Abstract: Provided is a training method of a neural network that is applied to an audio signal encoding method using an audio signal encoding apparatus, the training method including generating a masking threshold of a first audio signal before training is performed, calculating a weight matrix to be applied to a frequency component of the first audio signal based on the masking threshold, generating a weighted error function obtained by correcting a preset error function using the weight matrix, and generating a second audio signal by applying a parameter learned using the weighted error function to the first audio signal.

Abstract: Provided is an apparatus and method for encoding/decoding audio based on a block. A method of encoding an audio signal may include dividing each of frame of input signal that constitute an audio signal into a plurality of subframes; transforming the subframes to a frequency domain; determining a two-dimensional (2D) intra block using the subframes transformed to the frequency domain; and encoding the 2D intra block. The 2D intra block may be a block that two-dimensionally displays frequency coefficients of the subframes transformed to the frequency domain using a time and a frequency.

Abstract: Disclosed is an audio watermark insertion method. The audio watermark insertion method includes performing a modulated complex lapped transform (MCLT) on a first audio signal, inserting a bit string of a watermark in the first audio signal obtained by performing the MCLT, performing an inverse modified discrete cosine transform (IMDCT) on the first audio signal in which the bit string is inserted, and obtaining a second audio signal, which is the first audio signal in which the watermark is inserted, by performing an overlap-add on a signal obtained by performing the IMDCT and a neighbor frame signal.

Abstract: Disclosed is a bitstream generation method performed by an acoustic data transmission (ADT) encoder, the method including receiving a first audio signal, receiving additional information converted into a bitstream, and transmitting a second audio signal obtained by inserting the bitstream into the first audio signal, to an ADT decoder.

Abstract: An encoding method of an encoder is provided. The encoder generates first MDCT coefficients by transforming an input signal, and generates MDCT indices by quantizing the first MDCT coefficients. The encoder generates second MDCT coefficients by dequantizing the MDCT indices, and calculates MDCT residual coefficients using differences between the first MDCT coefficients and the second MDCT coefficients. The encoder generates a residual index by encoding the MDCT residual coefficients, and generates gain indices corresponding to gains from the first MDCT coefficients and the second MDCT coefficients.

Abstract: Provided are a method and an apparatus for decoding an audio signal. A method for decoding an audio signal encoded by a layered sinusoidal pulse coding scheme using one or more sinusoidal pulses includes decoding the encoded audio signal, setting a smoothing frequency band of the decoded audio signal according to a layer structure of the layered sinusoidal pulse coding scheme, dividing the smoothing frequency band into one or more subbands, and smoothing the decoded audio signal on a subband-by-subband basis. Accordingly, a decoding operation time can be reduced and the quality of a synthesized signal can be improved by variably setting a frequency band to be smoothed, when decoding an audio signal encoded by a layered sinusoidal pulse coding scheme using one or more sinusoidal pulses.

Abstract: Provided are a method and an apparatus for encoding and decoding an audio signal. A method for encoding an audio signal includes receiving a transformed audio signal, dividing the transformed audio signal into a plurality of subbands, performing a first sinusoidal pulse coding operation on the subbands, determining a performance region of a second sinusoidal pulse coding operation among the subbands on the basis of coding information of the first sinusoidal pulse coding operation, and performing the second sinusoidal pulse coding operation on the determined performance region, wherein the first sinusoidal pulse coding operation is performed variably according to the coding information. Accordingly, it is possible to further improve the quality of a synthesized signal by considering the sinusoidal pulse coding of a lower layer when encoding or decoding an audio signal in an upper layer by a layered sinusoidal pulse coding scheme.

