounts for effects of semiconductor process variation in the first array of elements.

RLS-DCD ADAPTATION HARDWARE ACCELERATOR FOR INTERFERENCE CANCELLATION IN FULL-DUPLEX WIRELESS SYSTEMS

An adaptation hardware accelerator comprises a calculation unit configured to receive a plurality of inputs at one or more predefined time intervals, wherein each time interval corresponds to a calculation iteration, the plurality of inputs being associated with a plurality of adaptive filters each having a plurality of taps, and determine a correlation data and a cross-correlation data based thereon for a given calculation iteration. The correlation data comprises a correlation matrix comprising a plurality of sub-matrices, wherein determining the correlation matrix comprises determining only the submatrices in an upper triangular portion and a diagonal portion of the correlation matrix. Further, the adaptation hardware accelerator comprises an adaptation core unit configured to determine a plurality of adaptive weights associated with the plurality of adaptive filters, respectively, based on an optimized RLS based adaptive algorithm, by utilizing the correlation data and the cross correlation data. In addition, the hardware accelerator unit comprises a convergence detector unit configured to determine a convergence parameter; and a controller configured to generate an iteration signal for each of the predefined time intervals based on the convergence parameter. The iteration signal communicates to the calculation unit and the adaptation core unit to continue with a next calculation iteration or to conclude, wherein the conclusion indicates a determination of a final value of the plurality of the adaptive weights by the adaptation core unit. 1. An adaptation hardware accelerator , comprising:a calculation unit configured to receive a plurality of inputs at one or more predefined time intervals, wherein each time interval corresponds to a calculation iteration, the plurality of inputs being associated with a plurality of adaptive filters each having a plurality of taps, and determine a correlation data and a cross-correlation data based thereon for a given ...

TRANSMITTER, RECEIVER AND A METHOD FOR DIGITAL MULTIPLE SUB-BAND PROCESSING

Highly efficient digital domain sub-band based receivers and transmitters. 1. A transmitter that comprises:a set of first filters with outputs connected to a summing node;a set of upsamplers;a set of sub-band processors, wherein each sub-band processor comprises an interpolator, a cyclic shifter and a second filter;wherein each cyclic shifter is coupled between an interpolator and an upsampler;wherein each first filter follows an upsampler;wherein different interpolators are arranged to receive different sets of signals and perform up-sampling;wherein the set of first filters is arranged to output virtual sub-channels of information occupying disjoint spectral sub-bands;wherein each sub-band is associated with a pair of filters that comprises a first filter and a second filter, wherein the first filter has a milder frequency response outside the sub-band than a frequency response of the second filter outside the sub-band.2. The transmitter according to claim 1 , wherein each second filter substantially nullifies spectral components outside the sub-band associated with the second filter; wherein each first filter passes spectral components that belong to at least one sub-band that differs from a sub-band associated with the first filter.3. The transmitter according to claim 1 , wherein each upsampler performs a L-factor upsampling claim 1 , and wherein each interpolator performs a V-factor upsampling; wherein L and V are positive numbers.4. The transmitter according to claim 1 , wherein L and V differ from a number (M) of the sub-bands.5. The transmitter according to claim 1 , wherein V equals 4/3.6. The transmitter according to claim 1 , wherein each interpolator comprises:a serial to parallel converter;a N point fast Fourier transform (FFT) module arranged to output N element vectors;a zero padding and circular shift module arranged to perform a zero padding operation and a circular shift operation on the N element vectors to provide V*N element circled vectors;a N ...

SUBSPACE-CONSTRAINED PARTIAL UPDATE METHODS FOR REDUCED-COMPLEXITY SIGNAL ESTIMATION, PARAMETER ESTIMATION, OR DATA DIMENSIONALITY REDUCTION

An adaptive processor implements partial updates when it adjusts weights to optimize adaptation criteria in signal estimation, parameter estimation, or data dimensionality reduction algorithms. The adaptive processor designates some of the weights to be update weights and the other weights to be held weights. Unconstrained updates are performed on the update weights, whereas updates to the set of held weights are performed within a reduced-dimensionality subspace. Updates to the held weights and the update weights employ adapt-path operations for tuning the adaptive processor to process signal data during or after tuning. 1. A method of implementing partial updates in an adaptive processor that adjusts weights to optimize an adaptation criterion in a signal estimation or parameter estimation algorithm , the method comprising:selecting, from a set of weights, a set of update weights and a set of held weights;performing unconstrained updates to the set of update weights; andperforming updates to the set of held weights within a reduced-dimensionality subspace, wherein performing updates to the set of held weights and performing unconstrained updates to the set of update weights uses adapt-path operations for tuning the adaptive processor to process signal data during or after tuning.2. The method of claim 1 , wherein performing unconstrained updates to the set of update weights and performing updates to the set of held weights comprises constructing a set of enhanced weights from the set of update weights and a projection of the set of held weights onto the reduced dimensionality subspace claim 1 , and performing an unconstrained update to the set of enhanced weights.3. The method of claim 1 , wherein performing unconstrained updates to the set of update weights and performing updates to the set of held weights comprises at least one of performing a single-port partial-update adaptation algorithm claim 1 , an uncoupled multiport adapt-path weight update algorithm ...

FILTER COEFFICIENT CALCULATION DEVICE, SIGNAL GENERATION DEVICE INCLUDING THE SAME, FILTER COEFFICIENT CALCULATION METHOD, AND SIGNAL GENERATION METHOD

A filter coefficient calculation device includes a function unit that has a plurality of functions to be executed by an FIR filter, a function selection unit that selects one or a plurality of functions from among the plurality of functions, and a filter coefficient calculation unit that calculates a filter coefficient in the selected one or plurality of functions, and is configured such that the function unit includes a first transfer function calculation unit, a second transfer function calculation unit, and a third transfer function calculation unit which calculate a transfer function of the FIR filter in the respective functions, and the filter coefficient calculation unit performs inverse Fourier transform on the transfer function in the selected one function or a product of the transfer functions in the plurality of functions to obtain an impulse response of the FIR filter and calculates the impulse response as the filter coefficient. 1. A signal generation device comprising:a filter coefficient calculation device;signal generation means for generating a signal for testing a test target device; anda finite impulse response filter which has a plurality of predetermined functions executable by the finite impulse response filter depending on a filter coefficient, for setting a filter coefficient calculated by the filter coefficient calculation device, receiving the signal from the signal generation means, performing one or some functions on the signal, and outputting the signal to the test target device,wherein the filter coefficient calculation device comprises:a plurality of transfer function calculation means for calculating transfer functions of the finite impulse response filter, each of the plurality of transfer function calculation means corresponds to each of the plurality of predetermined functions of the finite impulse response filter;selection means for selecting one or some of the transfer functions in the plurality of the transfer function ...

SUBBAND ADAPTIVE FILTER FOR SYSTEMS WITH PARTIALLY ACAUSAL TRANSFER FUNCTIONS

A noise reduction system includes sensors configured to generate an input signal, an adaptive filter configured to represent a transfer function of a path traversed by the input signal, one or more processing devices, and one or more transducers. The processing devices receive the input signal and generate an updated set of filter coefficients of the adaptive filter by separating the input signal into frequency subbands; determining for each subband, coefficients of a corresponding subband adaptive module; and combining the coefficients of multiple subband adaptive modules. Determining the coefficients of the corresponding subband adaptive module includes selecting a subset of a precomputed set of filter coefficients of the adaptive filter. The processing devices process a portion of the input signal using the updated set of filter coefficients of the adaptive filter to generate an output that destructively interferes with another signal traversing the path represented by the transfer function. 1. A method comprising:receiving, at one or more processing devices, an input signal;receiving, at the one or more processing devices, a feedback signal; determining, for each of multiple frequency subbands of the input signal, one or more coefficients of a corresponding subband adaptive filter, and', 'combining one or more coefficients of multiple subband adaptive filters to generate the updated set of filter coefficients of the adaptive filter,, 'generating, based on the input signal and the feedback signal, an updated set of filter coefficients of an adaptive filter, wherein generating the updated set of filter coefficients of the adaptive filter comprises (a) obtaining a precomputed set of filter coefficients of the adaptive filter, and', '(b) selecting a subset of the precomputed set of filter coefficients of the adaptive filter; and, 'wherein determining the one or more coefficients of the corresponding subband adaptive filter comprisesprocessing a portion of the input ...

Method and apparatus for providing adaptive display and filtering of sensors and sensor data

An approach is provided for adaptive display and filtering of sensors and sensor data. A sensor manager determines one or more signals associated with one or more sensors. The sensor manager then processes and/or facilitates a processing of the one or more signals for comparison against one or more predetermined signals. The sensor manager determines one or more parameters for one or more filters based, at least in part, on the comparison, wherein the one or more filters operate, at least in part, on the one or more sensors, one or more other signals determined form the one or more sensors, or a combination thereof.

SYSTEMS AND METHODS FOR AN ADJUSTABLE FILTER ENGINE

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.

BIQUAD STAGE HAVING A SELECTABLE BIT PRECISION

An apparatus includes a plurality of delay elements, a plurality of multipliers and an accumulator to form a biquad stage; and a precision logic circuit. The biquad stage includes feedback paths; at least one feedback path has an adjustable bit precision; and the precision logic is adapted to regulate the bit precision of the feedback path(s) based at least in part on at least one parameter that is associated with the biquad stage. 1. An apparatus comprising:a plurality of delay elements, a plurality of multipliers and an accumulator to form a biquad stage; anda precision logic circuit; the biquad stage comprises a plurality of feedback paths;', 'at least one feedback path of the plurality of feedback paths has an adjustable bit precision; and', 'the precision logic circuit is adapted to regulate the bit precision of the at least one feedback path based at least in part on at least one parameter associated with the biquad stage., 'wherein2. The apparatus of claim 1 , wherein the precision logic circuit regulates the bit precision based at least in part on at least one filtering coefficient.3. The apparatus of claim 2 , wherein the precision logic circuit is adapted to:{'sub': 1', '2', '2, 'determine a metric 1−a−a, where a1 is a first feedback coefficient of the biquad stage and, ais a second filtering coefficient of the biquad stages; and'}regulate the bit precision based on a comparison of the metric to a threshold.4. The apparatus of claim 2 , further comprising:{'sub': 1', '2, 'a coefficient calculation circuit to determine the aand afeedback coefficients based at least in part on at least one parameter associated a transfer function of the biquad stage.'}5. The apparatus of claim 1 , wherein the apparatus further comprises:a circuit to regulate the bit precision based at least in part on at least one parameter associated with a transfer function of the biquad stage.6. The apparatus of claim 1 , wherein:the at least one feedback path comprises a first delay ...

