2258 results about "Adaptive filter" patented technology

A method of filtering a speech signal is presented. The method involves providing a filter (404) suited for reduction of distortion caused by speech coding, estimating acoustic noise in the speech signal, adapting the filter in response to the estimated acoustic noise to obtain an adapted filter, and applying the adapted filter to the speech signal so as to reduce acoustic noise and distortion caused by speech coding in the speech signal.

The method, program and information processing system involves filtering and securing data (security sensitive words-characters-data objects) in a source document. The adaptive filter uses a compilation of additional data (typically networked) and identifies the sensitive words / objects in the compilation of additional data, retrieves contextual, semiotic and taxonomic words / objects from the additional data compilation that are related to the sensitive words / objects. A filter is compiled with the retrieved data and the filter is used to extract sensitive words / objects and the retrieved data (words / objects) from the source document to obtain extracted data and remainder data therefrom. Contextual words related to the security sensitive words / objects are obtained based upon statistical analysis of the additional data compilation. Semiotic words related to the security words are synonyms, antonyms, and pseudonyms, syntactics relative to the target words and retrieved words, and pragmatics relative to the sensitive words and retrieved words as reflected in the compilation of additional data.

A method and apparatus is provided for decoding an encoded baseline video stream and an enhancement stream. The baseline video stream is decoded, upscaled and enhanced by applying adaptive filters specified by the enhancement stream. Baseline upscaled images are then coded to motion compensate enhanced high resolution images using previously decoded enhanced images, thus recycling these enhanced images. The enhancement stream provides the best predictor method for the decoder to combine blocks from previous enhanced images and upscaled images to produce a motion compensated enhanced image. Likewise, forward and backward motion compensated images are blended according to feature classification and filter extraction methods provided by the enhancement stream to produce a bidirectionally predicted frame. Lastly, the decoder applies residual data from the enhancement stream to produce a completed enhanced image.

Apparatus includes generator generating reference signal based on noise emitted from sound source, detector detecting level of reference signal and change in level, unit comparing change with threshold-value range and produce compared result, filter filtering reference signal, adaptive filter having variable filter coefficient, unit updating filter coefficient according to change of level of reference signal for obtaining an updated filter coefficient, unit stopping updating of filter coefficient in response to compared result when change falls outside threshold-value range, unit storing updated filter coefficient each time filter coefficient is updated, generator generating control signal using stored filter coefficient, unit generating control sound based on control signal, microphone detecting synthesis sound pressure of control sound and noise to produce an error signal, and unit setting stored filter coefficient to more accurate coefficient than stored filter coefficient based on error signal, and signal acquired by filtering control signal through filter.

A system for reducing signal disturbances in ECG, which disturbances are caused by cardio-pulmonary resuscitation (CPR). The system includes a measuring device for measuring one or more signals derived from parameters such as compression depth, lung inflation etc. as a result of CPR. The parameter signals form one or more reference signals that correlate with the signal disturbances. One or more adaptive filters filter the signals out from the signal that constitutes the ECG signals so as to remove disturbances caused by CPR.

The invention provides a multi-sensor system that uses an algorithm based on adaptive filtering to monitor a patient's respiratory rate. The system features a first sensor selected from the following group: i) an impedance pneumography sensor featuring at least two electrodes and a processing circuit configured to measure an impedance pneumography signal; ii) an ECG sensor featuring at least two electrodes and an ECG processing circuit configured to measure an ECG signal; and iii) a PPG sensor featuring a light source, photodetector, and PPG processing circuit configured to measure a PPG signal. Each of these sensors measures a time-dependent signal which is sensitive to respiratory rate and, during operation, is processed to determine an initial respiratory rate value. An adaptive digital filter is determined from the initial respiratory rate. The system features a second sensor (e.g. a digital 3-axis accelerometer) that attaches to the patient's torso and measures an ACC signal indicating movement of the chest or abdomen that is also sensitive to respiratory rate. This second signal is processed with the adaptive filter to determine a final value for respiratory rate.

