Abstract: Method and device of extending a signal band of a voice or audio signal are provided. The bandwidth extension method includes the steps of: performing a modified discrete cosine transform (MDCT) process on an input signal to generate a first transform signal; generating a second transform signal and a third transform signal on the basis of the first transform signal; generating normalized components and energy components of the first transform signal, the second transform signal, and the third transform signal therefrom; generating an extended normalized component from the normalized components and generating an extended energy component from the energy components; generating an extended transform signal on the basis of the extended normalized component and the extended energy component; and performing an inverse MDCT (IMDCT) process on the extended transform signal.

Abstract: A speech signal encoding method and a speech signal decoding method are provided. The speech signal encoding method includes the steps of specifying an analysis frame in an input signal; generating a modified input based on the analysis frame; applying a window to the modified input; generating a transform coefficient by performing an MDCT (Modified Discrete Cosine Transform) on the modified input to which the window has been applied; and encoding the transform coefficient. The modified input includes the analysis frame and a self replication of all or a part of the analysis frame.

Abstract: The present invention relates to a method for processing an audio signal, comprising the following steps: performing a linear predictive analysis on the current frame of an audio signal so as to generate a first target vector, which is a target vector of a first stage, on the basis of a plurality of linear prediction transform coefficients; performing vector quantization on the first target vector so as to acquire a predetermined number of first temporary candidate code vectors of the first stage; calculating first temporary candidate errors, which are errors between the first temporary candidate code vectors and the first target vector; and determining a first number, which is the number of the first candidate code vectors, on the basis of the first temporary candidate errors, and acquiring first final candidate code vectors in the same amount as the first number.

Abstract: Disclosed is an audio signal processing method comprising the steps of: receiving an audio signal containing current frame data; generating a first temporary output signal for the current frame when an error occurs in the current frame data, by carrying out frame error concealment with respect to the current frame data a random codebook; generating a parameter by carrying out one or more of short-term prediction, long-term prediction and a fixed codebook search based on the first temporary output signal; and memory updating the parameter for the next frame; wherein the parameter comprises one or more of pitch gain, pitch delay, fixed codebook gain and a fixed codebook.

Abstract: Disclosed is an audio signal processing method comprising the steps of: receiving an audio signal containing current frame data; generating a first temporary output signal for the current frame when an error occurs in the current frame data, by carrying out frame error concealment with respect to the current frame data a random codebook; generating a parameter by carrying out one or more of short-term prediction, long-term prediction and a fixed codebook search based on the first temporary output signal; and memory updating the parameter for the next frame; wherein the parameter comprises one or more of pitch gain, pitch delay, fixed codebook gain and a fixed codebook.

Abstract: Disclosed is an audio signal processing method comprising the steps of: receiving an audio signal containing current frame data; generating a first temporary output signal for the current frame when an error occurs in the current frame data, by carrying out frame error concealment with respect to the current frame data a random codebook; generating a parameter by carrying out one or more of short-term prediction, long-term prediction and a fixed codebook search based on the first temporary output signal; and memory updating the parameter for the next frame; wherein the parameter comprises one or more of pitch gain, pitch delay, fixed codebook gain and a fixed codebook.

Abstract: Method and device of extending a signal band of a voice or audio signal are provided. The bandwidth extension method includes the steps of: performing a modified discrete cosine transform (MDCT) process on an input signal to generate a first transform signal; generating a second transform signal and a third transform signal on the basis of the first transform signal; generating normalized components and energy components of the first transform signal, the second transform signal, and the third transform signal therefrom; generating an extended normalized component from the normalized components and generating an extended energy component from the energy components; generating an extended transform signal on the basis of the extended normalized component and the extended energy component; and performing an inverse MDCT (IMDCT) process on the extended transform signal.

Abstract: A speech signal encoding method and a speech signal decoding method are provided. The speech signal encoding method includes the steps of specifying an analysis frame in an input signal; generating a modified input based on the analysis frame; applying a window to the modified input; generating a transform coefficient by performing an MDCT (Modified Discrete Cosine Transform) on the modified input to which the window has been applied; and encoding the transform coefficient. The modified input includes the analysis frame and a self replication of all or a part of the analysis frame.

Abstract: The present invention relates to a method for processing an audio signal, comprising the following steps: performing a linear predictive analysis on the current frame of an audio signal so as to generate a first target vector, which is a target vector of a first stage, on the basis of a plurality of linear prediction transform coefficients; performing vector quantization on the first target vector so as to acquire a predetermined number of first temporary candidate code vectors of the first stage; calculating first temporary candidate errors, which are errors between the first temporary candidate code vectors and the first target vector; and determining a first number, which is the number of the first candidate code vectors, on the basis of the first temporary candidate errors, and acquiring first final candidate code vectors in the same amount as the first number.

Abstract: Disclosed is an audio signal processing method comprising the steps of: receiving an audio signal containing current frame data; generating a first temporary output signal for the current frame when an error occurs in the current frame data, by carrying out frame error concealment with respect to the current frame data a random codebook; generating a parameter by carrying out one or more of short-term prediction, long-term prediction and a fixed codebook search based on the first temporary output signal; and memory updating the parameter for the next frame; wherein the parameter comprises one or more of pitch gain, pitch delay, fixed codebook gain and a fixed codebook.

Abstract: A method, apparatus, and system for encoding or decoding a broadband voice signal are provided. The method includes extracting a linear prediction coefficient (LPC) from the broadband voice signal; outputting a linear prediction (LP) residual signal; pitch-searching a spectrum of the LP residual signal; extracting spectral magnitudes and phases of the LP residual signal, which correspond to a damping factor; obtaining, from among the extracted spectral magnitudes and phases, a first spectral magnitude and a first phase at which a power value of the LP residual signal is minimized; quantizing the first spectral magnitude and the first phase; and decoding the broadband voice signal. The apparatus includes a linear prediction coefficient (LPC) analyzer; an LPC inverse filter; a pitch searching unit; a sinusoidal analyzer; and a phase and spectral magnitude quantizer. The system includes a broadband voice encoding apparatus and a broadband voice decoding apparatus.

Abstract: A method, apparatus, and system for encoding or decoding a broadband voice signal are provided. The method includes extracting a linear prediction coefficient (LPC) from the broadband voice signal; outputting a linear prediction (LP) residual signal; pitch-searching a spectrum of the LP residual signal; extracting spectral magnitudes and phases of the LP residual signal, which correspond to a damping factor; obtaining, from among the extracted spectral magnitudes and phases, a first spectral magnitude and a first phase at which a power value of the LP residual signal is minimized; quantizing the first spectral magnitude and the first phase; and decoding the broadband voice signal. The apparatus includes a linear prediction coefficient (LPC) analyzer; an LPC inverse filter; a pitch searching unit; a sinusoidal analyzer; and a phase and spectral magnitude quantizer. The system includes a broadband voice encoding apparatus and a broadband voice decoding apparatus.