CARDIAC ELECTRICAL SIGNAL GROSS MORPHOLOGY-BASED NOISE DETECTION FOR REJECTION OF VENTRICULAR TACHYARRHYTHMIA DETECTION

A medical device system, such as an extra-cardiovascular implantable cardioverter defibrillator ICD, senses R-waves from a first cardiac electrical signal by a first sensing channel and stores a time segment of a second cardiac electrical signal in response to each sensed R-wave. The medical device system determines a morphology parameter correlated to signal noise from time segments of the second cardiac electrical signal, detects a noisy signal segment based on the signal morphology parameter; and withholds detection of a tachyarrhythmia episode in response to detecting a threshold number of noisy signal segments. 1. A medical device system comprising: a first sensing channel configured to receive a first cardiac electrical signal via a first sensing electrode vector coupled to the medical device system and to sense R-waves in response to crossings of a sensing amplitude threshold by the first cardiac electrical signal, and', 'a second sensing channel configured to receive a second cardiac electrical signal via a second sensing electrode vector coupled to the medical device system and different than the first sensing electrode vector;, 'a sensing circuit comprisinga memory; and store a time segment of the second cardiac electrical signal in the memory for each of the plurality of R-waves sensed by the first sensing channel;', determine a morphology parameter correlated to signal noise from the stored time segment; and', 'detect the stored time segment as being a noisy signal segment based on the morphology parameter determined for the respective stored time segment; and, 'for each of a plurality of the stored time segments,'}, 'withhold detection of a tachyarrhythmia episode in response to detecting at least a threshold number of the stored time segments as noisy signal segments., 'a control circuit coupled to the sensing circuit and the memory, the control circuit configured to2. The system of claim 1 , further comprising a notch filter claim 1 , wherein the time ...

Digital equalizer with overlappable filter taps

In one illustrative embodiment, an equalizer includes: a shift register, an array of multipliers, an array of multiplexers, and a summer. The shift register provides receive signal samples at each tap. Each multiplier in the array multiplies one of said receive signal samples by a respective coefficient to produce a product, with at least one of said multipliers coupled to a fixed tap. Each multiplexer in the array supplies an associated one of said multipliers with a receive signal sample from a selectable tap. The summer sums the products to produce a filtered output signal. To reduce hardware requirements, coefficient multipliers may be multiplexed to a reduced set of taps, and the dynamic range of the coefficients may be increased by overlapping the sets for different multipliers. Methods of tap selection and coefficient adaptation are disclosed.

CUSTOMIZABLE DATA AGGREGATING, DATA SORTING, AND DATA TRANSFORMATION SYSTEM

A customizable data aggregating, data sorting, and data transformation system is disclosed. In particular, the system may allow for the application of various filters to a sample of data corresponding to various measurables associated with objects. A mean and standard deviation for each of the measurables in the filtered sample of data may be calculated and may be utilized in determining z-scores for a first set of raw measurements corresponding to the measurables. Once the z-scores for the first set of raw measurements are determined, selected weights may be applied to each of the z-scores to determine a weighted z-score for each of the measurables in the first set. Each weighted z-score may then be aggregated to generate a score for an object associated with the first set. The score for the object may be utilized to rank the object relative to other objects in the filtered sample of data. 1. A system , comprising:a memory that stores instructions; and applying a first filter to a sample of data corresponding to a plurality of measurable actions performed by a plurality of objects to generate a first filtered sample of data corresponding to a first subset of the plurality of measurable actions;', 'calculating a mean value for each of the plurality of measurable actions in the first filtered sample of data and a standard deviation value for each of the plurality of measurable actions in the first filtered sample of data;', 'selecting a first set of raw measurements for a first set of measurable actions for a first object of the plurality of objects, wherein the first set of measurable actions correspond to the first subset of the plurality of measureable actions;', 'calculating a z-score for each of the measurable actions in the first set of measurable actions, wherein the z-score is calculated based on the mean value, the standard deviation value, and the first set of raw measurements;', 'calculating, based on a weight to be applied to each z-score for each of the ...

METHOD FOR VALIDATING A SOFTWARE

A method for validating a software, particularly a driver-assistance software. The method includes receiving of a sensor signal that is to be processed by the software; determining of a reference signal by an adaptive filter based on the sensor signal, the reference signal representing an anticipated sensor signal; determining an error signal based on the sensor signal and the reference signal, the error signal representing a performance of the software; determining an anti-sensor signal by a machine-learning unit based on the sensor signal, the machine-learning unit being trained with sensor signals already evaluated; controlling of the adaptive filter by a control unit based on the determined error signal and the adaptive anti-sensor signal; and validation of the software based on the determined error signal. 1. A method for validating a software , comprising the following steps:receiving a sensor signal that is to be processed by the software;determining, by an adaptive filter, a reference signal based on the sensor signal, the reference signal representing an anticipated sensor signal;determining an error signal based on the sensor signal and the reference signal, the error signal representing a performance of the software;determining, by a machine-learning unit, an anti-sensor signal based on the sensor signal, the machine-learning unit being trained with sensor signals already evaluated;controlling, by a control unit, the adaptive filter based on the determined error signal and the determined anti-sensor signal; andvalidating the software based on the determined error signal.2. The method as recited in claim 1 , wherein the software is a driver-assistance software.3. The method as recited in claim 1 , wherein the adaptive filter is actively controlled by the control unit.4. The method as recited in claim 1 , wherein the control unit uses a modified least-mean-square (LMS) algorithm as a control algorithm for controlling the adaptive filter.5. The method as ...

Adaptive filter control

A sound processing circuit comprises a first input for receiving a first input signal, and a second input for receiving a second input signal. A first adaptive filter receives the first input signal, and an error calculation block calculates an error between the second input signal and the output of the first adaptive filter, and outputting an error signal. A second adaptive filter receives the error signal, and an output calculation block subtracts an output of the second adaptive filter from the first input signal to generate an output signal. The adaptation of first and second adaptive filters is controlled based on a magnitude coherence between the first and second input signals.

METHOD AND APPARATUS FOR PARALLELIZED QRD-BASED OPERATIONS OVER A MULTIPLE EXECUTION UNIT PROCESSING SYSTEM

Methods and apparatuses relating to QR decomposition using a multiple execution unit processing system are provided. A method includes receiving input values at the processing system and generating a first set of values based on the input values, where at least some of the first values are computed in parallel. A second set of values are generated recursively based on values in the first set. A third set of values are generated based on values in the second set, where at least some of the values in the third set are computed in parallel. The recursive component may be simplified to consist of one or more low latency operations. The processing performance of operations relating to QR decomposition may therefore be improved by using the parallelism available in multiple execution unit systems. 1. A method for adapting a filter in signal processing , the method comprising:{'sub': i', 'i', 'i', 'N, 'generating values vbased on values uin an input signal, the values vbeing generated in parallel, where i=0, 1, 2, . . . , N, u=d, wherein d is an output signal received from the filter;'}{'sub': i', 'i, 'generating values Γrecursively based on the values v;'}{'sub': i', 'i', 'i', 'i, 'generating values D(i) and L(i) based on values sand the values Γ, the values D(i) and L(i) being generated in parallel, where the values sare conjugates or complex conjugates of the values u; and'}{'sub': 'i', 'generating a signal W according to the values u, D(i) and L(i).'}2. The method of claim 1 , wherein the generating values vinvolves generating square values of input signal values u.3. The method of claim 2 , wherein the generating values Γinvolves generating values Γaccording to equation Γ=Γ+v claim 2 , where Γ=1 and i=1 claim 2 , 2 claim 2 , 3 claim 2 , . . . claim 2 , N.6. The method of claim 5 , further comprising outputting the signal W to the filter.7. An apparatus for adapting a filter in signal processing claim 5 , the apparatus comprising: [{'sub': 'i', 'a first module for ...

Digital quadrature modulator and switched-capacitor array circuit

A digital quadrature modulator holds local oscillator circuitry configured to provide local oscillator signals, and local oscillator polarity logic circuitry configured to select an In-phase and a Quadrature local oscillator signal according to a sign bit of an In-phase control word and a sign bit of a Quadrature control word, respectively. The modulator holds a number of local oscillator control logic circuits, each configured to generate a conditioned signal by gating one or both of the selected local oscillator signals according to values of the In-phase control word and/or values of the Quadrature control word. The modulator has one or more sets of switched-capacitor units, where each unit has an output provided by an output capacitor, and where a signal at the input side of the output capacitor is controlled by a conditioned signal. The outputs of at least two of the switched-capacitor units are combined in a common node.

METHOD AND DEVICE FOR AUDIO SIGNAL PROCESSING

The present invention relates to a method and an apparatus for processing a signal, which are used for effectively reproducing an audio signal, and more particularly, to a method and an apparatus for processing a signal, which are used for implementing binaural rendering for reproducing multi-channel or multi-object audio signals in stereo with a low calculation amount. 1. A method for processing an audio signal , the method comprising:receiving an input audio signal including a plurality of subband signals, wherein the plurality of subband signals include a first subband group containing only low-frequency subband signals determined based on a predetermined frequency band and a second subband group containing only high-frequency subband signals determined based on the predetermined frequency band;performing a first filtering on each subband signal of the first subband group;performing a second filtering on each subband signal of the second subband group, wherein the second filtering is distinct from the first filtering.2. The method of claim 1 , wherein the first filtering is performed by using a first parameter set including one or more parameters for a corresponding subband and the second filtering is performed by using a second parameter set including one or more parameters for a corresponding subband claim 1 , andwherein characteristics of the first parameter set is different from characteristics of the second parameter set.3. The method of claim 2 , wherein the number of one or more parameters of a second parameter set for a subband of the first subband group is equal to or less than the number of one or more parameters of a first parameter set for a subband of the first subband group.4. The method of claim 2 , wherein the first parameter set includes a set of subband filter coefficients obtained as at least a portion of proto-type subband filter coefficients for the corresponding subband claim 2 , andwherein the first filtering performs fast convolution on ...

SIGNAL CONDITIONING CIRCUIT AND A RELAY/CIRCUIT BREAKER CONTROL APPARATUS INCLUDING SUCH A SIGNAL CONDITIONING CIRCUIT

There is a need to monitor and control the state of relays and circuit breakers within power distribution systems. The monitoring of a relay state, i.e. open or closed, is often performed by applying a monitoring signal to sensing contacts added to the relay. Manufactures of such systems have chosen many different voltages for their own. monitoring systems making it difficult to interconnect dissimilar monitoring systems. A signal conditioning circuit is provided that can cope with a large input voltage range and can be configured to allow may items of equipment(which may be new item or legacy items) to be connected to a controller. 1. A signal conditioning circuit , the signal conditioning circuit comprising at least one input node connected to an input stage for processing an input signal received at the at least one input node , and an adjustable load to apply a load at the at least one input node.2. A signal conditioning circuit as claimed in in which the conditioning circuit converts the input signal received at the at least one input node to a binary signal.3. A signal conditioning circuit as claimed in in which the signal conditioning circuit includes at least one programmable attenuator claim 1 , or at least one fixed attenuator and a programmable amplifier.4. A signal conditioning circuit as claimed in in which the signal conditioning circuit includes a programmable threshold circuit and/or comparator.5. A signal conditioning circuit as claimed in in which the adjustable load responds to the rate of change of voltage at the at least one input signal node to increase the current drawn by the load.6. A signal conditioning circuit as claimed in . in which the adjustable load responds to the voltage at the at least one input node.7. A signal processing circuit as claimed in in which the adjustab passes a current claim 1 , and where the current is controlled by a current source.8. A signal processing circuit as claimed in claim I further comprising a ...

