53 results about "Linear predictive analysis" patented technology

For determining a long-term-prediction delay parameter characterizing a long term prediction in a technique using signal modification for digitally encoding a sound signal, the sound signal is divided into a series of successive frames, a feature of the sound signal is located in a previous frame, a corresponding feature of the sound signal is located in a current frame, and the long-term-prediction delay parameter is determined for the current frame while mapping, with the long term prediction, the signal feature of the previous frame with the corresponding signal feature of the current frame. In a signal modification method for implementation into a technique for digitally encoding a sound signal, the sound signal is divided into a series of successive frames, each frame of the sound signal is partitioned into a plurality of signal segments, and at least a part of the signal segments of the frame are warped while constraining the warped signal segments inside the frame. For searching pitch pulses in a sound signal, a residual signal is produced by filtering the sound signal through a linear prediction analysis filter, a weighted sound signal is produced by processing the sound signal through a weighting filter, the weighted sound signal being indicative of signal periodicity, a synthesized weighted sound signal is produced by filtering a synthesized speech signal produced during a last subframe of a previous frame of the sound signal through the weighting filter, a last pitch pulse of the sound signal of the previous frame is located from the residual signal, a pitch pulse prototype of given length is extracted around the position of the last pitch pulse of the sound signal of the previous frame using the synthesized weighted sound signal, and the pitch pulses are located in a current frame using the pitch pulse prototype.

Provided is a highband coding apparatus and method for a wideband coding system. The coding apparatus and method can reduce a pre-echo phenomenon by encoding the highband based on lowband encoding information and Temporal Noise Shaping technique. A highband encoding apparatus includes: a domain converter for converting the domain of an input highband signal into a frequency domain; a linear prediction order determiner for determining a linear prediction order based on the lowband encoding information; a linear prediction analyzer for analyzing a highband signal of the frequency domain based on the determined linear prediction order to thereby generate a linear prediction coefficient; a linear prediction coefficient quantizer for quantizing the linear prediction coefficient based on the lowband encoding information; and a residual signal quantizer for obtaining a residual signal by dequantizing the quantized linear prediction coefficient and quantizing the residual signal.

An audio encoding apparatus capable of improving a frame cancellation error tolerance without increasing a number of bits of a fixed codebook in a CELP type audio encoding. A linear prediction analyzer analyzes an input digital speech signal and outputs linear predictive coefficients. A linear predictive coefficients quantizer quantizes the linear predictive coefficients. A low-frequency-band component encoder encodes a down-sampled linear-predictive residual signal by a pulse-code-modulation encoder and generates low-frequency-band component encoded information, while a high-frequency-band component encoder encodes an error signal between a linear-predictive residual signal and an up-sampled signal of a decoded down-sampled linear-predictive residual signal by a code-excited-linear-prediction encoder and generates high-frequency-band component encoded information.

To provide a three-dimensional acoustic effect to a listener in a reproduction field, via a headphone in particular, a three-dimensional acoustic apparatus is formed by a linear synthesis filter having filter coefficients that are the linear predictive coefficients obtained by performing a linear predictive analysis on an impulse response which represents the acoustic characteristics to be added to the original signal to achieve this effect. By passing the signal through this acoustic characteristics adding filter, the desired acoustic characteristics are added to the original signal, and by dividing the power spectrum of the impulse response of these acoustic characteristics into critical bandwidths and performing this linear predictive analysis based on impulse signal determined based from power spectrum signals representing the signal sound of each of these critical bandwidths, the filter coefficients of the linear synthesis filter are determined.

To provide a three-dimensional acoustic effect to a listener in a reproduction field, via a headphone in particular, a three-dimensional acoustic apparatus is formed by a linear synthesis filter having filter coefficients that are the linear predictive coefficients obtained by performing a linear predictive analysis on an impulse response which represents the acoustic characteristics to be added to the original signal to achieve this effect. By passing the signal through this acoustic characteristics adding filter, the desired acoustic characteristics are added to the original signal, and by dividing the power spectrum of the impulse response of these acoustic characteristics into critical bandwidths and performing this linear predictive analysis based on impulse signal determined based from power spectrum signals representing the signal sound of each of these critical bandwidths, the filter coefficients of the linear synthesis filter are determined.

A system and method facilitating signal enhancement utilizing an adaptive filter is provided. The invention includes an adaptive filter that filters an input based upon a plurality of adaptive coefficients and modifies the adaptive coefficients based on a feedback output. A feedback component provides the feedback output based, at least in part, upon a non-linear function of the acoustic reverberation reduced output. Optionally, the system can further include a linear prediction (LP) analyzer and / or a LP synthesis filter. The system can enhance signal(s), for example, to improve the quality of speech that is acquired by a microphone by reducing reverberation. The system utilizes, at least in part, the principle that certain characteristics of reverberated speech are measurably different from corresponding characteristics of clean speech. The system can employ a filter technology (e.g., reverberation reducing) based on a non-linear function, for example, the kurtosis metric.

A speech synthesis method subjects a reference speech signal to windowing to extract a speech pitch wave having a window function of a window length double a pitch period of the reference speech signal from the reference speech signal. A linear prediction coefficient is generated by subjecting the reference speech signal to a linear prediction analysis. The speech pitch wave is subjected to inverse-filtering based on the linear prediction coefficient to produce a residual pitch wave, which is then stored as information of a speech synthesis unit in a voiced period in a storage. Speech using the information of the speech synthesis unit is then synthesized.

