574 results about "Linear prediction" patented technology

The present invention describes a novel methodology for adjusting a bandwidth extension algorithm by adapting one or more of enhancing perception parameters (e.g., a signal level, a signal energy and/or a gain) of a high-band encoded signal based on the characteristics of the input signal and an encoding performance in a low band with a codec utilizing audio-band-split coding by separate encoders and decoders for each audio band. The adaptation is based on the low-band coding algorithm. It can be at least two types of such an algorithm: e.g., an algebraic code excitation linear prediction (ACELP) algorithm for a speech-like input signal and a transform algorithm of a non-speech-like input signal, such that when the ACELP coding is selected, the corresponding enhancing perception parameter is gradually tuned down and when the encoding algorithm is changed to the transform coding, the corresponding enhancing perception parameter is gradually tuned up.

The present invention relates to a method and device for improving concealment of frame erasure caused by frames of an encoded sound signal erased during transmission from an encoder (106) to a decoder (110), and for accelerating recovery of the decoder after non erased frames of the encoded sound signal have been received. For that purpose, concealment/recovery parameters are determined in the encoder or decoder. When determined in the encoder (106), the concealment/recovery parameters are transmitted to the decoder (110). In the decoder, erasure frame concealment and decoder recovery is conducted in response to the concealment/recovery parameters. The concealment/recovery parameters may be selected from the group consisting of: a signal classification parameter, an energy information parameter and a phase information parameter. The determination of the concealment/recovery parameters comprises classifying the successive frames of the encoded sound signal as unvoiced, unvoiced transition, voiced transition, voiced, or onset, and this classification is determined on the basis of at least a part of the following parameters: a normalized correlation parameter, a spectral tilt parameter, a signal-to-noise ratio parameter, a pitch stability parameter, a relative frame energy parameter, and a zero crossing parameter.

An imaging system incorporating adaptive compression which includes determining linear predictive differential residuals from an imager array pixel row. The differential residuals are classified into categories, each category having a range of differential residuals associated with it. The categories are analyzed to produce an ordered list having categories with most to least frequent residuals falling within a respective residual range associated with a respective category. The ordered list is then used to select a variable length encoding table with a matching ordered list. Variable length encoded category and range position offset data is output to a serializer unit, where the range position offset refers to a position in a range associated with a particular category.

An audio signal decoder includes a transform domain path configured to obtain a time-domain representation of a portion of an audio content on the basis of a first set of spectral coefficients, a representation of an aliasing-cancellation stimulus signal and a plurality of linear-prediction-domain parameters. The transform domain path applies a spectrum shaping to the first set of spectral coefficients to obtain a spectrally-shaped version thereof. The transform domain path obtains a time-domain representation of the audio content on the basis of the spectrally-shaped version of the first set of spectral coefficients. The transform domain path includes an aliasing-cancellation stimulus filter to filter the aliasing-cancellation stimulus signal in dependence on at least a subset of the linear-prediction-domain parameters. The transform domain path also includes a combiner configured to combine the time-domain representation of the audio content with an aliasing-cancellation synthesis signal to obtain an aliasing reduced time-domain signal.

An audio decoder for providing a decoded representation of an audio content on the basis of an encoded representation of the audio content comprises a linear-prediction-domain decoder core configured to provide a time-domain representation of an audio frame on the basis of a set of linear-prediction domain parameters associated with the audio frame and a frequency-domain decoder core configured to provide a time-domain representation of an audio frame on the basis of a set of frequency-domain parameters, taking into account a transform window out of a set comprising a plurality of different transform windows. The audio decoder comprises a signal combiner configured to overlap-and-add-time-domain representations of subsequent audio frames encoded in different domains, in order to smoothen a transition between the time-domain representations of the subsequent frames. The set of transform windows comprises one or more windows specifically adapted for a transition between a frequency-domain core mode and a linear-prediction-domain core mode.