System and Method for Adaptive Filter

In one embodiment, a method for training an adaptive filter includes receiving, by a processor from a device, an input signal and a training reference signal and determining a correlation matrix in accordance with the input signal, the training reference signal, and a filter type. The method also includes determining a plurality of coefficients in accordance with the correlation matrix and adjusting the adaptive filter in accordance with the plurality of coefficients. 1. A method for training an adaptive filter , the method comprising:receiving, by a processor from a device, an input signal and a training reference signal;determining a correlation matrix in accordance with the input signal, the training reference signal, and a filter type;determining a plurality of coefficients in accordance with the correlation matrix; andadjusting the adaptive filter in accordance with the plurality of coefficients.2. The method of claim 1 , further comprising determining a U vector claim 1 , wherein determining the correlation matrix further comprises determining the correlation matrix in accordance with the U vector.3. The method of claim 2 , wherein determining the U vector comprises determining the U vector in accordance with the input signal and the filter type.4. The method of claim 2 , wherein determining the U vector comprises determining the U vector in accordance with the training reference signal and the filter type.5. The method of claim 2 , further comprising determining a size of the U vector.6. The method of claim 1 , further comprising determining a size of the correlation matrix.7. The method of claim 1 , further comprising:determining a first forgetting factor in accordance with the filter type;determining a second forgetting factor in accordance with the filter type; anddetermining a third forgetting factor in accordance with the filter type, wherein determining the correlation matrix further comprises determining the correlation matrix in accordance with the ...

METHOD FOR REDUCING LOUDSPEAKER PHASE DISTORTION

A method for reducing loudspeaker magnitude and/or phase distortion, in which one or more filters pertaining to one or more drive units is automatically generated or modified based on the response of each specific drive unit. The drive unit response may be determined by electromechanical modelling of the drive unit. Drive unit models may be enhanced by electromechanical and/or acoustic measurement such that the resulting filter becomes specific to each specific drive unit. 1. A method for reducing loudspeaker magnitude and/or phase distortion , in which one or more filters pertaining to one or more drive units is automatically generated or modified based on the response of each specific drive unit and in which improved drive unit model or measurement data is stored remotely and sent over the internet to update the filter or filters for a specific drive unit.2. The method of claim 1 , in which the drive unit response is determined by modelling the drive unit.3. The method of claim 1 , in which the drive unit response is determined by electro-mechanical modelling of the drive unit.4. The method of claim 3 , in which the electro-mechanical modelling is enhanced by electro-mechanical measurement of a specific drive unit such that the resulting filter becomes specific to that drive unit.5. The method of in which the electro-mechanical modelling of the drive unit is defined using any one or more of the parameters f claim 3 , Q claim 3 , R claim 3 , Lor L.6. The method of claim 2 , in which the drive unit response is determined by acoustic modelling of the drive unit.7. The method of claim 2 , in which the modelling incorporates any electronic passive filtering in front of the drive unit.8. The method of claim 3 , in which the electro-mechanical modelling is enhanced by electro-mechanical measurement of the passive filtering in front of each drive unit.9. The method of claim 2 , in which the modelling is enhanced by the use of acoustic measurements of a specific drive unit ...

METHOD OF COMPRESSING CONVOLUTION PARAMETERS, CONVOLUTION OPERATION CHIP AND SYSTEM

A method for compressing multiple original convolution parameters into a convolution operation chip includes steps of: determining a range of the original convolution parameters; setting an effective bit number for the range; setting a representative value, wherein the representative value is within the range; calculating differential values between the original convolution parameters and the representative value; quantifying the differential values to a minimum effective bit to obtain a plurality of compressed convolution parameters; and transmitting the effective bit number, the representative value and the compressed convolution parameters to the convolution operation chip. 1. A method for compressing multiple original convolution parameters into a convolution operation chip , comprising steps of:determining a range of the original convolution parameters;setting an effective bit number for the range;setting a representative value, wherein the representative value is within the range;calculating differential values between the original convolution parameters and the representative value;quantifying the differential values to a minimum effective bit to obtain a plurality of compressed convolution parameters; andtransmitting the effective bit number, the representative value and the compressed convolution parameters to the convolution operation chip.2. The method according to claim 1 , wherein the original convolution parameters are filter coefficients or weight coefficients of filters.3. The method according to claim 1 , wherein the representative value is an average value claim 1 , a maximum value or a minimum value of the original convolution parameters.4. The method according to claim 1 , wherein the step of determining the range of the original convolution parameters comprises determining a maximum value and a minimum value of the original convolution parameters.5. The method according to claim 1 , wherein a minimum effective bit of the original convolution ...

Method And System for A Distributed Optoelectronic Receiver

Methods and systems for a distributed optoelectronic receiver are disclosed and may include an optoelectronic receiver having a grating coupler, a splitter, a plurality of photodiodes, and a plurality of transimpedance amplifiers (TIAs). The receiver receives a modulated optical signal utilizing the grating coupler, splits the received signal into a plurality of optical signals, generates a plurality of electrical signals from the plurality of optical signals utilizing the plurality of photodiodes, communicates the plurality of electrical signals to the plurality of TIAs, amplifies the plurality of electrical signals utilizing the plurality of TIAs, and generates an output electrical signal from coupled outputs of the plurality of TIAs. Each TIA may be configured to amplify signals in a different frequency range. One of the plurality of electrical signals may be DC coupled to a low frequency TIA of the plurality of TIAs. 1. A method for communication , the method comprising: receiving a modulated optical signal utilizing the grating coupler;', 'splitting the received signal into a plurality of optical signals using the splitters;', 'generating a plurality of electrical signals from the plurality of optical signals utilizing the photodiodes;', 'communicating the plurality of electrical signals to the TIAs;', 'amplifying the plurality of electrical signals utilizing the TIAs, wherein each TIA is configured to amplify signals in a different frequency range; and', 'generating an output electrical signal from coupled outputs of the TIAs., 'in an optoelectronic receiver finite impulse response (FIR) filter comprising a grating coupler, splitters, photodiodes, and transimpedance amplifiers (TIAs)2. The method according to claim 1 , comprising DC coupling one of the plurality of electrical signals to a low frequency TIA of the TIAs.3. The method according to claim 1 , comprising AC coupling one of the plurality of electrical signals to a high frequency TIA of the TIAs.4. ...

SYSTEMS, APPARATUSES AND METHODS FOR ADAPTIVE NOISE REDUCTION

An apparatus includes a sensor module configured for receiving sensed information indicative of a sensed signal. The sensed signal includes a source signal component and a source noise component. The apparatus also includes a reference module configured for reference information indicative of a reference signal. The reference signal also includes a reference noise component. The apparatus also includes a filter module configured as a fixed lag Kalman smoother. The filter module is configured for adaptively filtering the reference signal to generate an estimate of the source noise component. The apparatus also includes a processing module configured for calculating an output signal based on the sensed signal and the estimate of the source noise component. The apparatus also includes an interface module configured for transmitting an indication of the output signal. The filter module is further configured for, based on the output signal, tuning the Kalman smoother. 2. The apparatus of claim 1 , wherein the sensed information includes sensed photoplethysmographic information claim 1 , further comprising a photoplethysmographic sensor configured for capturing the photoplethysmographic information.3. The apparatus of claim 1 , wherein the reference information includes accelerometer information claim 1 , further comprising an accelerometer configured for capturing the reference information.5. The apparatus of claim 4 , the interface module further configured for modifying at least one of the sensed signal threshold claim 4 , the reference signal threshold claim 4 , the state noise covariance threshold claim 4 , or the state noise covariance increment.6. The apparatus of claim 1 , the filter module configured to tune the Kalman smoother by modifying the oblivion coefficient.7. The apparatus of claim 1 , the filter module configured to tune the Kalman smoother by modifying the order of the Kalman smoother.8. The apparatus of claim 1 , the processing module configured to ...

Adaptive equalizer system

One example includes an equalizer system. The system includes a filter system configured to receive digital sample blocks associated with an input signal and to provide equalized digital sample blocks associated with the respective digital sample blocks based on adaptive tap weights. Each of the digital sample blocks includes samples and each of the equalized digital sample blocks includes equalized samples. The system also includes a sample set selector to select a subset of equalized samples from each of the equalized digital sample blocks at the output of the filter and an error estimator configured to implement an error estimation algorithm on the subset of the equalized samples to determine a residual error associated with the equalized samples. The system further includes a tap weight generator configured to generate the adaptive tap weights in response to the residual error and to provide the adaptive tap weights to the filter.

CUSTOMIZABLE DATA AGGREGATING, DATA SORTING, AND DATA TRANSFORMATION SYSTEM

A customizable data aggregating, data sorting, and data transformation system is disclosed. In particular, the system may allow for the application of various filters to a sample of data corresponding to various measurables associated with objects. A mean and standard deviation for each of the measurables in the filtered sample of data may be calculated and may be utilized in determining z-scores for a first set of raw measurements corresponding to the measurables. Once the z-scores for the first set of raw measurements are determined, selected weights may be applied to each of the z-scores to determine a weighted z-score for each of the measurables in the first set. Each weighted z-score may then be aggregated to generate a score for an object associated with the first set. The score for the object may be utilized to rank the object relative to other objects in the filtered sample of data. 1. A system , comprising:a memory that stores instructions; and applying a first filter to a sample of data corresponding to a plurality of measurable actions performed by a plurality of objects to generate a first filtered sample of data corresponding to a first subset of the plurality of measurable actions;', 'calculating a mean value for each of the plurality of measurable actions in the first filtered sample of data and a standard deviation value for each of the plurality of measurable actions in the first filtered sample of data;', 'selecting a first set of raw measurements for a first set of measurable actions for a first object of the plurality of objects, wherein the first set of measurable actions correspond to the first subset of the plurality of measureable actions;', 'calculating a z-score for each of the measurable actions in the first set of measurable actions, wherein the z-score is calculated based on the mean value, the standard deviation value, and the first set of raw measurements;', 'calculating, based on a weight to be applied to each z-score for each of the ...

Method and apparatus for nonlinear signal processing

The present disclosure relates to a concept of nonlinear signal processing which may be used for predistortion for RF power amplifiers. The concept includes generating time variant filter coefficients for a linear filter circuit based on a nonlinear mapping of an input signal, and filtering the input signal with the linear filter circuit using the time variant filter coefficients in order to generate a filtered output signal. Thus, it is proposed to implement a non-linear filter by a time-varying linear filter where the time-varying coefficients are derived from the input signal.