A six state Kalman filter is adapted based on a current acceleration mode of an INS device. Gyro measurements are used to determine the acceleration mode and the Kalman filter estimates bias and small angle error of the measurements based on the acceleration mode. The bias error corrects the gyro measurement and the small angle error is used along with the corrected gyro measurement to update an attitude sensed by the gyro.

Ringing artifacts are removed from a quantized image by an image de-ringing filter that includes a determination unit, an adaptive filter and a nonlinear low-pass filter. The determination unit determines whether each selected pixel of a first set of selected pixels of an image contains a ringing artifact based on, for example, gray-level values of selected pixels within a determination kernel of pixels relating to the selected pixel. The adaptive filter generates a filtered gray-level value for each pixel determined by the determination unit to contain a ringing artifact based on, for example, gray-level values of selected pixels within a filtering kernel of pixels relating to the pixel. The nonlinear low-pass filter generates a low-pass-filtered gray-level value for each selected pixel of a second set of selected pixels of the image.

An active noise cancellation system includes an adaptive filter, a signal source, an acoustic actuator, a microphone, a secondary path and an estimation unit. The adaptive filter receives a reference signal representing noise, and provides a compensation signal in response to the received reference signal. The signal source provides a measurement signal. The acoustic actuator radiates the compensation signal and the measurement signal to the listening position. The microphone receives a first signal that is a superposition of the radiated compensation signal, the radiated measurement signal, and the noise signal at the listening position, and provides a microphone signal in response to the received first signal. The secondary path includes a secondary path system that represents a signal transmission path between an output of the adaptive filter and an output of the microphone. The estimation unit estimates a transfer characteristic of the secondary path system in response to the measurement signal and the microphone signal.

The present invention relates to a blind adaptive filter for narrowband interference cancellation, which includes an adaptive filter, a delay unit coupled to the adaptive filter for generating a delayed signal with a predetermined delay length from the output signal of the adaptive filter, and an error calculation unit coupled to the adaptive filter and the delay unit. The error calculation unit compares the output signal from the adaptive filter and the delayed signal from the delay unit to extract error information, and feedback the first error information to the adaptive filter. The first error information is formed of a transfer function including a number of coefficients, and used to adjust the adaptive filter and remove interference in the next input signal. The disclosed technique is also applicable in wideband receivers, as well as resisting multiple strong narrowband interferences having a frequency sweep rate of tens of milliseconds.

The invention provides a multi-sensor system that uses an algorithm based on adaptive filtering to monitor a patient's respiratory rate. The system features a first sensor selected from the following group: i) an impedance pneumography sensor featuring at least two electrodes and a processing circuit configured to measure an impedance pneumography signal; ii) an ECG sensor featuring at least two electrodes and an ECG processing circuit configured to measure an ECG signal; and iii) a PPG sensor featuring a light source, photodetector, and PPG processing circuit configured to measure a PPG signal. Each of these sensors measures a time-dependent signal which is sensitive to respiratory rate and, during operation, is processed to determine an initial respiratory rate value. An adaptive digital filter is determined from the initial respiratory rate. The system features a second sensor (e.g. a digital 3-axis accelerometer) that attaches to the patient's torso and measures an ACC signal indicating movement of the chest or abdomen that is also sensitive to respiratory rate. This second signal is processed with the adaptive filter to determine a final value for respiratory rate.

The present invention teaches a new audio coding system that can code both general audio and speech signals well at low bit rates. A proposed audio coding system comprises a linear prediction unit for filtering an input signal based on an adaptive filter; a transformation unit for transforming a frame of the filtered input signal into a transform domain; a quantization unit for quantizing a transform domain signal; a long term prediction unit for determining an estimation of the frame of the filtered input signal based on a reconstruction of a previous segment of the filtered input signal; and a transform domain signal combination unit for combining, in the transform domain, the long term prediction estimation and the transformed input signal to generate the transform domain signal.