Abstract: Provided are a method and an apparatus for encoding and decoding an audio signal. A method for encoding an audio signal includes receiving a transformed audio signal, dividing the transformed audio signal into a plurality of subbands, performing a first sinusoidal pulse coding operation on the subbands, determining a performance region of a second sinusoidal pulse coding operation among the subbands on the basis of coding information of the first sinusoidal pulse coding operation, and performing the second sinusoidal pulse coding operation on the determined performance region, wherein the first sinusoidal pulse coding operation is performed variably according to the coding information. Accordingly, it is possible to further improve the quality of a synthesized signal by considering the sinusoidal pulse coding of a lower layer when encoding or decoding an audio signal in an upper layer by a layered sinusoidal pulse coding scheme.

Abstract: Provided is a method and apparatus for encoding and decoding an enhancement layer to reduce quantization error in a G.711 codec. Exponent indices of additional mantissa information of each sample are calculated based upon exponent information of each sample in a frame. A process of allocating 1 bit to each sample with a current exponent index is repeated, the exponent index starting from the maximum value while decreasing by 1 at every repetition until the total number of bits allocated to the samples is equal to the total number of available bits in the frame. And the most significant bits, as many as the number of bits allocated to each sample, are extracted from the additional mantissa information of each sample in the frame.

Abstract: An encoding method of an encoder is provided. The encoder generates first MDCT coefficients by transforming an input signal, and generates MDCT indices by quantizing the first MDCT coefficients. The encoder generates second MDCT coefficients by dequantizing the MDCT indices, and calculates MDCT residual coefficients using differences between the first MDCT coefficients and the second MDCT coefficients. The encoder generates a residual index by encoding the MDCT residual coefficients, and generates gain indices corresponding to gains from the first MDCT coefficients and the second MDCT coefficients.

Abstract: Provides is an embedded code-excited linear prediction speech coding/decoding apparatus and method that can deal with the capacity change of speech transmission channel by modeling an error signal not coded at a core speech coder based on a transmission rate in a multiple pulse search mode or gain compensation mode and then transmitting it in an optimum mode. The apparatus includes a core speech coding unit for coding an input speech signal with spectral envelop and an excitation signal, a transmission rate determination unit for allocating the number of bits additionally allowed depending on a capacity of a transmission channel, and an embedded excitation signal coding unit for coding a residual excitation signal that is not coded in the core speech coding unit based on the number of additionally allowed bits using one of a multiple pulse excitation coding mode and a gain compensation mode.

Abstract: Provided are a method and an apparatus for encoding and decoding an audio signal. A method for encoding an audio signal includes receiving a transformed audio signal, dividing the transformed audio signal into a plurality of subbands, performing a first sinusoidal pulse coding operation on the subbands, determining a performance region of a second sinusoidal pulse coding operation among the subbands on the basis of coding information of the first sinusoidal pulse coding operation, and performing the second sinusoidal pulse coding operation on the determined performance region, wherein the first sinusoidal pulse coding operation is performed variably according to the coding information. Accordingly, it is possible to further improve the quality of a synthesized signal by considering the sinusoidal pulse coding of a lower layer when encoding or decoding an audio signal in an upper layer by a layered sinusoidal pulse coding scheme.

Abstract: Provided are a method and an apparatus for decoding an audio signal. A method for decoding an audio signal encoded by a layered sinusoidal pulse coding scheme using one or more sinusoidal pulses includes decoding the encoded audio signal, setting a smoothing frequency band of the decoded audio signal according to a layer structure of the layered sinusoidal pulse coding scheme, dividing the smoothing frequency band into one or more subbands, and smoothing the decoded audio signal on a subband-by-subband basis. Accordingly, a decoding operation time can be reduced and the quality of a synthesized signal can be improved by variably setting a frequency band to be smoothed, when decoding an audio signal encoded by a layered sinusoidal pulse coding scheme using one or more sinusoidal pulses.

Abstract: Provided is a method and apparatus for encoding and decoding an enhancement layer to reduce quantization error in a G.711 codec. Exponent indices of additional mantissa information of each sample are calculated based upon exponent information of each sample in a frame. A process of allocating 1 bit to each sample with a current exponent index is repeated, the exponent index starting from the maximum value while decreasing by 1 at every repetition until the total number of bits allocated to the samples is equal to the total number of available bits in the frame. And the most significant bits, as many as the number of bits allocated to each sample, are extracted from the additional mantissa information of each sample in the frame.