Method And System for A Distributed Optoelectronic Receiver

Methods and systems for a distributed optoelectronic receiver are disclosed and may include an optoelectronic receiver having a grating coupler, a splitter, a plurality of photodiodes, and a plurality of transimpedance amplifiers (TIAs). The receiver receives a modulated optical signal utilizing the grating coupler, splits the received signal into a plurality of optical signals, generates a plurality of electrical signals from the plurality of optical signals utilizing the plurality of photodiodes, communicates the plurality of electrical signals to the plurality of TIAs, amplifies the plurality of electrical signals utilizing the plurality of TIAs, and generates an output electrical signal from coupled outputs of the plurality of TIAs. Each TIA may be configured to amplify signals in a different frequency range. One of the plurality of electrical signals may be DC coupled to a low frequency TIA of the plurality of TIAs. 1. A method for communication , the method comprising: receiving a modulated optical signal utilizing the grating coupler;', 'splitting the received signal into a plurality of optical signals using the splitters;', 'generating a plurality of electrical signals from the plurality of optical signals utilizing the photodiodes;', 'amplifying the plurality of electrical signals utilizing the TIAs, wherein each TIA is configured to amplify signals in a different frequency range; and', 'discriminating, utilizing the processor, optical frequencies of the received signal based on magnitudes of output signals from each of the TIAs., 'in an optoelectronic receiver comprising a grating coupler, splitters, photodiodes, a processor, and transimpedance amplifiers (TIAs)2. The method according to claim 1 , comprising DC coupling one of the plurality of electrical signals to a low frequency TIA of the TIAs.3. The method according to claim 1 , comprising AC coupling one of the plurality of electrical signals to a high frequency TIA of the TIAs.4. The method according ...

DYNAMIC BIAS CONTROL

Systems and methods for controlling a power amplifier includes combining a digital modulated data signal with a digital bias signal to generate a combined digital signal, the digital bias signal generated based on an envelope for the modulated data signal; converting, by a digital-to-analog converter, the combined digital signal into a combined analog signal, the combined analog signal comprising an analog modulated data signal and an analog envelope bias signal; and separating the analog modulated data signal and the analog bias signal onto separate signal paths, wherein the converting is performed using a single digital-to-analog converter. 126-. (canceled)27. A system comprising:a digital envelope detector operable to generate a digital bias according to a digital signal;a digital-to-analog converter (DAC) operable to generate a biased analog signal according to the digital signal and the digital bias;a power amplifier operable to generate an amplifier output according to an amplifier input and an amplifier control;a high pass filter operable to attenuate an analog bias of the biased analog signal to generate the amplifier input; anda low pass filter operable to pass the analog bias of the biased analog signal to generate the amplifier control.28. The system of claim 27 , wherein the control input of the power amplifier is operable to control a bias of the power amplifier.29. The system of claim 27 , wherein the digital signal comprises a data signal modulated on a carrier.30. The system of claim 27 , wherein the DAC comprises a digital combiner.31. The system of claim 27 , wherein the system comprises an adaptive filter operably coupled between the digital envelope detector and the DAC.32. The system of claim 31 , wherein the system comprises a return path operably coupled to the amplifier output.33. The system of claim 32 , wherein the return path comprises an analog-to-digital converter (ADC).34. The system of claim 32 , wherein an output of the ADC provides a ...

ANALOG/DIGITAL CONVERTER AND MILIMETER WAVE RADAR SYSTEM

A modulator includes an analog integrator including an analog circuit and a quantizer quantizing its output signal. An external input signal is input thereto. A modulator is coupled to the latter stage of the modulator, and includes a quantizer. A probe signal generation circuit injects a probe signal to the modulator. An adaptive filter searches for a transfer function of the modulator by observing an output signal of the quantizer in accordance with a probe signal. Another adaptive filter searches for a transfer function of the modulator by observing an output signal of the quantizer in accordance with the probe signal. A noise cancel circuit cancels a quantization error generated by the quantizer using search results of the adaptive filters. 1. A MASH (Multi stAge Noise SHaping) type and sigma-delta type analog/digital converter comprising:a first modulator which includes a first analog integrator including an analog circuit and a first quantizer quantizing an output signal of the first analog integrator, and to which an external input signal as an analog signal is input;a second modulator which is coupled to a latter stage of the first modulator and includes a second quantizer;a probe signal generation circuit which injects a probe signal to the first modulator;a first adaptive filter which searches for a transfer function of the first modulator by observing an output signal of the first quantizer in accordance with the probe signal;a second adaptive filter which searches for a transfer function of the second modulator by observing an output signal of the second quantizer in accordance with the probe signal; anda noise cancel circuit which cancels an quantization error generated by the first quantizer using a search result of the first adaptive filter and a search result of the second adaptive filter.2. The analog/digital converter according to claim 1 ,wherein the probe signal is a pseudo random signal.3. The analog/digital converter according to claim 2 , ...

SUBBAND ADAPTIVE FILTER FOR SYSTEMS WITH PARTIALLY ACAUSAL TRANSFER FUNCTIONS

A noise reduction system includes sensors configured to generate an input signal, an adaptive filter configured to represent a transfer function of a path traversed by the input signal, one or more processing devices, and one or more transducers. The processing devices receive the input signal and generate an updated set of filter coefficients of the adaptive filter by separating the input signal into frequency subbands; determining for each subband, coefficients of a corresponding subband adaptive module; and combining the coefficients of multiple subband adaptive modules. Determining the coefficients of the corresponding subband adaptive module includes selecting a subset of a precomputed set of filter coefficients of the adaptive filter. The processing devices process a portion of the input signal using the updated set of filter coefficients of the adaptive filter to generate an output that destructively interferes with another signal traversing the path represented by the transfer function. 1. A method for estimating coefficients of an adaptive filter , the method comprising:receiving, at one or more processing devices, an input signal;receiving, at the one or more processing devices, a feedback signal; (i) separating the input signal into multiple frequency subbands,', '(ii) separating the feedback signal into the multiple frequency subbands,', '(iii) determining, for each frequency subband, one or more coefficients of a corresponding subband adaptive module, and', '(iv) combining one or more coefficients of multiple subband adaptive modules to generate the updated set of filter coefficients of the adaptive system identification filter,, 'generating, based on the input signal and the feedback signal, an updated set of filter coefficients of an adaptive system identification filter configured to represent a transfer function of a path traversed by the input signal, wherein generating the updated set of filter coefficients of the adaptive system identification ...

Drift suppression filter, proximity detector and method

A portable device including drift-compensated capacitive proximity sensor that exploits a special method of drift compensation based on the variation of the measured proximity signal. The drift is tracked when the variation is within a stated interval, and frozen when the variation is outside. The sensor is capable of following a drift not only when the phone is inactive, but also when it is close to the body of the user, by freezing the tracking when the capacity varies steeply, as when the user moves the device, and resuming it when the variation is within acceptable limits.

Apparatus, System, and Method for an Acoustic Response Monitor

An acoustic response monitor. The acoustic response monitor includes a speaker, a microphone, and a response analyzer in electrical communication with the speaker and the microphone. The speaker is configured to generate a sound in response to an excitation signal. The microphone is configured to generate a microphone signal in response to a sound. The response analyzer is configured to generate an adaptive filter to minimize a difference between the excitation signal as modified by the adaptive filter and the microphone signal. The response analyzer may be configured to determine a difference between the adaptive filter and a previously generated adaptive filter. The response analyzer may be configured to trigger an alarm if the difference exceeds a predetermined threshold. 1. An acoustic response monitor comprising:a speaker configured to generate a sound in response to an excitation signal;a microphone configured to generate a microphone signal in response to a sound; generate an adaptive filter to minimize a difference between the excitation signal as modified by the adaptive filter and the microphone signal;', 'determine a difference between the adaptive filter and a previously generated adaptive filter;', 'trigger an alarm if the difference exceeds a predetermined threshold., 'a response analyzer in electrical communication with the speaker and the microphone, the response analyzer configured to2. The acoustic response monitor of claim 1 , wherein the adaptive filter is correlated to an acoustic response of a monitored volume.3. The acoustic response monitor of claim 2 , wherein a determination that a characteristic of the monitored volume has changed is correlated to the difference exceeding the predetermined threshold.4. The acoustic response monitor of claim 2 , wherein the monitored volume comprises a gas.5. The acoustic response monitor of claim 2 , wherein the monitored volume comprises a liquid.6. The acoustic response monitor of claim 2 , wherein the ...

Compensating for Channel Distortion During Contactless Communication

Systems, methods, and devices are provided for compensating for distortion of a contactless communication channel. The electronic device may include a radio frequency system that itself includes antenna to transmit and receive data using near-field communication (NFC) and an NFC signal processing circuitry. The NFC signal processing circuitry may receive an NFC signal via a communication channel formed between the electronic device and another electronic device and may determine a baseband reference waveform associated with the electromagnetic NFC signal and may determine an error between a portion of the electromagnetic NFC signal and the baseband reference waveform. Furthermore, the NFC signal processing circuitry may determine whether the error is outside of an acceptable error threshold range and, in response to the error being outside of the acceptable error threshold range, train a filter response of the NFC signal processing circuitry to estimate the communication channel. 1. An electronic device having a radio frequency system comprising:an antenna configured to receive data using near-field communication (NFC); and receive an electromagnetic NFC signal via a communication channel formed between the electronic device and a other electronic device via the antenna;', 'evaluate the electromagnetic NFC signal to identify one NFC communication protocol of a plurality of NFC communication protocols used by the electromagnetic NFC signal; and', 'adaptively estimate, as an adaptive estimation, the communication channel based, at least in part, on the identified one NFC communication protocol., 'NFC signal processing circuitry configured to2. The electronic device of claim 1 , wherein the one NFC communication protocol is identified using a plurality of stored reference signals corresponding to at least some of the plurality of NFC communication protocols.3. The electronic device of claim 1 , wherein a filter response of the NFC signal processing circuitry is trained ...

Adaptive automatic equalizer

An adaptive automatic equalizer of the type comprising a correction circuit for calculating a correction value for the weighting coefficients of an input signal at respective delay stages stored in the coefficient memory devices based on a difference signal between the output signal of the equalizer and a desired signal, and for correcting the weighting coefficients in accordance with the correction value, a detector for detecting a rapid change of the input signal, and an inhibiting circuit for inhibiting a substantial correction of the weighting coefficients effected by the correcting device when the rapid change is detected by the detector during an interval between the rapid change and return of the input signal to a stable state.

Signal processing circuit

A signal processing circuit has an IIR filter (11) and an FIR filter (12) that always have equivalent transmission functions, where, when in an adjusting mode for adjusting to any given transmission function, the configuration is set in the IIR filter (11), and then after the adjustment has been completed, or when in signal processing mode, the configuration is changed to the FIR filter of the equivalent transmission function.