A cabin communication system for improving clarity of a voice spoken within an interior cabin having ambient noise includes a microphone for receiving the spoken voice and the ambient noise and for converting the spoken voice and the ambient noise into an audio signal, the audio signal having a first component corresponding to the spoken voice and a second component corresponding to the ambient noise, a speech enhancement filter for removing the second component from the audio signal to provide a filtered audio signal, the speech enhancement filter removing the second component by processing the audio signal by a method taking into account elements of psycho-acoustics of a human ear, and a loudspeaker for outputting a clarified voice in response to the filtered audio signal.

To achieve vibration-dependent noise reduction by the use of a microphone to pick up an audio signal and a vibration sensor. The microphone apparatus according to a preferred embodiment of the present invention is one having one or more microphone, one or more sensor, noise extraction means for extracting the noise bandwidth section from the output signal of the sensor, an adaptive filter coordinated with the microphone for receiving the output signal of the noise extraction means as the reference input signal, and an adder for subtracting the output signal of the adaptive filter from the output signal of the microphone, wherein the vibration detection directions of the microphone and the sensor match, and so do the output polarities of the vibration signal.

One embodiment of the present invention relates to an adaptive filtering apparatus comprising first and second real valued adaptive filters, respectively configured to receive an adaptive filter input signal based upon a transmission signal in a transmission path. The first real valued adaptive filter is configured to operate a real valued adaptive filter algorithm on the input signal to estimate a first intermodulation noise component (e.g., an in-phase component) in a desired signal and to cancel the estimated noise. The second real valued adaptive filter is configured to operate a real valued adaptive filter algorithm on the input signal to estimate a second intermodulation noise component (e.g., a quadrature phase component) in the desired signal and to cancel the estimated noise. Accordingly, each filter operates a real valued adaptive algorithm to cancel a noise component, thereby removing complex cross terms between the components from the adaptive filtering process.

An active noise cancellation system reduces, at a listening position, the power of a noise signal being radiated from a noise source to the listening position. The system includes an adaptive filter that receives a reference signal representing the noise signal, and provides a compensation signal. A bass management unit receives the compensation signal and applies a phase shift to the compensation signal to provide a phase shifted compensation signal. A first acoustic radiator receives the phase shifted compensation signal and radiates audio indicative thereof to the listening position. A second acoustic radiator receives the compensation signal and radiates audio indicative thereof to the listening position. The transfer function characteristics from the input of the bass management system to the listening position approximately matches a desired transfer function.

An adaptive filter is used to filter a target version of a visual image that is produced by processing an original version of the visual image, the characteristics of the adaptive filter being determined in accordance with one or more characteristics of the original version of the visual image. The orientation and / or strength of filtering of the adaptive filter can be adjusted based on local properties of the original image, which can enable the adaptive filter to avoid introducing blurring across true edges in an image.

Adaptive noise control for reducing power of an acoustic noise signal radiated from a noise source to a listening position comprises providing an electrical reference signal correlated with the acoustic noise signal; filtering the electrical reference signal with an adaptive filter to provide an electrical output signal; multiplying the electrical output signal of the adaptive filter by a gain factor to provide a first electrical compensation signal; filtering and multiplying the electrical output signal of the adaptive filter by the inverse of the gain factor to provide a second electrical compensation signal, the second gain factor being equal to 1 subtracted by the first gain factor; radiating the first electrical compensation signal to the listening position with an acoustic transducer; sensing a residual electrical error signal at the listening position; adding the second electrical compensation signal to the electrical error signal to provide a compensated error signal; and adapting filter coefficients of the adaptive filter as a function of the compensated error signal and the reference signal.