A system and method is provided that employs a frequency domain interpolative CODEC system for low bit rate coding of speech which comprises a linear prediction (LP) front end adapted to process an input signal that provides LP parameters which are quantized and encoded over predetermined intervals and used to compute a LP residual signal. An open loop pitch estimator adapted to process the LP residual signal, a pitch quantizer, and a pitch interpolator and provide a pitch contour within the predetermined intervals is also provided. Also provided is a signal processor responsive to the LP residual signal and the pitch contour and adapted to perform the following: provide a voicing measure, where the voicing measure characterizes a degree of voicing of the input speech signal and is derived from several input parameters that are correlated to degrees of periodicity of the signal over the predetermined intervals; extract a prototype waveform (PW) from the LP residual and the open loop pitch contour for a number of equal sub-intervals within the predetermined intervals; normalize the PW by a gain value of the PW; encode a magnitude of the PW; and separate stationary and nonstationary components of the PW using a low complexity alignment process and a filtering process that introduce no delay. The ratio of the energy of the nonstationary component of the PW to that of the stationary component of the PW is averaged across 5 subbands to compute the nonstationarity measure as a frequency dependent vector entity. A measure of the degree of voicing of the residual is also computed using openloop pitchgain, pitch variance, relative signal power, PW correlation and PW nonstationarity in low frequency subbands. The nonstationarity measure and voicing measure are encoded using a 6-bit spectrally weighted vector quantization scheme using a codebook partitioned based on a voiced/unvoiced decision. At the decoder, a stationary component of PW is reconstructed as a weighted combination of the previous PW phase vector, a random phase perturbation and a fixed phase vector obtained from a voiced pitch pulse.

The gain smoothing method and device modify the amplitude of an innovative codevector in relation to background noise present in a previously sampled wideband signal. The gain smoothing device comprises a gain smoothing calculator for calculating a smoothing gain in response to a factor representative of voicing in the sampled wideband signal, a factor representative of the stability of a set of linear prediction filter coefficients, and an innovative codebook gain. The gain smoothing device also comprises an amplifier for amplifying the innovative codevector with the smoothing gain to thereby produce a gain-smoothed innovative codevector. The function of the gain-smoothing device improves the perceived synthesized signal when background noise is present in the sampled wideband signal.

A system and method is provided that employs a frequency domain interpolative CODEC system for low bit rate coding of speech which comprises a linear prediction (LP) front end adapted to process an input signal that provides LP parameters which are quantized and encoded over predetermined intervals and used to compute a LP residual signal. An open loop pitch estimator adapted to process the LP residual signal, a pitch quantizer, and a pitch interpolator and provide a pitch contour within the predetermined intervals is also provided. Also provided is a signal processor responsive to the LP residual signal and the pitch contour and adapted to perform the following: provide a voicing measure, where the voicing measure characterizes a degree of voicing of the input speech signal and is derived from several input parameters that are correlated to degrees of periodicity of the signal over the predetermined intervals; extract a prototype waveform (PW) from the LP residual and the open loop pitch contour for a number of equal sub-intervals within the predetermined intervals; normalize the PW by a gain value of the PW; encode a magnitude of the PW; and directly quantize the PW in a magnitude domain without further decomposition of the PW into complex components, where the direct quantization is performed by a hierarchical quantization method based on a voicing classification using fixed dimension vector quantizers (VQ's).

The invention provides audio bandwidth expansion coding and decoding devices which can reconstruct a high frequency signal by using the lower code rate so as to improve the quality of the output audio signal. The audio bandwidth expansion coding device comprises a frequency division module, a core coder module, a linear prediction analysis module, a spectrum envelope quantization module, a high and low frequency correlation computation module and a gain computation module; and the audio bandwidth expansion decoding device comprises a core decoder module, a high-frequency spectrum envelope reconstruction module, a high-frequency residual error reconstruction module, a high-frequency gain adjustment module, a high-frequency signal synthesis module and a wideband signal reconstruction module.

An audio encoding apparatus capable of improving the frame cancellation error tolerance, without increasing the number of bits of a fixed code book, in a CELP type audio encoding. In this apparatus, a low frequency component waveform encoding part (210) calculates, based on a quantized LPC received from an LPC encoding part (202), a linear prediction residual signal of a digital audio signal received from an A/D converter (112), then performs a down sampling of the calculation result to extract the low frequency components comprising bands, which are lower than a predetermined frequency, in the audio signal, and then waveform encodes the extracted low frequency components to produce encoded low-frequency component information. Then, the low frequency component waveform encoding part (210) inputs this encoded low-frequency component information to a packetizing part (231), while inputting the quantized low-frequency component waveform encoded signal (sound source waveform), which has been produced by the waveform encoding, to a high frequency component encoding part (220).