For determining a long-term-prediction delay parameter characterizing a long term prediction in a technique using signal modification for digitally encoding a sound signal, the sound signal is divided into a series of successive frames, a feature of the sound signal is located in a previous frame, a corresponding feature of the sound signal is located in a current frame, and the long-term-prediction delay parameter is determined for the current frame while mapping, with the long term prediction, the signal feature of the previous frame with the corresponding signal feature of the current frame. In a signal modification method for implementation into a technique for digitally encoding a sound signal, the sound signal is divided into a series of successive frames, each frame of the sound signal is partitioned into a plurality of signal segments, and at least a part of the signal segments of the frame are warped while constraining the warped signal segments inside the frame. For searching pitch pulses in a sound signal, a residual signal is produced by filtering the sound signal through a linear prediction analysis filter, a weighted sound signal is produced by processing the sound signal through a weighting filter, the weighted sound signal being indicative of signal periodicity, a synthesized weighted sound signal is produced by filtering a synthesized speech signal produced during a last subframe of a previous frame of the sound signal through the weighting filter, a last pitch pulse of the sound signal of the previous frame is located from the residual signal, a pitch pulse prototype of given length is extracted around the position of the last pitch pulse of the sound signal of the previous frame using the synthesized weighted sound signal, and the pitch pulses are located in a current frame using the pitch pulse prototype.

Alternate window optimization procedures and / or LSP interpolation factor optimization procedures are used to improve the ITU-T G.729 speech coding standard (the “Standard”) by replacing the window used by the Standard with an optimized window and / or replacing the LSP interpolation factor used by the standard with an optimized LSP interpolation factor. Optimized windows created using the alternate window optimization procedure and / or optimized LSP interpolation factors created using the LSP interpolation factor optimization procedure yield improvements in the objective quality of synthesized speech produced by the Standard. In many cases, improvements are obtained using shorter windows, which results in reduced computational cost and / or smaller future buffering requirements, which results in lowered coding delay. The improved Standard, procedures, and optimized windows and LSP interpolation factors can all be implemented as computer readable software code and in optimization devices.

A formant tracking apparatus and a formant tracking method are provided. The formant tracking apparatus includes: a framing unit dividing an input voice signal into a plurality of frames; a linear prediction analyzing unit obtaining linear prediction coefficients for each frame; a segmentation unit segmenting each of the linear prediction coefficients into a plurality of segments; a formant candidate determining unit obtaining formant candidates by using the linear prediction coefficients, and summing the formant candidates for each segment to determine formant candidates for each segment; a formant number determining unit determining a number of tracking formants for each segment among the formant candidates satisfying a predetermined condition; and a tracking unit searching the tracking formants as many as the number of the tracking formants determined in the formant number determining unit among the formant candidates belonging to each segment.

A linear prediction coefficient of a signal represented in a frequency domain is obtained by performing linear prediction analysis in a frequency direction by using a covariance method or an autocorrelation method. After the filter strength of the obtained linear prediction coefficient is adjusted, filtering may be performed in the frequency direction on the signal by using the adjusted coefficient, whereby the temporal envelope of the signal is shaped. This reduces the occurrence of pre-echo and post-echo and improves the subjective quality of the decoded signal, without significantly increasing the bit rate in a bandwidth extension technique in the frequency domain represented by spectral band replication.

A part of the image inputted (S1) from a fingerprint sensor is segmented (S3), and the fundamental frequency is computed by second-order linear prediction analysis (S5). The sum (S) of the fundamental frequencies in the x and y directions is computed and used as a judgment measure of the texture degree (S7). A predetermined threshold and the measure S determined in S7 are compared, and the degree of skin texture is evaluated to one of three levels (S9 to S17). The ratio between the fundamental frequencies in the x and y directions of the object image is computed and used as an evaluation measure of the texture flow (S19). The image is rotated, and a similar evaluation measure is determined. Based on these measures, the texture flow is judged at one of two levels (S31 to S35). The results of the judgment on the texture degree and the texture flow are displayed on a display screen (S37).

The invention is directed to a single channel mask estimation method capable of improving reverberant speech identification for CI users. The method is based on the energy of the reverberant signal and the residual signal computed from linear prediction (LP) analysis. The mask is estimated by comparing the energy ratio of the two signals at different frequency bins with an adaptive threshold. As the threshold is updated for each frame of speech based on the energy ratios of the reverberant and LP residual signals computed from previous frames, it is amenable for real-time implementation. It can thus be used as a specialized (for reverberant environments) sound coding strategy used for cochlear implant applications.

An apparatus and method for encoding / decoding a speech signal which determines a variable bit rate based on reserved bits obtained from a target bit rate, is provided. The variable bit rate is determined based on a source feature of the speech signal and the reserved bits is obtained based on the target bit rate. The apparatus for encoding the speech signal may include a linear predictive (LP) analysis unit / quantization unit to determine an immittance spectral frequencies (ISF) index, a closed loop pitch search unit, a fixed codebook search unit, a gain vector quantization (VQ) unit to determine a gain vector quantization (VQ) index, and a bit rate control unit to control at least two indexes of the ISF index, the pitch index, the code index, and the gain VQ index to be encoded to be variable bit rates based on a source feature of a speech signal and the reserved bits.

A method and apparatus for a voice transcoder that converts a bitstream representing frames of data encoded according to a first voice compression standard to a bitstream representing frames of data according to a second voice compression standard using perceptual weighting that uses tuned weighting factors, such that the bitstream of a second voice compression standard to produce a higher quality decoded voice signal than a comparable tandem transcoding solution. The method includes pre-computing weighting factors for a perceptual weighting filter optimized to a specific source and destination codec pair, pre-configuring the transcoding strategies, mapping CELP parameters in the CELP parameter space according to the selected coding strategy, performing Linear Prediction analysis if specified by the transcoding strategy, perceptually weighting the speech using with tuned weighting factors, and searching for adaptive codebook and fixed-codebook parameters to obtain a quantized set of destination codec parameters.