A adaptive filter interpolation method and system for the demosaicing of color images. In general, input pixels are input in a Bayer-mosaiced pattern (only one color per pixel), and output pixels are in full RGB mode (three color values per pixel). For each pixel location, in raster scan order, the processing steps can be summarized as follows. Following a regular raster scanning order (from left to right and top to bottom), for each pixel location horizontal and vertical gradients are first computed (whose computation depends on the available color for that pixel), and from those the appropriate interpolation filters are chosen from a small set of predetermined filters. Then, the chosen filters are applied to interpolate the missing data.

An earpiece for providing an audio signal to two ears of a user, includes an audio interface for providing the audio signal, a bone conduction speaker coupled to the audio interface for outputting an output signal, an adaptive filter with an adjustable impulse response to provide a filtered signal that once played to the second ear of a user, by an in-ear speaker, reduces an audio effect of the leakage signal on the second ear of the user, an in-ear speaker coupled to the audio interface for providing an in-ear audio signal to the second ear of the user, in response to the filtered signal and the second input signal, an in-ear microphone for sensing a sensed audio signal, and a processor for updating an impulse response of the adaptive filter in order to reduce a contribution of the leakage signal to the sensed audio signal.

An audio system (100) is provided with improved adaptive filter (206) to automatically adjust signal gain depending on the ambient noise level. The original music signal passes through a normalized adaptive filter (206), and is subtracted from the ambient room signal detected by a microphone (120), resulting in an error signal that is an estimate of the ambient noise. The error signal is used to update a set of adaptation coefficients so that the normalized adaptive filter more accurately simulates the room transfer function, resulting in an better estimate of the ambient noise. The audio system (100) is calibrated automatically upon initial use to determine adaptation coefficients and noise threshold level to prevent runaway gain. System parameters are adjusted using a controller (124) with a user-friendly interface (400).

An echo cancellation device (ECD) comprises an echo canceller (EC) including a transfer function estimator (EST, H) and a subtractor (ADD) and a residual echo suppression device (G, ADD2). The residual echo suppression device (G) comprises a residual echo filter (G) having an adjustable filter function (g). This filter function (g) can be adapted to either remove from the subtractor output (TNE') the spectral characteristics relating to the reception signal (RFE) and / or to emphasize in the subtractor output signal (TNE') a background signal spectral content relating to the transmission signal (TNE). A noise generation means (NGM') can be provided at the output of the adaptable filter (G) for injecting a noise process in to the filter output signal (TNE') prior to a speech coding in a speech coder (COD). The noise process masks in the filter output signal a spectral content relating to the reception signal (RFE).

A receiver determines a symbol synch time for recovering data from a symbol of signal samples generated in accordance with Orthogonal Frequency Division Multiplexing. Each symbol includes a guard period which carries data repeated from a data bearing part of the symbol and pilot signal samples. The receiver comprises a pilot assisted tracker which is operable to determine an adjustment to the symbol synch time from a pilot assisted channel impulse response estimate, a guard adapted filter processor comprising a filter and a filter controller operable to adapt the impulse response of the filter to the signal samples from the guard period. The controller is operable to excite the filter with the symbol signal samples to generate an output signal which provides a further representation of the channel impulse response. A symbol time adjustment estimator is operable to adjust the symbol synch time in accordance with the adjustment provided by at least one of the pilot assisted tracker and the guard adapted filter processor. The receiver provides an improved estimate of the symbol synch time by combining a pilot assisted tracker with a guard adapted filter processor. The pilot assisted tracker estimates the symbol synch time from a channel impulse response estimate generated from pilot signal samples. By combining the synch time adjustment estimated from the pilot assisted channel impulse response, with the adjustment estimated by the guard adapted filter processor, an ambiguity in a relative time of arrival of signal paths of the channel impulse response with respect to the main signal path is obviated.