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Adaptive automatic equalizer

내용 없음. No content.

Method of processing hf signals with stage of adaptive filters with different response with common feedback by decision

FIELD: electronics. SUBSTANCE: invention relates to adaptive filtration, in particular to digital signal processing systems, in which for protection against signal distortion adaptive antenna systems are used in communication channel. Signal is processed by a broadband adaptive antenna system (BAAS) operating based on a Wiener algorithm and a minimum mean square error (MMSE) criterion, wherein output signal y(k) from BAAS, which is used as a first stage, is processed with an additional adaptive filter, which is used as a second stage, controlled by a Kalman algorithm, total reference estimation signal for BAAS and additional adaptive filter and error signal (e) 2 (k) for additional adaptive filter, generated by difference between total reference estimation signal and output signal with additional adaptive filter, transmitted to a decision circuit, which generates total reference estimation signal , which provides combined control communication between BAAS and additional adaptive filter, wherein size L II of sliding window of Kalman algorithm is less than size L I of sliding window of Wiener algorithm controlled by total reference estimation signal and error signal (e) 1 (k) for BAAS, generated by difference between total reference estimation signal and output signal y(k) from BAAS. EFFECT: technical result consists in improvement of quality of processing signals by reducing effect of interference and noise. 1 cl, 2 dwg РОССИЙСКАЯ ФЕДЕРАЦИЯ (19) RU (11) (51) МПК H03H 21/00 (13) 2 599 928 C2 (2006.01) ФЕДЕРАЛЬНАЯ СЛУЖБА ПО ИНТЕЛЛЕКТУАЛЬНОЙ СОБСТВЕННОСТИ (12) ОПИСАНИЕ (21)(22) Заявка: ИЗОБРЕТЕНИЯ К ПАТЕНТУ 2014147617/08, 26.11.2014 (24) Дата начала отсчета срока действия патента: 26.11.2014 (73) Патентообладатель(и): Тумачек Александр Сергеевич (RU) (43) Дата публикации заявки: 20.06.2016 Бюл. № 17 R U Приоритет(ы): (22) Дата подачи заявки: 26.11.2014 (72) Автор(ы): Тумачек Александр Сергеевич (RU), Незванов Александр Юрьевич (RU), Тарасов Геннадий Алексеевич (RU) (45 ...

Scale coefficient generation circuit

(57)【要約】本公報は電子出願前の出願データであるた め要約のデータは記録されません。

Device and method for detecting errors in adaptive filter

1. 청구범위에 기재된 발명이 속하는 기술분야 1. TECHNICAL FIELD OF THE INVENTION 본 발명은 적응필터의 에러 검출 장치 및 그 방법에 관한것임. The present invention relates to an error detection apparatus and method for an adaptive filter. 2. 발명이 해결하고자하는 과제 2. The problem to be solved by the invention 본 발명은 디지털 통신 등에서 입력신호에서 발생될 수 있는 에러값을 소정의 수학식을 이용하여 미리 검출해 메모리에 저장시켜, 필터링된 신호의 에러 검출 속도를 현저하게 향상시킬 수 있는 에러 검출 장치 및 그 방법을 제공하는데 그 목적이 있다. The present invention provides an error detection apparatus capable of significantly improving the error detection speed of a filtered signal by detecting an error value that may be generated in an input signal in a digital communication or the like by using a predetermined equation in advance and storing the result in a memory. The purpose is to provide a method. 3. 발명의 해결방법의 요지 3. Summary of Solution to Invention 본 발명은 다중 채널 간섭에 의해 왜곡된 신호성분을 필터링된 입력신호의 에러값을 검출하여 계수 갱신 정보로 이용하는 에러를 검출하는 적응필터에 있어서, 입력신호를 필터링하여 에러가 어느 정도 제거된 신호를 출력하는 필터부; 상기 필터부의 출력신호를 입력받아 양자화한 신호를 출력하는 슬라이스부; 외부로부터 소정의 기분값을 입력받아 상기 필터부의 출력신호와 상기 슬라이스부의 출력신호의 오차를 계산하는 오차 계산부; 상기 필터부의 출력신호에서 발생될 수 있는 에러값을 미리 계산하여 저장한 저장부; 상기 오차계산부의 출력신호를 이용하여 상기 저장수단의 어드레스 신호를 출력하는 어드레스 신호 발생부; 및 어드레스 신호에 따른 상기 저장수단의 출력을 입력받아 상기 필터부의 필터링 정도를 조정하는 계수 갱신부를 포함한다. The present invention provides an adaptive filter for detecting an error using a signal component distorted by multi-channel interference as an update coefficient by detecting an error value of a filtered input signal. An output filter unit; A slice unit configured to receive an output signal of the filter unit and output a quantized signal; An error calculator configured to receive a predetermined mood value from an external device and calculate an error between the output signal of the filter unit and the output signal of the slice unit; A storage unit which calculates and stores an error value that may be generated in the output signal of the filter unit; An address signal generator for outputting an address signal of the storage means by using the output signal of the ...

Signal processing circuit

本发明提供一种信号处理电路，包括具有通常等价的传递函数的IIR滤波器(11)、以及FIR滤波器(12)这两种滤波器，在调整成任意的传递函数的调整模式的情况下，将结构设定成IIR滤波器(11)，在调整结束后的阶段、或在信号处理模式的情况下，将结构变更成等价的传递函数的FIR滤波器(12)。

Systems and methods for calibrating power measurements in an electrosurgical generator

The disclosed electrosurgical systems and methods accurately determine the power actually applied to a load by using equalizers to calibrate the power measurements. The electrosurgical systems include an electro surgical generator and an electrosurgical instrument coupled to the electrosurgical generator through an electrosurgical cable. The electrosurgical generator includes an electrical energy source, voltage and current detectors, equalizers that estimate the voltage and current applied to a load, and a power calculation unit that calculates estimated power based upon the estimated voltage and current. The methods of calibrating an electro surgical generator involve applying a resistive element across output terminals of the electrosurgical generator, applying a test signal to the resistive element, measuring the magnitude and phase angle of voltage and current components of the test signal within the electrosurgical generator, estimating the magnitude and phase angle of the voltage and current at the resistive element using equalizers, and determining gain correction factors and minimum phase angles for the equalizers.

Filter bank approach to adaptive filtering methods using independent component analysis

본 발명은 독립 성분 분석을 이용한 여파기 적응 알고리즘의 필터 뱅크 접근 방법에 관한 것으로, 특히 2차와 그 이상의 고차 통계적 특성을 고려할 수 있는 독립 성분 분석을 여파기 적응 알고리즘에 적용하되 필터 뱅크를 도입하여 그 성능을 개선하는 기술에 관한 것이다. The present invention relates to a filter bank approach of the filter adaptation algorithm using independent component analysis. In particular, the independent bank analysis can be applied to the filter adaptation algorithm by considering independent second order analysis and higher order statistical characteristics. It is about technology to improve. 기존의 독립 성분 분석을 이용한 여파기 적응 알고리즘을 실제적인 문제에 적용하기 위해서는 매우 많은 개수의 학습 여파기 계수가 필요하며, 이들에 대하여 학습을 진행하는 경우에 있어서 매우 많은 계산량이 필요하고, 수렴 속도가 매우 느리며, 결과 신호의 품질도 떨어지게 된다. In order to apply the existing filter adaptive algorithm using independent component analysis to a practical problem, a very large number of learning filter coefficients are required, and in the case of learning about them, a large amount of computation is required and the convergence speed is very high. It is slow and the quality of the resulting signal is degraded. 이에, 본 발명은 기존의 독립 성분 분석을 이용한 여파기 적응 알고리즘과는 달리, 필터 뱅크를 이용함으로써 여파기 적응을 위한 계산량을 크게 줄일 수 있고, 적응 알고리즘의 수렴 속도와 결과 신호의 품질을 개선시키는 독립 성분 분석을 이용한 여파기 적응 알고리즘의 필터 뱅크 접근 방법이 제시된다. Accordingly, the present invention, unlike conventional filter adaptive algorithm using independent component analysis, can significantly reduce the calculation amount for filter adaptation by using a filter bank, and improve the convergence speed of the adaptive algorithm and the quality of the resulting signal. A filter bank approach of the filter adaptation algorithm using analysis is presented. 따라서, 본 발명에 의한 독립 성분 분석을 이용한 여파기 적응 알고리즘의 필터 뱅크 접근 방법을 제공하면 필터 뱅크를 이용하지 않는 방법에 비하여 계산량, 수렴 속도, 결과 신호의 품질 면에서 더욱 개선된 성능을 얻을 수 있다. Therefore, by providing a filter bank approach of the filter adaptive algorithm using independent component analysis according to the present invention, it is possible to obtain more improved performance in terms of calculation amount, convergence speed, and quality of the ...

Method and device for detecting signal of data storing device

PURPOSE: A method and device for detecting a signal of a data storing device is provided to detect the original signal of a data storing device using different adaptive equalization methods according to data patterns being reproduced from the data storing device. CONSTITUTION: A training sequence(t(k)), an input signal(a(k)), and an output signal(u(k)) of an adaptive partial response target equalizer(PREQ)(10) adapt the adaptive partial response target equalizer(PREQ)(10) to a channel, which minimizes effects of a non-linear noise and an undershoot, as the optimum state while passing a non-linear LMS(Least Mean Square) adaptation device(13). The input signal(a(k)) becomes the signal(u(k)) adapted to a Viterbi algorithm equalizer(VAEQ)(15) by passing the adaptive partial response target equalizer(PREQ)(10). The signal(u(k)) enters to the non-linear LMS(Least Mean Square) adaptation device(13) and is used for calculating an equalization error of the adaptive partial response target equalizer(PREQ)(10) with the training sequence(t(k)) and the input signal(a(k)). The non-linear LMS(Least Mean Square) adaptation device(13) adapts tap coefficients of the adaptive partial response target equalizer(PREQ)(10) using a conventional LMS or signed LMS method using the equalization error value.

Electronic filters, hearing aids and methods

An electronic filter for an electroacoustic system. The system has a microphone for generating an electrical output from external sounds and an electrically driven transducer for emitting sound. Some of the sound emitted by the transducer returns to the microphone means to add a feedback contribution to its electical output. The electronic filter includes a first circuit for electronic processing of the electrical output of the microphone to produce a filtered signal. An adaptive filter, interconnected with the first circuit, performs electronic processing of the filtered signal to produce an adaptive output to the first circuit to substantially offset the feedback contribution in the electrical output of the microphone, and the adaptive filter includes means for adapting only in response to polarities of signals supplied to and from the first circuit. Other electronic filters for hearing aids, public address systems and other electroacoustic systems, as well as such systems, and methods of operating them are also disclosed.

