34results about How to "Low computing complexity" patented technology

An apparatus and method for lossless run-length data encoding is disclosed. In the first stage of the method, an input data sequence is transformed without increasing its volume in order to obtain long sequences of identical digits. In the second stage, every such sequence is replaced with a unique shorter sequence. The compressed data is decoded performing corresponding inverse operations. The method is very efficient for compression of some classes of digital images with long sequences of the same number, such as graphics, texts, signatures, and fingerprints.

An active stream is selected from one of a plurality of audio streams generated in a common acoustic environment by obtaining, for each stream obtaining, at a series of measurement instants tn, where n=1 . . . N, a final performance metric that is representative of the stream's goodness for representing near-end speech at each measurement instant tn. The final performance metrics for each stream are accumulated to determine an overall performance score. A switch to a new stream as the active stream occurs when the stream with the best overall performance score exceeds the overall performance score of the currently active stream by a threshold amount.

A method of reducing noise in a speech signal involves converting the speech signal to the frequency domain using a fast fourier transform (FFT), creating a subset of selected spectral subbands, determining the appropriate gain for each subband, and interpolating the gains to match the number of FFT points. The converted speech signal is then filtered using the interpolated gains as filter coefficients, and an inverse FFT performed on the processed signal to recover the time domain output signal.

A method of detecting double talk condition in hands free communication devices is disclosed. In general, the method in accordance with the teachings of this invention detects double talk conditions based on inherent frequency response differences between the transducers used and acoustical effect on the spectrum of the returned echo signal. An input signal from a far-end talker and an input signal from the output from an echo canceler are received by the detector. K spectral subbands are created for each input signal. From this K subbands q subbands are selected based on inherent frequency differences between the far-end transducer and a near-end transducer. The spectral echo residual power is estimated at each subband. The estimated spectral echo power and the output signal from the echo canceler for a selected subband are compared to a predetermined threshold. Based on this comparison, it is determined whether double talk conditions exist based on the comparison.

Provided are an LDPC decoding apparatus and method using a sequential decoding algorithm having a partial group, capable of reducing the number of an iterative decoding by more than half without degrading the performance and increasing an amount of computation. The LDPC decoding method includes the steps of: receiving a prior probability information (channel values) based on information on channel values associated with distance between symbol signals in constellation related to the received noise and LDPC encoded data, and initializing bit nodes; dividing check nodes into partial groups before updating check node information based on the prior probability information, and performing a decoding by applying a sequential decoding algorithm; determining whether a parity check equations are satisfied; and outputting decoded messages obtained when satisfying the parity check equation or after terminating an iterative processor by a termination algorithm.

A 3D-HEVC inter-frame information hiding method based on visual perception includes steps of information embedding and information extraction. In the step of information embedding, the human visual perception characteristic is considered, stereo salient images are obtained by a stereo image salient model, and the stereo salient images are divided into salient blocks and non-salient blocks with an otsu threshold. The coding quantization parameters are modified according to different modulation rules for different regions. Then, based on the modified quantization parameters, the coding-tree-units are coded to complete the information embedding. In the step of information extraction, no original video is needed, no any side information needs to be transmitted, and the secret information can be blindly extracted. The present invention combines with the human visual perception characteristic, and selects P frames and B frames as embedded frames for effectively reducing the decrease of the stereo video subjective quality.

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.

A method for processing bit symbols generated by a data source, in particular a video, still image or audio source, comprising the following steps of constructing a plurality of bit-planes from the data source, each bit-plane comprising a plurality of bit-plane symbols; scanning the bit-plane symbols of each bit-plane to generate a binary string of bit-plane symbols; and encoding the binary string of the bit-plane symbols using a statistical model, wherein the statistical model is based on statistical properties of a Laplacian probability distribution function which characterizes the data source.

Measurements of body sounds in the ear canal may be used for many applications. However, reliability is dependent on the ear canal being properly sealed to allow the body sounds to achieve a detectable level. An apparatus is therefore provided for determining a seal quality indication for a seal of an ear canal. An ear canal microphone (201) provides a microphone signal to an input (203) which is coupled to a circuit (205) for generating a first signal from the microphone signal. The first signal may be the same as the microphone signal. A circuit (209) then determines the seal quality in response to the frequency spectrum for the first signal. A frequency transformer (207) may perform a frequency transformation on the first signal to generate a frequency spectrum for the first signal and. The seal quality indication may specifically be generated based on a detection of a low frequency boost.

An apparatus and a method are provided for detecting N number of TX signals in a MIMO wireless communication system. The apparatus includes an RF processor, a channel estimator, and a signal detector. The RF processor converts signals, received through multiple antennas, into baseband signals. The channel estimator estimates channel information of the respective antennas by using the received signals. The signal detector arranges the baseband signals of the respective antennas on the basis of the channel information, calculates a threshold value of each stage, and selects symbols with a cumulative metric smaller than or equal to the threshold value as candidates at each stage, to detect a TX signal vector with N number of symbols.