In order to overcome problems when using an adaptive filter for cancellation of common-mode noise in digital subscriber loops, caused by a portion of the differential signal being converted to common mode, which degrades the performance of the filter, a noise cancellation technique is proposed which compensates for this cross-coupled signal energy. In particular, a digital noise detector is used to detect one or more noisy frequency bands of the common mode signal and pass only the digitized common mode signal in those detected frequency bands through the adaptive filter to produce a digital common mode noise estimate signal. A control unit adjusts coefficients of the adaptive filter to reduce correlation between the differential signal and common mode signal. It is also proposed to compensate for the effects of stray capacitive coupling across the usual hybrid device by including an equivalent capacitive component in a common mode noise estimation circuit.

Techniques for estimating adaptive noise canceling (ANC) performance in a personal audio device, such as a wireless telephone, provide robustness of operation by triggering corrective action when ANC performance is low, and / or by saving a state of the ANC system when ANC performance is high. An anti-noise signal is generated from a reference microphone signal and is provided to an output transducer along with program audio. A measure of ANC gain is determined by computing a ratio of a first indication of magnitude of an error microphone signal that provides a measure of the ambient sounds and program audio heard by the listener including the effects of the anti-noise, to a second indication of magnitude of the error microphone signal without the effects of the anti-noise. The ratio can be determined for different frequency bands in order to determine whether particular adaptive filters are trained properly.

The invention provides a multi-sensor system that uses an algorithm based on adaptive filtering to monitor a patient's respiratory rate. The system features a first sensor selected from the following group: i) an impedance pneumography sensor featuring at least two electrodes and a processing circuit configured to measure an impedance pneumography signal; ii) an ECG sensor featuring at least two electrodes and an ECG processing circuit configured to measure an ECG signal; and iii) a PPG sensor featuring a light source, photodetector, and PPG processing circuit configured to measure a PPG signal. Each of these sensors measures a time-dependent signal which is sensitive to respiratory rate and, during operation, is processed to determine an initial respiratory rate value. An adaptive digital filter is determined from the initial respiratory rate. The system features a second sensor (e.g. a digital 3-axis accelerometer) that attaches to the patient's torso and measures an ACC signal indicating movement of the chest or abdomen that is also sensitive to respiratory rate. This second signal is processed with the adaptive filter to determine a final value for respiratory rate.

From a mixed signal in which a first signal and a second signal are mixed, the second signal is removed at low processing cost and without delay. As a result, an estimated first signal which has low residue of the second signal and low distortion is obtained.An estimated first signal is generated by subtracting a pseudo second signal which is estimated to be mixed in a first mixed signal in which a first signal and a second signal are mixed from the first mixed signal. The pseudo second signal is obtained by a first adaptive filter using a second mixed signal in which the first signal and the second signal are mixed in a different proportion from the first mixed signal. A coefficient update amount of the first adaptive filter is made smaller as compared with a case when the estimated first signal is smaller than the first mixed signal, in case the estimated first signal is larger than the first mixed signal.

A personal audio device, such as a wireless telephone, includes an adaptive noise canceling (ANC) circuit that adaptively generates an anti-noise signal from a reference microphone signal that measures the ambient audio and an error microphone signal that measures the output of an output transducer plus any ambient audio at that location and injects the anti-noise signal at the transducer output to cause cancellation of ambient audio sounds. A processing circuit uses the reference and error microphone to generate the anti-noise signal, which can be generated by an adaptive filter operating at a multiple of the ANC coefficient update rate. Downlink audio can be combined with the high data rate anti-noise signal by interpolation. High-pass filters in the control paths reduce DC offset in the ANC circuits, and ANC coefficient adaptation can be halted when downlink audio is not detected.

An apparatus, method, and system for providing dc offset reduction in a communications channel include two or more feedback loops to generate dc offset correction signals, which in turn are combined with an input analog signal and a processed digital signal thereby reducing dc offset. Each feedback loop may include an adaptive filter. At least one feedback loop may be responsive to an error signal that represents the difference between the delayed input of a first detector, and its output. Further, the dc offset correction signal, partially delayed, may be added to the error signal, thereby improving the response time of the dc offset correction loop.