A system and method is provided that employs a frequency domain interpolative CODEC system for low bit rate coding of speech which comprises a linear prediction (LP) front end adapted to process an input signal providing LP parameters which are quantized and encoded over predetermined intervals and used to compute a LP residual signal. An open loop pitch estimator adapted to process the LP residual signal, a pitch quantizer, and a pitch interpolator also provides a pitch contour within the predetermined intervals. A voice activity detector adapted to process the LP parameters and the open loop pitch contour over the predetermined intervals is also provided as well as a signal processor responsive to the LP residual signal and the pitch contour and adapted to perform the following functions: extract a prototype waveform (PW) from the LP residual and the open loop pitch contour for a number of equal sub-intervals within the predetermined invervals; normalize the PW by a gain value of the PW; encode a magnitude of the PW; and provide a voicing measure where the voicing measure characterizes a degree of vocing of the input speech signal and is derived from several input parameters that are correlated to degrees of periodicity of the signal over the predetermined intervals. The voicing measure is provided for the purpose of regenerating a PW phase at a decoder; and providing improved quantization of the PW magnitude at an encoder. The voicing measure is encoded jointly with a PW nonstationarity measure vector using a spectrally weighted vector quantizer having a codebook partioned based on a voiced and unvoiced mode.

A single-axis receiver processing, for example, complex vestigial sideband modulated signals with an equalizer with forward and feedback filters. Forward and feedback filters have parameters that are initialized and adapted to steady state operation. Adaptive equalization employs linear predictive filtering and error term generation based on various cost criteria. Adaptive equalization includes recursive update of parameters for forward and feedback filtering as operation changes between linear and decision-feedback equalization of either single or multi-channel signals. An adaptive, linear predictive filter generates real-valued parameters that are employed to set the parameters of the feedback filter. In an initialization mode, filter parameters are set via a linear prediction filter to approximate the inverse of the channel's impulse/frequency response and a constant modulus error term for adaptation of the filter parameters. In an acquisition mode, equalization is as linear equalization with a constant modulus error term, and possibly other error terms in combination, for adaptation of the filter parameters. In a tracking mode, equalization is as decision feedback equalization with decision-directed error terms for adaptation of the filter parameters. For some equalizer configurations, feedback filtering is applied to real-valued decisions corresponding to complex-valued received data, and includes real-part extraction of the error term employed for recursive update of filtering parameters. Where a training sequence is available to the receiver, initial parameters for forward filtering are estimated by correlation of the received signal with the training sequence.

The invention discloses a robust dicyclic photovoltaic grid-connected control method based on power feedforward, which mainly comprises three parts: proportion integration (PI) control of exocyclic voltage, dead-beat control of endocyclic robust predictive current and power feedforward control, wherein the exocyclic PI control of the voltage is used for stabilizing the capacitive voltage of a direct-current side; the dead-beat control of the endocyclic robust predictive current is used for carrying out linear prediction on the voltage of a power grid at a next control period and carrying out non-linear prediction on grid-connected current by advanced control so as to obtain the command value of the grid-connected current at the next period; and then, PWM (pulse-width modulation) and grid-connected control are realized by the dead-beat control. According to the robust dicyclic photovoltaic grid-connected control method based on the power feedforward, a photovoltaic grid-connected inverter has higher robustness, wider stability margin and quicker dynamic response, and the requirement of grid-connected running of an inverter is better met.

The invention relates to an anti-overlap method for tracking a moving target based on mean shift and belongs to the image processing technology field. The method comprises the following steps: constructing a mean shift model and a Kalman filter model; pre-estimating by utilizing a Kalman filter to obtain the initial position of searching each frame mean shift; obtaining the outline of an object by an image difference method; and defining whether similar factors can shade the object or not. When the similar factors shade the object, the position of the object in current frame is predicted and serves as a starting point for predicting the next frame by attaching different weights to color information and movement information respectively, according to the difference of mobile status of the object. Accordingly, the linear prediction of target position replaces the function of Kalman filter. Experiments prove that the method can realize the tracking of a quickly moving object and has good robustness for shading.

An image correction device for use in a pattern inspection apparatus is disclosed, which has automatic adaptability to variations in density of a pattern image of a workpiece being tested. The device is operable to identify a two-dimensional (2D) linear predictive model parameters from the pattern image of interest and determine the value of a total sum of these identified parameters. This value is then used to switch between a corrected pattern image due to the 2D linear prediction modeling and a corrected image that is interpolated by bicubic interpolation techniques. A pattern inspection method using the image correction technique is also disclosed.

A method to improve packet loss concealment for generation of a synthetic speech signal in a algebraic code excited linear prediction decoder for a voice over packet network. One method improves features for coding gains in the decoder and for post-filtering of the signals. An alternative method uses a classification method for the signal based on the bitstream in the decoder.