Method for controlling an adaption step size and hearing aid

Method and device for spreading spectrum

本申请实施例公开了一种扩频的方法及装置，该方法包括：确定数据帧的截止频率；根据截止频率确定低频带范围[f 1 ,f 2 ]和待扩充频带范围[f 3 ,f 4 ]；根据低频带范围确定目标滤波器，利用目标滤波器对数据帧进行滤波处理，得到第一低频信息；对第一低频信息作非线性处理，处理结果包含第二低频信息和高频信息；对处理结果和数据帧进行快速傅里叶变换FFT，分别得到第一FFT结果和第二FFT结果；对第二FFT结果在[f 3 ,f 4 ]内的信息，用第一FFT结果在[f 3 ,f 4 ]内的信息进行替换，得到扩频后的第二FFT结果。采用本申请，可自适应的对各种类型的音频序列进行扩频，得到良好的扩频效果。

Systems and methods for calibrating power measurements in an electrosurgical generator

The disclosed electrosurgical systems and methods accurately determine the power actually applied to a load by using equalizers to calibrate the power measurements. The electrosurgical systems include an electrosurgical generator and an electrosurgical instrument coupled to the electrosurgical generator through an electrosurgical cable. The electrosurgical generator includes an electrical energy source, voltage and current detectors, equalizers that estimate the voltage and current applied to a load, and a power calculation unit that calculates estimated power based upon the estimated voltage and current. The methods of calibrating an electrosurgical generator involve applying a resistive element across output terminals of the electrosurgical generator, applying a test signal to the resistive element, measuring the magnitude and phase angle of voltage and current components of the test signal within the electrosurgical generator, estimating the magnitude and phase angle of the voltage and current at the resistive element using equalizers, and determining gain correction factors and minimum phase angles for the equalizers.

Adaptive digital filter, FM receiver, signal processing method, and program

入力端子（３０１）には、１つの実信号から生成した互いに位相が９０度ずれた２つの信号の一方を実部に他方を虚部に持つ複素信号が入力される。フィルタ部は、この入力信号と実信号のフィルタ係数との畳み込み演算によって複素信号の出力信号を生成して出力端子（３０２）に出力する。共通部（３１８）と個別部（３１９0〜３１９N-1）から構成される係数制御部は、出力信号の包絡線の値が目標信号に近づくようにフィルタ係数を更新する。

Signal processing method, signal processing device, and signal processing program

本发明提供一种信号处理方法、信号处理装置及信号处理程序。在信号处理方法中，接收多个接收信号，通过从根据多个接收信号生成的多个回波中减去由以多个接收信号为输入的多个自适应滤波器生成的模拟回波来降低多个回波，该信号处理方法的特征在于，使多个接收信号中的2个以上的接收信号分别延迟，以生成延迟接收信号，将接收信号与延迟接收信号输入到自适应滤波器中，以生成模拟回波。

Adaptive balance filter guaranteeing optimal matching to line

The adaptive filter (5) is connected to the transmission path (7) via a shaping filter (6) and its output is subtracted (12) from that of a corresp. filter (13) in the reception path (14). The main filter (15) linking the two paths directly has its transfer function coeffts. loaded by a transfer device (17) from the coefft. output (8) of the adaptive filter. The transfer takes place only when a quality comparison (19) reveals the adaptive filter to have greater echo attenuation than the main filter.

Digital filter for non-stationary signals

FIELD: computing.SUBSTANCE: invention relates to the field of computing, mainly to digital processing of discrete signals. The essence of the invention lies in the fact that the extended Kalman filter is supplemented with an adaptive digital filter with the NLMS adaptation algorithm, which effectively operates under conditions of a non-stationary input signal and has optimal computational complexity. The extended Kalman filter, supplemented by an adaptive digital filter with the NLMS adaptation algorithm, is a digital filter for non-stationary signals, allows the use of a non-linear process / system model and compensates for the error in the extended Kalman filter, effectively operating in a non-stationary process.EFFECT: improved quality of estimation of non-stationary processes / systems by compensating for the Kalman filter operation error.1 cl, 3 dwg РОССИЙСКАЯ ФЕДЕРАЦИЯ (19) RU (11) (13) 2 747 199 C1 (51) МПК H03H 17/04 (2006.01) H03H 21/00 (2006.01) ФЕДЕРАЛЬНАЯ СЛУЖБА ПО ИНТЕЛЛЕКТУАЛЬНОЙ СОБСТВЕННОСТИ (12) ОПИСАНИЕ ИЗОБРЕТЕНИЯ К ПАТЕНТУ (52) СПК H03H 17/0216 (2020.08); H03H 21/0012 (2020.08); G06F 17/10 (2020.08) (21)(22) Заявка: 2020122203, 05.07.2020 (24) Дата начала отсчета срока действия патента: (73) Патентообладатель(и): Федеральное государственное бюджетное образовательное учреждение высшего образования. "Юго-Западный государственный университет" (ЮЗГУ) (RU) Дата регистрации: 29.04.2021 (56) Список документов, цитированных в отчете о поиске: RU 2110883 C1, 10.05.1998. RU 2012997 C1, 15.05.1994. RU 98122956 A, 20.10.2000. SU 964980 A1, 07.10.1982. SU 93001060 A, 20.02.1996. SU 1800588 A1, 07.03.1993. SU 926762 A1, 07.05.1982. SU 1739483 A1, 07.06.1992. RU 194496 U1, 12.12.2019. US 7203233 B2, 10.04.2007. US 8078659 B2, 13.12.2011. US 8385864 B2, 26.02.2013. (45) Опубликовано: 29.04.2021 Бюл. № 13 2 7 4 7 1 9 9 R U (54) Цифровой фильтр для нестационарных сигналов (57) Реферат: Изобретение относится к области вычислительной техники, преимущественно к ...

Electronic filters, repeated signal charge conversion apparatus, hearing aids and methods

An electronic filter for filtering an electrical signal. Signal processing circuitry therein includes a logarithmic filter having a series of filter stages with inputs and outputs in cascade and respective circuits associated with the filter stages for storing electrical representations of filter parameters. The filter stages include circuits for respectively adding the electrical representations of the filter parameters to the electrical signal to be filtered thereby producing a set of filter sum signals. At least one of the filter stages includes circuitry for producing a filter signal in substantially logarithmic form at its output by combining a filter sum signal for that filter stage with a signal from an output of another filter stage. The signal processing circuitry produces an intermediate output signal, and a multiplexer connected to the signal processing circuit multiplexes the intermediate output signal with the electrical signal to be filtered so that the logarithmic filter operates as both a logarithmic prefilter and a logarithmic postfilter. Other electronic filters, signal conversion apparatus, electroacoustic systems, hearing aids and methods are also disclosed.

Apparatus and method for estimation of transmitted signals in receiver in operation of digital signal transmission

在数字信息传输系统中,接收机(14)接收一个信号(R(T)),系统的信号频带宽超出系统符号速率。相关和采样电路(15)接收一基频带信号(Y(T)),每个符号时间(TS)将信号采样信号(Y(K/8)),相关(23),产生信道估计(HF)并减频采样此信号(Y(K/8))构成被观察信号(Y(K/2))。此信号在前置滤波器(26)中滤波器(26)中滤波,并以符号速率(27,TS/1)采样按维持比算法产生估计符号(S(K))。

Method and apparatus for blind separation of delayed and filtered sources

소스 신호들의 성분들을 포함하고 있는 혼합 신호들로부터 소스 신호들을 복구시키는 방법, 채널 1과 2의 혼합 신호들은 채널 1과 2의 적응 가중치(adaptive weights)에 의해 곱해지게 되어 채널 1과 2의 곱 신호들(product signals)을 만든다. 채널 1의 필터된 피드백(feedback) 신호는 채널 2의 근사 신호(approximation signal)로부터 발생된다. 그리고 채널 2의 필터된 피드백 신호는 채널 1의 근사신호로부터 발생된다. 채널 1의 필터된 피드백 신호는 채널 1의 근사 신호를 발생시키기 위해 채널 1의 곱 신호에 더해지며, 채널 2의 필터된 피드백 신호는 채널2의 근사 신호를 발생시키기 위해 채널 2의 곱신호에 더해진다. 비선형 함수가 출력 신호들을 만들기 위해 상기 근사 신호들에 인가되어 진다. 적응 가중치들과 필터된 피드백 신호들은 출력 신호들의 엔트로피를 최대화시키기 위해 조정이 되어 진다.

Complex filter

The complex filter for simplifying the exiting complex filter by using an improved complex filter algorithm, includes the 1st filter multiplying a real part signal by a coefficient, the 2nd filter and the 3rd filter multiplying real and complex part signals by a coefficient, the 1st adder adding the 1st filtered signal to the 2nd filtered signal to generate a real part signal, the 2nd adder adding the 1st filtered signal to the 3rd filtered signal to generate an imaginary part signal. The filter uses logic "Zr+(Yr+Yi)Cr-Yi(Cr+Ci), Zi=(Yr+Yi)Cr-Yr(Cr-Ci)" to compose 3N multipliers and 3N adders, in comparison with the existing 4N multipliers and 4N-2 adders.

Transmission system, terminal unit, encoder, decoder and adaptive filter

FIELD: communications. SUBSTANCE: device has encoder, which performs signal encoding, transmitter, receiver and decoder. Decoder is equipped with prediction adaptive filter which is used for encoding of voice or sound signals in order to decrease bit transmission rate. Filter weights for prediction filter (10A, 10B) are determined by recursive calculation using Levinson-Derbin recursion for fixed-point arithmetic operations. EFFECT: simplified design, decreased cost of calculations without loss in precision. 10 cl, 4 dwg ссср ПЧ Го РОССИЙСКОЕ АГЕНТСТВО ПО ПАТЕНТАМ И ТОВАРНЫМ ЗНАКАМ (19) ВИ” 2 123 728 ' 13) СЛ 5 МК 40 9/14 12) ОПИСАНИЕ ИЗОБРЕТЕНИЯ К ПАТЕНТУ РОССИЙСКОЙ ФЕДЕРАЦИИ (21), (22) Заявка: 95105531109, 02.05.1994 (30) Приоритет: 05.05.1993 ОЕ Р4314921.9 (46) Дата публикации: 20.12.1998 (56) Ссылки: ЗЧ 8387064, 15.06.81. КЦ 2012997, 15.05.94. 5Ц 915095 А, 23.03.82. 4$ 5025404 А, 18.06.91. ЕР 044174595 А, 25.09.91. ЕР 0078581 А, 11.05.83. (71) Заявитель: Филипс Электроникс Н.В. (МГ) (72) Изобретатель: Рудольф Хофманн (0Е) (73) Патентообладатель: Филипс Электроникс Н.В. (М) (54) СИСТЕМА ПЕРЕДАЧИ, ТЕРМИНАЛЬНЫЙ БЛОК, КОДИРУЮЩЕЕ УСТРОЙСТВО, ДЕКОДИРУЮЩЕЕ УСТРОЙСТВО И АДАПТИВНЫЙ ФИЛЬТР (57) Реферат: Система передачи, содержащая кодирующее устройство, предназначенное для кодирования сигнала, передающее устройство, приемное устройство и декодирующее устройство. Для декодирующего устройства предусмотрен адаптивный фильтр прогнозирования, посредством которого речевые или звуковые сигналы кодируются при снижении их скоростей передачи битов. Коэффициенты фильтра для адаптивного фильтра прогнозирования (10А, 10В) определяются рекурсивным образом в соответствии с рекурсией Левинсона-Дербина при арифметических операциях над числами с фиксированной запятой. — Техническим результатом заявленной группы изобретений является снижение объема аппаратных средств и стоимости расчетов в системе передачи такого типа без снижения при этом точности расчета. 5 с. и 5 з.п. ф-лы, 4 ил. ГЕ 101 ...

