Abstract: A device for determining a sound source direction determines a direction in which a source of a reached sound exists, based on at least one of a sound pressure difference between a first sound pressure that is a sound pressure of a first frequency component of a first part of the reached sound acquired by a first microphone and a second sound pressure that is a sound pressure of the first frequency component of a second part of the reached sound acquired by a second microphone, and a phase difference between a first phase that is a phase of a second frequency component of the first part of the reached sound and a second phase that is a phase of the second frequency component of the second part of the reached sound.

Abstract: A device for determining a sound source direction determines a direction in which a source of a reached sound exists, based on at least one of a sound pressure difference between a first sound pressure that is a sound pressure of a first frequency component of a first part of the reached sound acquired by a first microphone and a second sound pressure that is a sound pressure of the first frequency component of a second part of the reached sound acquired by a second microphone, and a phase difference between a first phase that is a phase of a second frequency component of the first part of the reached sound and a second phase that is a phase of the second frequency component of the second part of the reached sound.

Abstract: A sound processing device includes a first calculation unit configured to calculate a suppression gain of noise by using respective input signals input from a plurality of microphones; an integration unit configured to obtain an integration gain by using a suppression gain of an acoustic echo and the suppression gain of the noise; an application unit configured to apply the integration gain to one input signal among the plurality of input signals; and a second calculation unit configured to calculate the suppression gain of the acoustic echo by using signals to which the integration gain is applied, output signals that are output to a replay device, and the one input signal.

Abstract: An encoding method executed by a computer, the method includes converting by the computer information about a transient included in a low-frequency component of an audio signal into information about a transient included in a high-frequency component of the audio signal, detecting, by the computer the transient of the high-frequency component of the audio signal based on the high-frequency component of the audio signal and on the information about the transient of the high-frequency component obtained by the converting; and encoding, by the computer the high-frequency component of the audio signal based on the transient detected by the detecting.

Abstract: An audio signal coding device divides a frequency spectrum obtained from an input digital signal to a plurality of bands, scales and quantizes divided frequency spectra based on a scalefactor of each of the bands and a common scale which is common to the plurality of bands, and codes quantized frequency spectra. The audio signal coding device includes a band number determination unit configured to calculate a number of coding bands for coding the quantized frequency spectra, and a common scale estimation unit configured to estimate the common scale in accordance with the number of coding bands.

Abstract: An audio coding device includes a time frequency transform unit that, with respect to each of a plurality of channels included in an audio signal, generates a time frequency signal indicating frequency components at each time by performing a time frequency transform on a signal of the channel; a transient detection unit that detects a transient with respect to each of the plurality of channels so as to obtain a transient detection time; a transient time correction unit that, when a difference in transient detection times between an early detection channel in which the transient detection time is earliest and a late detection channel that is a channel other than the early detection channel among the plurality of channels is within a range in which the transient; a grid determination unit that, with respect to each of the plurality of channels, and a coding unit that codes.

Abstract: An audio coding device includes a time-to-frequency converter that performs time-to-frequency conversion on each frame of a signal in at least one channel included in an audio signal in a predetermined length of time in order to convert the signal in the at least one channel to a frequency signal; a complexity calculator that calculates complexity of the frequency signal for each of the at least one channel. The audio further includes a bit allocation controller that determines a number of bits to be allocated to each of at least one channel so that more bits are allocated to the each of the at least one channel as the complexity of the each of at least one channel increases, and increases the number of bits to be allocated as an estimation error in the number; and a coder that codes the frequency signal.

Abstract: A disclosed encoding device converts an audio signal into frequency spectra, determines allowable error powers with respect to bands divided by the frequency of the audio signal by a predetermined with, detects a tonal frequency spectrum from the frequency spectra, and detects a band containing the frequency spectrum. Using the detection result and the allowable error powers, the encoding device performs correction such that allowable error powers determined by a power determining unit with respect to bands adjacent to the band detected by a detecting unit become smaller than the powers of the frequency spectra with respect to the adjacent bands, and quantizes each of frequency spectra having greater powers than the corrected allowable error powers.

Abstract: An encoding device includes, an estimation unit to estimate a decoded signal of a plurality of channels based on a down-mix signal obtained by down-mixing an input signal of the plurality of channels, similarity between the channels of the input signal, and an intensity difference between the channels of the input signal; an analysis unit to analyze a phase of the input signal and a phase of the decoded signal; a calculation unit to calculate phase information based on the phase of the input signal and the phase of the decoded signal; and a coding unit to encode the similarity between the channels of the input signal, the intensity difference between the channels of the input signal, and the phase information.

Abstract: A downmixing device includes: a matrix conversion unit configured to perform a matrix operation for an input signal; a rotation correction unit configured to rotate an output signal of the matrix conversion unit; a spatial information extraction unit configured to extract spatial information from the output signal of the rotation correction unit; and an error calculation unit configured to calculate an error amount of the matrix operation result for the input signal by performing a matrix operation for the output signal of the rotation correction unit and the spatial information extracted by the spatial information extraction unit using a matrix that is inverse to the matrix used for the matrix operation by the matrix conversion unit.

