Abstract: A frequency band expansion device includes processing circuitry to calculate a weighting coefficient based on a frequency gradient of the input signal; to generate a white noise signal; to generate a first white noise signal by performing filtering on the white noise signal; to generate a second white noise signal by regulating a phase characteristic of the white noise signal; to generate a third white noise signal by performing weighted addition on the first white noise signal and the second white noise signal by using the weighting coefficient; and to generate the output signal by adding together the input signal and a signal corresponding to the third white noise signal, wherein the processing circuitry is configured so that the phase characteristic of the second white noise signal becomes the same as the phase characteristic of the first white noise signal.

Abstract: The present invention is related to a method for coding excitation signal of a target speech comprising the steps of: extracting from a set of training normalised residual frames, a set of relevant normalised residual frames, said training residual frames being extracted from a training speech, synchronised on Glottal Closure Instant (GCI), pitch and energy normalised; determining the target excitation signal of the target speech; dividing said target excitation signal into GCI synchronised target frames; determining the local pitch and energy of the GCI synchronised target frames; normalising the GCI synchronised target frames in both energy and pitch, to obtain target normalised residual frames; determining coefficients of linear combination of said extracted set of relevant normalised residual frames to build synthetic normalised residual frames close to each target normalised residual frames; wherein the coding parameters for each target residual frames comprise the determined coefficients.

Abstract: Disclosed are a vector quantization device and others capable of adaptively adjusting a vector space of a code vector for quantization of a second stage by using a quantization result of a first stage and improving the quantization accuracy.

Abstract: The present research can decrease the amount of computation and enhance speech quality by using a global pulse replacement method in a fixed codebook search. The fixed codebook search method in a speech encoder based upon global pulse replacement, includes the steps of: (a) computing absolute values of the pulse-position likelihood-estimator vectors; (b) temporarily obtaining a codebook vector; (c) computing a mathematical equation by replacing a pulse; (d) determining whether a value computed based upon the mathematical equation is increased after pulse replacement; (e) obtaining a new codebook vector by replacing the pulse; and (f) maintaining a previous codebook vector.

Abstract: A method of concealing transmission error in a digital audio signal, wherein a signal that has been decoded after transmission is received, the samples decoded while the transmitted data is valid are stored, at least one short-term prediction operator and one long-term prediction operator are estimated as a function of stored valid samples, and any missing or erroneous samples in the decoder signal are generated using the estimated operators. The energy of the synthesized signal that is thus generated is controlled by means of a gain that is computed and adapted sample by sample.

Abstract: Provided is an audio encoding device which performs a closed loop search of a gain and a sound source vector without significantly increasing the calculation amount as compared to an open loop search. In the audio encoding device, firstly, a first parameter decision unit (121) performs a sound source search by an adaptive sound source codebook and then a second parameter decision unit (122) simultaneously performs by a closed loop, the sound source and the gain search by using a fixed sound source codebook. More specifically, for a combination of a fixed sound source vector and gain, the sum of a value obtained by multiplying a candidate fixed sound source vector by a candidate gain and a value obtained by multiplying an adaptive sound source vector by a candidate gain is subjected to a combination filter formed by a filter coefficient based on a quantization linear prediction coefficient so as to generate a combined signal.

Abstract: Disclosed is an audio encoding device capable of adjusting a spectrum inclination of a quantized noise without changing the Formant weight. The device includes: an HPF (131) which extracts a high-frequency component of the frequency region from an input audio signal; a high-frequency energy level calculation unit (132) which calculates an energy level of the high-frequency component in a frame unit; an LPF (133) which extracts a low-frequency component of the frequency region from the input audio signal; a low-energy level calculation unit (134) which calculates an energy level of a low-frequency component in a frame unit; an inclination correction coefficient calculation unit (141) multiplies the difference between SNR of the high-frequency component and SNR of the low-frequency component inputted from an adder (140) by a constant and adds a bias component to the product so as to calculate an inclination correction coefficient ?3.

Abstract: A hybrid approach is described for combining frequency warping and Gaussian Mixture Modeling (GMM) to achieve better speaker identity and speech quality. To train the voice conversion GMM model, line spectral frequency and other features are extracted from a set of source sounds to generate a source feature vector and from a set of target sounds to generate a target feature vector. The GMM model is estimated based on the aligned source feature vector and the target feature vector. A mixture specific warping function is generated each set of mixture mean pairs of the GMM model, and a warping function is generated based on a weighting of each of the mixture specific warping functions. The warping function can be used to convert sounds received from a source speaker to approximate speech of a target speaker.

Abstract: Provided is a voice encoding device for acquiring a satisfactory sound quality by making sufficient use of a tendency according to the noisiness or noiselessness of an input signal to be encoded. In this voice encoding device, a weight adding unit (206) in a searching loop (204) of a fixed code note searching unit (202) uses a function calculated from a code vector synthesized with a target to be encoded and spectrum enveloping information, as a calculated value to become the searching reference of the code vector stored in a fixed code note, and adds the weight according to the pulse number to form the code vector, to that calculated value.

Abstract: In accordance with one aspect of the invention, a selector supports the selection of a first encoding scheme or the second encoding scheme based upon the detection or absence of the triggering characteristic in the interval of the input speech signal. The first encoding scheme has a pitch pre-processing procedure for processing the input speech signal to form a revised speech signal biased toward an ideal voiced and stationary characteristic. The pre-processing procedure allows the encoder to fully capture the benefits of a bandwidth-efficient, long-term predictive procedure for a greater amount of speech components of an input speech signal than would otherwise be possible. In accordance with another aspect of the invention, the second encoding scheme entails a long-term prediction mode for encoding the pitch on a sub-frame by sub-frame basis.

Abstract: A multi-channel audio decoder provides a reduced complexity processing to reconstruct multi-channel audio from an encoded bitstream in which the multi-channel audio is represented as a coded subset of the channels along with a complex channel correlation matrix parameterization. The decoder translates the complex channel correlation matrix parameterization to a real transform that satisfies the magnitude of the complex channel correlation matrix. The multi-channel audio is derived from the coded subset of channels via channel extension processing using a real value effect signal and real number scaling.

Abstract: Disclosed is a system and method for implementing compression coding of audio signals, such as speech signals, using two long-term prediction (LTP) models. The method determines the parameters of a second long-term prediction model on the basis of the parameters of at least one first LTP model. The present invention is aimed at switching from an LTP model with a single coefficient (monotap) to an LTP model with several coefficients, (multitap) and vice versa, as well as at switching between two multitap LTP models. The complexity of the method may be adjusted, especially as a function of a desired compromise between a target complexity and a desired quality. A device for implementing the method according to the invention is, moreover, very useful for multiple codings in cascade (transcodings) or in parallel (multi-codings and multi-mode codings).