Compensation Filter

오디오 시스템을 위한 프리필터(5)는 룸(2)내의 확성기(1)를 구비하여 구성되고, 이는 선형위상 보정필터응답에 의해 확성기(1)에 기인하는 진폭 및 위상에러를 보정함과 더불어 룸(2)의 진폭응답을 보정하는 반면, 최소 위상보정필터단을 채용함으로써 최소 가능 여분 위상왜곡을 도입한다. 테스트신호 발생기(8)는 주파수 반복주기 보다 더 큰 위상 반복주기와 함께 스위프되는 주기적인 주파수로 이루어진 신호를 발생시킨다. 룸(2)의 다양한 장소에 위치한 마이크로폰(7)은 룸(2)과 확성기(1)에 의해 처리된 오디오신호를 측정하고, 계수계산기(6; 예컨대, 디지탈 신호 처리장치)는 룸에 응답하는 신호를 추출함으로써 필요한 최소 위상보정이 확성기(1)에 대해 미리 계산된 선형위상보정과 종속되어진다. 필터(5)는 계수계산기(6)와 동일한 디지탈 신호처리기를 구비하여 구성된다. 고충실도의 오디오 재생과, 카스테레오 재생에 적용된다. The prefilter 5 for the audio system comprises a loudspeaker 1 in the room 2, which compensates for amplitude and phase errors due to the loudspeaker 1 by means of a linear phase correction filter response. While the amplitude response of (2) is corrected, the minimum possible extra phase distortion is introduced by employing the minimum phase correction filter stage. The test signal generator 8 generates a signal consisting of a periodic frequency swept with a phase repetition period greater than the frequency repetition period. Microphones 7 located at various places in the room 2 measure the audio signals processed by the room 2 and the loudspeaker 1, and the counter 6 (for example, a digital signal processor) responds to the room. The minimum phase correction required by extracting the signal is subject to the linear phase correction previously calculated for the loudspeaker 1. The filter 5 is provided with the same digital signal processor as the coefficient calculator 6. It is applied to high fidelity audio reproduction and car stereo reproduction.

Digital filter

본 발명은 필터 연산에 의한 출력 지연 없이 필터의 크기를 줄이면서 4탭 실수 필터와 1탭 복소수 필터를 하나의 필터 구조로 결합시킨 디지털 필터에 관한 것이다. 특히 본 발명은 4탭 실수 필터와 1탭 복소수 필터를 결합한 후 모드 선택 신호에 따라 실수 필터와 복소수 필터 동작을 결정하고, 연산 선택 신호에 따라 한 단위 시간 주기 안에 필터의 출력을 얻으면서, 하나의 연산자가 한 심볼 클럭(한 단위 시간 주기)안에 두 번의 연산을 수행하게 한다. 이로 인해 실수 필터와 복소수 필터가 한 심볼 클럭 동안 하나의 연산자로 한 번의 연산을 수행하는 종래의 필터에 비교할 때 본 발명에서는 사용되는 승산기와 가산기의 숫자를 크게 줄일 수 있다. 4탭 실수 필터, 1탭 복소수 필터

Unknown system identification method and apparatus using band division adaptive filter

Digital Noise Filter

이 발명은 디지털 노이즈 필터에 관한 것으로, 디지털 신호를 입력받아, 업 또는 카운팅을 하기 위한 카운팅 수단과; 상기 카운팅 수단의 출력신호를 입력받아, 계수치가 일정치에 도달하면 출력을 내보내기 위한 출력 판단수단과; 상기 출력 판단수단내의 일부분 출력을 입력받고, 디지털입력신호와 클럭신호를 입력받아 상기 카운팅 수단의 카운팅동작을 개시 또는 중단하기 위한 상태천이 제어수단을 포함하여 구성되어, 적은 수의 플립플롭으로 폭이 긴 노이즈라도 쉽게 제거하며, 적은 비용으로 제작할 수 있는 잇점이 있는 디지털 노이즈 필터에 관한 것이다.

Equalization filter generator and method for operating the same

The present invention relates to an equalization filter generator and a method of operating the same. To this end, the equalization filter which generator applies taps to a partial interval equalizer for placing an interval as a portion of a symbol time in a blind communication system and generates an equalization filter used for the blind equalization of a signal, includes: a matrix generation part for receiving a signal transmitted through one or more channels in the blind communication system and generating a signal matrix for the signal; a singular value decomposition part for performing singular value decomposition on the signal matrix; and an equalization filter generation part for generating an equalization filter for the signal based on the result of the singular value decomposition. A channel matrix (H) constituting the signal matrix is a full rank tall matrix having a size of L(N+1) x (N+K+1). It is possible to improve the equalization performance of the equalizer.

A kind of active complex filter based on neutral net

本发明属于模拟集成电路技术领域，特别涉及一种基于神经网络的有源复数滤波器，包括有源复数滤波器模块、自动修正控制模块和自动调整电路。本发明利用神经网络在不同温度和工艺角下对模数转换器ADC转换过的滤波器输出信号进行学习，得到可以调节滤波单元的网络权重值，之后遇到相同温度和工艺角时以实现对滤波单元的调节，稳定输出信号。通过人工神经网络产生非线性控制电压，自动修正输出电压，减小因工艺容差和温度漂移等产生的偏差，锁定滤波器的输出响应。本发明结构简单，容易实现，有效降低了滤波器的复杂度，不会引入新的误差。且修正范围广，精度高，具有通用性，能用于不同滤波器的修正。

Adaptive digital filter, fm receiver, signal processing method, and record medium readable by computer recorded program thereof

입력단자(301)에는, 1개의 실신호로부터 생성되고 위상이 서로 90도 벗어난 2개의 신호 중 일방을 실수부로, 타방을 허수로 가지는 복소신호가 입력된다. 필터부는, 그 입력신호와 실신호의 필터계수와의 컨볼루션연산에 의해 복소신호의 출력신호를 생성하여 출력단자(302)에 출력한다. 공통부(318)와 개별부(319 0 ~319 N-1 )로 구성된 계수제어부는, 출력신호의 포락선의 값이 목표신호에 근접하도록 필터계수를 갱신한다. 복소신호,포락선,필터계수,적응 디지털 필터 The input terminal 301 is inputted with a complex signal having one real signal and one imaginary one of two signals generated from one real signal and having 90 degrees out of phase with each other. The filter unit generates an output signal of a complex signal by convolution operation between the input signal and the filter coefficient of the real signal and outputs the output signal to the output terminal 302. The coefficient control unit composed of the common unit 318 and the individual units 319 0 to 319 N-1 updates the filter coefficient so that the value of the envelope of the output signal approaches the target signal. Complex signal, envelope, filter coefficient, adaptive digital filter

Hearing aid having compensation for acoustic feedback

A hearing aid (figs. 1,3) includes a filter (27) in an electrical feedback path (k-m), the characteristics of which filter (27) are calculated to model acoustic coupling between the receiver (11) and microphone (5) of the aid. A limiter (15) is inserted in the main electrical pathway (b-c-d-e-f-g) between the microphone (5) and the receiver (11) to provide stability in the presence of sudden sound bursts. A noise signal (N) is injected continuously into the electrical circuit and is used to adapt the characteristics of the filter (27) to accommodate changes in the acoustic coupling. The level of the noise signal (N) can be varied to match changes in residual signal level to maintain signal to noise ratio and to optimise the rate of adaption commensurate with satisfactory hearing function whilst the noise itself is unobtrusive to the user.

Signal detection method and apparatus of data storage device

본 발명은 데이타 저장기기의 신호검출방법 및 장치에 관한 것으로서, 입력신호 a[k]를 원하는 형태의 채널로 변환해 주는 적응 계수 FIR 필터링과정; 훈련신호 t[k+1], t[k-1], t[k]를 비선형 특성이 포함된 신호 f[k]를 소정의 식에 의해 비선형신호로 만들어 주는 비선형 첨가과정; 비선형신호 f[k]를 원하는 채널과 콘볼루션시켜서 기준신호 d[k]를 생성하는 과정; 기준신호 d[k]로부터 출력신호 x[k]를 감산하여 오류를 출력하는 과정; 및 오류 e[k]를 이용하여 최소평균자승 기법에 의해 적응형 부분응답 목표등화기의 탭계수를 등화하는 계수 적응과정을 구비한다. 따라서, 저장기기로부터 재생되는 데이타 패턴에 따라 PREQ의 계수 적응기법을 비선형 조합에 의해 실시함으로써 적응 등화기의 계수설정에 방해가 되는 비선형 잡음의 영향을 최소화할 수 있으므로 비선형 왜곡이 심하고 ISI가 많이 존재하는 신호를 신뢰적으로 검출할 수 있다.