Abstract: An audio encoding device includes, a time-frequency transform unit that transforms signals of channels included in an audio signal having a first number of channels into frequency signals respectively, a down-mix unit that generates an audio frequency signal having a second number of channels, a low channel encoding unit that generates a low channel audio code by encoding the audio frequency signal, a space information extraction unit that extracts space information representing spatial information of a sound, an importance calculation unit that calculates importance on the basis of the space information, a space information correction unit that corrects the space information, a space information encoding unit that generates a space information code, and a multiplexing unit that generates an encoded audio signal by multiplexing the low channel audio code and the space information code.

Abstract: A decoding apparatus decodes a first encoded data that is encoded from a low-frequency component of an audio signal, and a second encoded data that is used when creating a high-frequency component of an audio signal from a low-frequency component and encoded in accordance with a certain bandwidth, into the audio signal. In the decoding apparatus, a high-frequency component detecting unit divides the high-frequency component into bands with a certain interval range correspondingly to the certain bandwidth, and detects magnitude of the high-frequency components corresponding to each of the bands. A high-frequency component compensating unit compensates the high-frequency components based on the magnitude of the high-frequency components corresponding to each of the bands detected by the high-frequency component detecting unit.

Abstract: A sound processing device includes a first calculation unit configured to calculate a suppression gain of noise by using respective input signals input from a plurality of microphones; an integration unit configured to obtain an integration gain by using a suppression gain of an acoustic echo and the suppression gain of the noise; an application unit configured to apply the integration gain to one input signal among the plurality of input signals; and a second calculation unit configured to calculate the suppression gain of the acoustic echo by using signals to which the integration gain is applied, output signals that are output to a replay device, and the one input signal.

Abstract: An audio decoding apparatus and method are provided. The audio decoding apparatus includes a spectrum converting part configured to divide the first frequency spectrum in each channel of the first audio signal in a time direction or in a frequency direction to calculate a first signal sequence having the same time resolution and the same frequency resolution in all the channels of the first audio signal, a down-mixing part configured to perform weighted addition on the signals at the same time and within the same frequency band included in the first signal sequence in all the channels to calculate a second signal sequence having channels of a second number different from the first number of channels.

Abstract: A decoded sound analysis unit (104) calculates, regarding the frequency-region stereo signals L(b) and R(b) decoded by the PS decoding unit (103), a second degree of similarity (109) and a second intensity difference (110) from the decoded sound signals. A spectrum correction unit (105) detects a distortion added by the parametric-stereo conversion by comparing the second degree of similarity (109) and the second intensity difference (110) calculated at the decoding side with the first degree of similarity (107) and the first intensity difference (108) calculated and transmitted from the encoding side, and corrects the spectrum of the frequency-region stereo decoded signals L(b) and R(b).

Abstract: An SBR encoder includes a filter bank that receives an input signal, a time/frequency grid generator that controls a number of bits of various parameters, a parameter calculator that calculates various parameters, a parameter coding unit that encodes the parameters, an upper-limit number-of-bit storage unit that stores an upper limit of the number of bit of encoded data of high-frequency component finally generated in a high-pass encoding process, and a number-of-bit controller. The number-of-bit controller controls the high-pass encoding process by preferentially encoding a parameter having a large influence to sound quality and not encoding a parameter having a small influence to the sound quality relative to a plurality of parameters, so that the number of bits of the encoded data of high-frequency component finally generated in the high-pass encoding process becomes equal to or less than the upper limit to be stored in the upper-limit number-of-bit storage unit.

Abstract: An encoding method executed by a computer, the method includes converting by the computer information about a transient included in a low-frequency component of an audio signal into information about a transient included in a high-frequency component of the audio signal, detecting, by the computer the transient of the high-frequency component of the audio signal based on the high-frequency component of the audio signal and on the information about the transient of the high-frequency component obtained by the converting; and encoding, by the computer the high-frequency component of the audio signal based on the transient detected by the detecting.

Abstract: An audio decoding method includes: acquiring, from encoded audio data, a reception audio signal and first auxiliary decoded audio information; calculating coefficient information from the first auxiliary decoded audio information; generating a decoded output audio signal based on the coefficient information and the reception audio signal; decoding to result in a decoded audio signal based on the first auxiliary decoded audio signal and the reception audio signal; calculating, from the decoded audio signal, second auxiliary decoded audio information corresponding to the first auxiliary decoded audio information; detecting a distortion caused in a decoding operation of the decoded audio signal by comparing the second auxiliary decoded audio information with the first auxiliary decoded audio information; correcting the coefficient information in response to the detected distortion; and supplying the corrected coefficient information as the coefficient information when generating the decoded output audio signal.

Abstract: A data embedding device for embedding data in a speech code obtained by encoding a speech in accordance with a speech encoding method based on a voice generation process of a human being, includes an embedding judgment unit, every speech code, judging whether or not data should be embedded in the speech code, and an embedding unit embedding data in two or more parameter codes of a plurality of parameter codes constituting the speech code for which it is judged by the embedding judgment unit that the data should be embedded.

Abstract: An audio information processing apparatus and method include dividing an audio signal, determining a time period having a power change ratio of an audio signal larger than a first threshold value as an attack candidate, searching the time period of the attack candidate and a time period immediately before the time period of the attack candidate for an attack starting point, correcting a power of an audio signal included in the time period, and determining whether a power change ratio of the audio signal included in the time period is larger than a second threshold value for attack detection which is larger than the first threshold value.