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936 results about "Nonlinear distortion" patented technology

Nonlinear distortion is a term used (in fields such as electronics, audio and telecommunications) to describe the phenomenon of a non-linear relationship between the "input" and "output" signals of - for example - an electronic device.

System and methods for recognizing sound and music signals in high noise and distortion

A method for recognizing an audio sample locates an audio file that most closely matches the audio sample from a database indexing a large set of original recordings. Each indexed audio file is represented in the database index by a set of landmark timepoints and associated fingerprints. Landmarks occur at reproducible locations within the file, while fingerprints represent features of the signal at or near the landmark timepoints. To perform recognition, landmarks and fingerprints are computed for the unknown sample and used to retrieve matching fingerprints from the database. For each file containing matching fingerprints, the landmarks are compared with landmarks of the sample at which the same fingerprints were computed. If a large number of corresponding landmarks are linearly related, i.e., if equivalent fingerprints of the sample and retrieved file have the same time evolution, then the file is identified with the sample. The method can be used for any type of sound or music, and is particularly effective for audio signals subject to linear and nonlinear distortion such as background noise, compression artifacts, or transmission dropouts. The sample can be identified in a time proportional to the logarithm of the number of entries in the database; given sufficient computational power, recognition can be performed in nearly real time as the sound is being sampled.
Owner:APPLE INC

System and methods for recognizing sound and music signals in high noise and distortion

A method for recognizing an audio sample locates an audio file that most closely matches the audio sample from a database indexing a large set of original recordings. Each indexed audio file is represented in the database index by a set of landmark timepoints and associated fingerprints. Landmarks occur at reproducible locations within the file, while fingerprints represent features of the signal at or near the landmark timepoints. To perform recognition, landmarks and fingerprints are computed for the unknown sample and used to retrieve matching fingerprints from the database. For each file containing matching fingerprints, the landmarks are compared with landmarks of the sample at which the same fingerprints were computed. If a large number of corresponding landmarks are linearly related, i.e., if equivalent fingerprints of the sample and retrieved file have the same time evolution, then the file is identified with the sample. The method can be used for any type of sound or music, and is particularly effective for audio signals subject to linear and nonlinear distortion such as background noise, compression artifacts, or transmission dropouts. The sample can be identified in a time proportional to the logarithm of the number of entries in the database; given sufficient computational power, recognition can be performed in nearly real time as the sound is being sampled.
Owner:APPLE INC

Sampled amplitude read channel employing an adaptive non-linear correction circuit for correcting non-linear distortions in a read signal

A sampled amplitude read channel is disclosed for magnetic disk storage systems comprising an adaptive non-linear correction circuit for correcting non-linear distortions in the read signal, such as asymmetry caused by the non-linear response of a magneto-resistive (MR) read head. The analog read signal is sampled and the discrete time sample values equalized into a desired partial response prior to sequence detection. The non-linear correction circuit is inserted into the read path prior to the sequence detector and adaptively tuned by a least-mean-square (LMS) adaptation circuit. In one embodiment, the non-linear correction circuit is a discrete-time Volterra filter comprising a linear response for implementing an equalizing filter, and a non-linear response for attenuating non-linear distortions in the read signal. The filter coefficients of both the linear and non-linear sections of the Volterra filter are adaptively adjusted by the LMS adaptation circuit. In an alternative embodiment, the non-linear correction circuit operates in the analog domain, prior to the sampling device, where the cost and complexity can be minimized. The analog correction circuit implements an inverse response to that of the non-linearity in the read signal, and the response is adaptively tuned using an LMS update value computed in discrete-time for a Volterra filter, without actually implementing a Volterra filter. Further, the LMS update value for the analog correction circuit can be implemented using a simple squaring circuit.
Owner:CIRRUS LOGIC INC

Compensator for removing nonlinear distortion

The present invention is a computationally-efficient compensator for removing nonlinear distortion. The compensator operates in a digital post-compensation configuration for linearization of devices or systems such as analog-to-digital converters and RF receiver electronics. The compensator also operates in a digital pre-compensation configuration for linearization of devices or systems such as digital-to-analog converters, RF power amplifiers, and RF transmitter electronics. The compensator effectively removes nonlinear distortion in these systems in a computationally efficient hardware or software implementation by using one or more factored multi-rate Volterra filters. Volterra filters are efficiently factored into parallel FIR filters and only the filters with energy above a prescribed threshold are actually implemented, which significantly reduces the complexity while still providing accurate results. For extremely wideband applications, the multi-rate Volterra filters are implemented in a demultiplexed polyphase configuration which performs the filtering in parallel at a significantly reduced data rate. The compensator is calibrated with an algorithm that iteratively subtracts an error signal to converge to an effective compensation signal. The algorithm is repeated for a multiplicity of calibration signals, and the results are used with harmonic probing to accurately estimate the Volterra filter kernels. The compensator improves linearization processing performance while significantly reducing the computational complexity compared to a traditional nonlinear compensator.
Owner:LINEARITY LLC
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