Noise cancellation method and noise cancellation device

Method and system for filtering nonstationary and non-Gaussian noise in PLC communication signal

本发明公开了一种PLC通信信号中非平稳非高斯噪声的滤除方法及系统。所述滤除方法包括：获取PLC通信系统内的实测PLC通信信号序列以及多尺度分解层数；根据多尺度分解层数以及多层低通滤波器截止频率确定每一层滤波后的PLC通信信号序列；根据当前层滤波后的PLC通信信号序列确定当前层时间窗口长度；构建解析序列；确定解析序列在所述当前层时间窗口长度下的伪Wigner‑Ville分布；根据伪Wigner‑Ville分布恢复每一层所述滤波后的PLC通信信号序列，确定恢复PLC通信信号序列；从所述多尺度分解层数的最后一层开始，根据当前层的所述恢复PLC通信信号序列以及上一层的插值后PLC通信信号序列滤除所述PLC通信系统内的非平稳非高斯噪声。本发明能够有效滤除PLC通信系统中非平稳非高斯噪声。

Device for separation of signals

一种用来从多信号的混合信号中分离出至少两个信号的方法,该方法包括一个基于一个判据的、为估算分离结构中的参数而对其求极小值的修改判据和一个调整修正项,其特征在于该修正项包括一个含有待估算的参数的第一向量;一个含有与真参数有关的可能信息和一权重矩阵的第二向量;和一个所述第一向量和第二向量之间的差、所述权重矩阵与所述第一向量和第二向量之差的共轭转置三者的乘积。

Low power consumption adaptive equalizing filter for communication

(57)【要約】 【課題】本発明は、通信用装置において、送信波形が線 路特性等により変形させられたことを、等化する必要が ある場合に於いて、特に、消費電力を低減できる適応等 化トランスバーサルフィルタを実現する。 【解決手段】通信用装置において、送信波形が線路特性 等により変形させられたことを等化するためのトランス バーサルフィルタを有し、トランスバーサルフィルタの 出力は、判定器に接続し、期待値を出力する。また、差 分器を備え、入力はトランスバーサルフィルタの出力と 期待値が入力され、当該信号の誤差成分が出力される。 この、誤差信号は、各次の係数を決める回路に入力され る。各次の係数を決める回路は、レジスタと乗算器と差 分器にて構成され、レジスタの出力から、前記誤差信号 と定数αとの乗算結果を減算した結果が、レジスタの入 力および各次の乗算係数となる。この時、前記誤差の大 きさに応じて、各次の乗算係数の更新演算を制御する機 構を有している。

Selective active cancellation system for repetitive phenomena

Apparatus for improving reproduction performance by adjusting filter coefficients of equalizer and method thereof

본 발명에는 등화기의 필터 계수를 조절하여 재생 성능을 높이는 장치 및 방법이 개시되어 있다. 본 발명은 비터비 검출기의 전단에 구성되는 디지털 필터로 구성되는 등화기의 출력값 중 비터비 검출기에서 사용되는 기준 레벨에 대응하는 레벨을 검출하여 검출된 레벨과 미리 설정된 기준값과의 레벨 에러를 검출하고, 레벨 에러가 최소가 되도록 등화기의 필터 계수를 조정함으로써 비터비 검출기의 데이터 검출 성능을 높인다.

Transmission system, terminal unit, coder, decoder, and adaptive filter

FIELD: electronic engineering. SUBSTANCE: transmission system has signal coder, transmitter, receiver, and decoder. The latter is provided with adaptive prediction filter used to encode voice or sound signals in case their bit transmission speed reduces. Factors for adaptive prediction filter are determined by recursive way according to Levinson-Derbin recursion using fixed-point arithmetic. EFFECT: improved precision of calculations when coding in transmission system without increasing amount of calculations. 11 cl ООС ПЧ ГЭ (19) РОССИЙСКОЕ АГЕНТСТВО ПО ПАТЕНТАМ И ТОВАРНЫМ ЗНАКАМ ВИ "” 2138 030 ' (51) МПК 13) Сл СО 9/14 12) ОПИСАНИЕ ИЗОБРЕТЕНИЯ К ПАТЕНТУ РОССИЙСКОЙ ФЕДЕРАЦИИ (21), (22) Заявка: 95105532/09, 02.05.1994 (24) Дата начала действия патента: 02.05.1994 (30) Приоритет: 05.05.1993 ОЕ Р 4314921.9 (46) Дата публикации: 20.09.1999 (56) Ссылки: Проект спецификации С 728 с фиксированной запятой, документ АМ.93-Д.З, Исследовательская группа ХУ МККТТ /Международный консультативный комитет по телеграфии и телефонии/, Лондон, 29-30 марта 1993 г. 4$ 5142656 А, 25.08.92. 4$ 4520491 А, 28.05.85. 4$ 5025404 А, 18.06.91. $Ц 1305884 АЛ, 23.04.87. 5Ц 1367164 АЛ, 15.01.88. 54 1374243 А\Л, 15.02.88. (85) Дата перевода заявки РСТ на национальную фазу: 04.01.95 (86) Заявка РСТ: 1В 94/00087 (02.05.94) (87) Публикация РСТ: М/О 94/25961 (10.11.94) (98) Адрес для переписки: 129010, Москва, Большая Спасская ул., 25, стр.3, Союзпатент Патентному поверенному Емельянову Е.И. (71) Заявитель: Филипс Электроникс Н.В. (МГ) (72) Изобретатель: Рудольф Хофманн (0Е) (73) Патентообладатель: Филипс Электроникс Н.В. (М) (54) СИСТЕМА ПЕРЕДАЧИ, ТЕРМИНАЛЬНЫЙ БЛОК, КОДИРУЮЩЕЕ УСТРОЙСТВО, ДЕКОДИРУЮЩЕЕ УСТРОЙСТВО И АДАПТИВНЫЙ ФИЛЬТР (57) Реферат: Сущность изобретения: система передачи содержит кодирующее устройство, предназначенное для кодирования сигналов, передающее устройство, приемное устройство и декодирующее устройство. Для декодирующего устройства предусмотрен адаптивный фильтр прогнозирования, ...

Method and apparatus for detecting and correcting misconvergence of a blind equalizer

Convergence of blind fractionally spaced equalizers is improved, and misconvergence is corrected by training the equalizers to detect convergence of one adaptive filter, copying the tap weights of the converged adaptive filter to the other adaptive filters and shifting the tap weights of the other adaptive filters according to the expected phase difference between the respective filters. In a two-dimensional orthogonal modulation scheme the converged weights of a first filter are copied to a second filter and shifted π/2. For the two dimensional orthogonal modulation scheme, the probability of a proper convergence can be increased by choosing initial tap weights for the two adaptive filters with a 3π/4 phase difference.

Method and system for on-line blind source separation

A method and apparatus is disclosed for performing blind source separation using convolutive signal decorrelation. For a first embodiment, the method accumulates a length of input signal (mixed signal) that comprises a plurality of independent signals from independent signal sources. The invention then divides the length of input signal into a plurality of T-length periods (windows) and performs a discrete Fourier transform (DFT) on the signal within each T-length period. Thereafter, estimated cross-correlation values are computed using a plurality of the averaged DFT values. A total number of K cross-correlation values are computed, where each of the K values is averaged over N of the T-length periods. Using the cross-correlation values, a gradient descent process computes the coefficients of an FIR filter that will effectively separate the source signals within the input signal. A second embodiment of the invention is directed to on-line processing of the input signal - i.e., processing the signal as soon as it arrives with no storage of the signal data. In particular, an on-line gradient algorithm is provided for application to non-stationary signals and having an adaptive step size in the frequency domain based on second derivatives of the cost function. The on-line separation methodology of this embodiment is characterized as multiple adaptive decorrelation.

Digital zooming system

Median filter

본 발명은 중앙값 필터에 관한 것으로, 종래의 중앙값 필터는 그 중앙값을 선택하기 위해 출력하기 위해 개의 비교기와 의 비교기 출력의 조합으로 중앙값 선택회로가 구성된다. 그러나 이때

Adaptive audio systems and sound reproduction systems

An algorithm is described for sparsely updating the filter coefficients of an adaptive filter of an adaptive audio system to facilitate a high sampling rate without excessive processing capacity. Computer simulations are used to illustrate the potential for using an inverse filter matrix to manipulate the position of stereophonic images. The possibilities suggested could be used in conjunction with existing stereophonic recordings in order to improve stereophonic imaging in situations where the relative locations of loudspeakers and listener are constrained to be non-ideal. In one arrangement, real sound sources (S1, S2) provide to the listener the impression that the sources are at more widely-spaced positions (V1, V2). In another arrangement, employing four sound sources (S1, S2, S3, S4) the impression is created at two listener locations (M1, M2; M3, M4) that the sound is coming from a pair of sound sources positioned symmetrically of those listener locations.

Transmission system including at least a coder

本发明的传输系统包括至少一个对信号(10a)进行编码的编码器(101)。该系统还包括至少一个发射机(102)、至少一个接收机(103)和至少一个译码器(104)。至少为编码器(101)提供了至少一个自适应预测滤波器(10A，1B)，利用该自适应预测滤波器编码语音或音频信号，同时减小了它们的位速率。根据“列文森-德宾递推”以定点算法递推地确定自适应预测滤波器(10A，10B)的滤波系数。

Scale factor generation circuit

Method and device for performing an approximate arithmetical division

A method and a device for performing an approximate division of a constant number by a variable number in binary form. The variable number is presumed to consist of a character bit and a plurality of bits which state the absolute value of the number. The number is converted by first forming a digital word, by substituting with logic zeroes any logic ones that have a lower significance than the most significant logic one of the bits. There is then formed a new number in binary form, by reading the character bit of the variable number as a character bit and by reading the bits in the digital word in a reversed order. The device may mainly comprise a single gate network.

Dual-processing interference cancelling system and method

A dual-processing interference cancelling system and method for processing a broadband input in a computationally efficient manner. Dual processing divides the input into higher and lower frequency bands and applies adaptive filter processing to the lower frequency band while applying non-adaptive filter processing to the higher frequency band. Various embodiments are shown including those based on sub-bands, broadband processing with band-limited adaptation, and broadband processing with an external main-channel generator.

Apparatus for improving reproduction performance by adjusting filter coefficients of equalizer and method thereof

본 발명에는 등화기의 필터 계수를 조절하여 재생 성능을 높이는 장치 및 방법이 개시되어 있다. 본 발명은 비터비 검출기의 전단에 구성되는 디지털 필터로 구성되는 등화기의 출력값 중 비터비 검출기에서 사용되는 기준 레벨에 대응하는 레벨을 검출하여 검출된 레벨과 미리 설정된 기준값과의 레벨 에러를 검출하고, 레벨 에러가 최소가 되도록 등화기의 필터 계수를 조정함으로써 비터비 검출기의 데이터 검출 성능을 높인다. The present invention discloses an apparatus and method for adjusting the filter coefficients of an equalizer to enhance regeneration performance. The present invention detects a level error between a detected level and a predetermined reference value by detecting a level corresponding to a reference level used in the Viterbi detector among the output values of an equalizer composed of a digital filter configured in front of the Viterbi detector. In addition, the data detection performance of the Viterbi detector is improved by adjusting the filter coefficients of the equalizer so that the level error is minimized.

Method and device for extracting a useful signal of finite spatial extension at each instant and variable with time.

L'invention concerne un Procédé d'extraction d'un signal utile d'extension spatiale finie et variable avec le temps par un réseau de N capteurs, N étant plus grand ou égal à 3, recevant ce signal utile additionné de q bruits additifs cohérents spatialement, q étant inférieur à N, comprenant les étapes suivantes: - une étape (42) d'acquisition des signaux bruts sur la sortie de chaque capteur; - une étape (43) de filtrage passe-bande desdits signaux pour se restreindre à la bande de fréquence des signaux utiles; - une étape (44) de numérisation desdits signaux filtrés; - une étape (46) de calcul des signaux d'erreur de prédiction spatiale du bruit; - une étape d'analyse des signaux d'erreur de prédiction de manière à réaliser la détection du signal utile et sa séparation des q bruits additifs. L'invention concerne également un dispositif d'extraction d'un signal utile. Application notamment au domaine de la détection magnétique.