Abstract: A caller identification system provides a caller identifier of an actual user rather than the conventional telephone number dependent on a caller terminal. Further, a caller base billing method and its system can bill an actual user by carrying out user authentication through a certification authority for a caller identifier that is independent of a caller terminal. The method includes the steps of receiving authentication information of the caller from a caller terminal; verifying the authentication information; and carrying out call setup if the authentication information and the certificate match, wherein the authentication information is generated by using a caller identifier and a private key that are applied by the caller, with no dependency on the caller terminal, and caller information corresponding to the caller identifier is transmitted to a callee terminal during call setup.

Abstract: The present invention overcomes problems of tandem coding method such as degradation of speech quality, increased system latency and computations. An apparatus for trans-coding between code excited linear prediction (CELP) type codecs with different bandwidths, includes: a format parameter translating unit for generating output formant parameters by translating formant parameters from input CELP format to output CELP format; a formant parameter quantizing unit for receiving the output format formant parameters and quantizing the output format formant filter coefficients; an excited parameter translating unit for generating output excitation parameters by translating excitation parameters from input CELP format to output CELP format; and an excitation quantizing unit for receiving the output format excitation parameters and quantizing the output format excitation parameters.

Abstract: An apparatus for concealing a highband error in a spilt-band wideband voice codec in accordance with the present invention is disclosed. The apparatus includes: a lowband LPC coefficient extracting unit for extracting a lowband linear predictive coding (LPC) coefficient from a lowband voice signal passed by a lowband decoding unit; a highband excitation signal generating unit for generating a highband excitation signal based on the lowband voice signal and the lowband LPC coefficient; a highband LPC coefficient generating unit for generating a highband LPC coefficient based on the lowband LPC coefficient; a highband voice synthesizing unit for synthesizing a highband voice signal based on the highband excitation signal and the highband LPC coefficient; and a high pass filtering unit for removing a lowband component of the synthesized highband voice signal by the highband voice synthesis unit and generating the synthesized highband voice signal.

Abstract: A transcoder for use between speech codecs using different Code-Excited Linear Prediction (CELP) type and a method therefor are disclosed. The transcoder includes a decoding unit of an input CELP codec, a transcoding filter, a transcoding filter design unit, and an encoding unit of an output CELP codec. By substituting a post-filter and a perceptual weighting filter of a prior art with one transcoding filter, the calculation amount of the transcoder is reduced, and speech quality decoded at a receiving end is improved.

Abstract: Provides is an embedded code-excited linear prediction speech coding/decoding apparatus and method that can deal with the capacity change of speech transmission channel by modeling an error signal not coded at a core speech coder based on a transmission rate in a multiple pulse search mode- or gain compensation mode and then transmitting it in an optimum mode. The apparatus includes a core speech coding unit for coding an input speech signal with spectral envelop and an excitation signal, a transmission rate determination unit for allocating the number of bits additionally allowed depending on a capacity of a transmission channel, and an embedded excitation signal coding unit for coding a residual excitation signal that is not coded in the core speech coding unit based on the number of additionally allowed bits using one of a multiple pulse excitation coding mode and a gain compensation mode.

Abstract: A caller identification system provides a caller identifier of an actual user rather than the conventional telephone number dependent on a caller terminal. Further, a caller base billing method and its system can bill an actual user by carrying out user authentication through a certification authority for a caller identifier that is independent of a caller terminal. The method includes the steps of receiving authentication information of the caller from a caller terminal; verifying the authentication information; and carrying out call setup if the authentication information and the certificate match, wherein the authentication information is generated by using a caller identifier and a private key that are applied by the caller, with no dependency on the caller terminal, and caller information corresponding to the caller identifier is transmitted to a callee terminal during call setup.