A method for speech processing in a code excitation linear prediction (CELP) based speech system having a plurality of modes including at least a first mode and a consecutive second mode. The method includes providing an input speech signal, dividing the speech signal into a plurality of frames, dividing at least one of the plurality of frames into sub-frames including a plurality of pulses, selecting a first number of pulses for the first mode, with a second number of remaining pulses in the frame plus the first number of pulses in the first mode for the second mode, providing a plurality of sub-modes between the first mode and the second mode, forming a base layer, forming an enhancement layer, generating a bit stream including a basic bit stream and an enhancement bit stream, wherein the basic bit stream is used to update memory states of the speech system.

Sound source separation, without permutation, using convolutional mixing independent component analysis based on a priori knowledge of the target sound source is disclosed. The target sound source can be a human speaker. The reconstruction filters used in the sound source separation take into account the a priori knowledge of the target sound source, such as an estimate the spectra of the target sound source. The filters may be generally constructed based on a speech recognition system. Matching the words of the dictionary of the speech recognition system to a reconstructed signal indicates whether proper separation has occurred. More specifically, the filters may be constructed based on a vector quantization codebook of vectors representing typical sound source patterns. Matching the vectors of the codebook to a reconstructed signal indicates whether proper separation has occurred. The vectors may be linear prediction vectors, among others.

The invention provides a method and device for predicting telecom traffic. The method comprises the following steps: determining the prediction granulation of the telecom traffic; selecting a historical sample and a prediction sample; respectively calculating the growth rates and the initial values of the telecom traffics of the historical sample and the prediction sample by using a unary linear regression model; calculating the prediction traffics of the historical sample and the prediction sample; reading the actual traffic of the historical sample; and adjusting the second prediction traffic according to the deviation between the traffic of first prediction and the actual traffic to obtain the prediction traffic adjusted by a user. The behavior mode of the user is obtained through analyzing the deviation between the linear prediction value and the actual value in the historical sample, and accordingly, the linear prediction value of future telecom traffic data is adjusted to obtainmore accurate telecom network traffic, thereby providing decision support for the user to plan, organize and manage network more accurately.

Encoding and/or decoding a wideband signal produces high frequency band spectra from low frequency band spectral information. Linear prediction filter coefficients are determined for the entire wideband spectrum of an input signal. An energy value in each of a plurality of sub-bands in the high frequency band is determined and encoded. The short-term correlation removed input signal is then down-sampled to form a low frequency band signal. At a decoder, the high frequency band signal is generated using the encoded low frequency band signal. The energy in each sub-band of the high frequency band is adjusted using the encoded energy value. Thus, the spectral envelope for the entire wideband signal is synthesized and decoded using linear predictive synthesis.

In methods and apparatus for encoding a gain parameter in a generalized linear predictive analysis-by-synthesis (GLPAS) coder, a subframe gain parameter is determined for each of a plurality of successive subframes of a frame, and a quantized frame gain parameter is determined for each frame using a delayed decision quantizer operating on the subframe gain parameters. The subframe gain parameters may be treated as components of a gain vector and the gain vector may be vector quantized to determine the quantized frame gain parameter. Encoder parameters are efficiently aligned with decoder parameters to ensure proper end-to-end operation. Alternatively, tree quantization or trellis quantization may be applied to the subframe gain parameters to determine the quantized frame gain parameter. The methods and apparatus are particularly applicable to low bit rate speech coding.

The related echo eliminator comprises: a linear predicted decoder to decode data into remote speech signal and extract the short activating signal and linear prediction coding parameter, a decorrelation filter to receive the predicted coding parameter and error signal for decorrelation and generate decorrelated error signal, an adaptive filter to receive all of the remote speech and short activating signal and decorrelated error signal and generate the estimated echo, and a subtracter to generate the error signal. This invention combines with LPC device, reduces algorithm complexity, and prompts algorithm convergence.

The invention relates to a speech information hiding secret communication system which organically integrates techniques, such as information hiding, passwords, communication, and the like. In the invention, MELP2.4KBps secret speech is hidden in the public speech of G.721, G.728, GSM and G.729 of a code-excited linear prediction code by connecting a public telephone network PSTN, mobile communication GSM/CDMA, videoconferences and a VoIP network and utilizing a speech information hiding and extraction algorithm, and a private channel is established in a public communication channel to carry out the real-time secret communication of secret speech information. The system adopts an embedded type technique based on DSP, and designs corresponding functional modules, such as speech coding and decoding, speech encryption and decryption, speech information hiding and extraction, and the like. Under the condition that a certain communication rate and a certain speech quality are ensured, secret speech real-time communication is carried out on various public channels, and the communication quality reaches the requirement of a communication standard.