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 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: An audio signal interpolation device comprises a spectral movement calculation unit which determines a spectral movement which is indicative of a difference in each of spectral components between a frequency spectrum of a current frame of an input audio signal and a frequency spectrum of a previous frame of the input audio signal stored in a spectrum storing unit. An interpolation band determination unit determines a frequency band to be interpolated by using the frequency spectrum of the current frame and the spectral movement. A spectrum interpolation unit performs interpolation of spectral components in the frequency band for the current frame by using either the frequency spectrum of the current frame or the frequency spectrum of the previous frame.

Abstract: To alleviate degradation of sound quality which may be caused by pre-echoes and bit starvation. An acoustic analyzer analyzes an audio signal to calculate perceptual entropy indicating how many bits are required for quantization. A coded bit count monitor monitors the number of coded bits produced from the audio signal and calculates the number of available bits for the current frame. Based on the combination of the perceptual entropy and the number of available bits, a frame division number determiner determines a division number N for dividing a frame of the audio signal into N blocks. An orthogonal transform processor divides a frame by the determined division number and subjects each divided block of the audio signal to an orthogonal transform process, thereby obtaining orthogonal transform coefficients. A quantizer quantizes the orthogonal transform coefficients on a divided block basis.

Abstract: In a gain adjusting method and a gain adjusting device for adjusting gain of a processed voice signal that is obtained by signal processing of an input voice signal, a masking power of the processed voice signal is computed, and gain is adjusted for every frequency if the frequency is masked according to the masking power, such that a difference between the processed voice signal and the input voice signal where the frequency is not masked is canceled.

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: 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.

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 decoding apparatus includes a unit decoding and inversely quantizing coded data to obtain frequency domain audio signal data, a unit computing from the coded data one of the number of scale bits composed of the number of bits corresponding to the scale value of the coded data and the number of spectrum bits composed of the number of bits corresponding to the spectrum value of the coded data, a unit estimating a quantization error of the frequency domain audio signal data based on one of the number of scale bits and the number of spectrum bits of the coded data, a unit computing a correction amount based on the estimated quantization error and correct the frequency domain audio signal data obtained by the frequency domain data obtaining unit based on the computed correction amount, and a unit converting the corrected frequency domain audio signal data into the audio signal.

Abstract: An audio encoding apparatus that encodes audio signals of a plurality of channels, includes an adaptive bit allocation control unit that adaptively controls a number of encoding bits assigned to the audio signal of each channel in accordance with perceptual entropy of the audio signal of each of the channels, a fixed bit allocation control unit that fixedly controls the number of encoding bits assigned to the audio signal of each of the channels in predetermined allocations, and a channel encoding unit that encodes the audio signal of each of the channels based on the number of adaptive allocation bits assigned by the adaptive bit allocation control unit and the number of fixed allocation bits assigned by the fixed bit allocation control unit.

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: An encoding apparatus compresses a stereo signal using a sum signal and a difference signal of a left component signal and a right component signal of the stereo signal. The encoding apparatus includes a calculating unit that calculates complexity of the sum signal and complexity of the difference signal; a setting unit that sets, based on the complexity, an allocation rate of bits to be allocated in quantizing the sum signal and the difference signal; and a quantizing unit that quantizes the sum signal and the difference signal based on the allocation rate.

Abstract: An encoding apparatus converts an input signal into a frequency-domain spectrum signal, divides the converted spectrum signal into an arbitrary number of segments with respect to a time axis and a frequency axis, calculates a spectrum power of each segment and a feature parameter that represents a feature of the corresponding spectrum power, calculates a masking threshold using the calculated spectrum power of each segment, detects a segment having a spectrum power equal to or less than the calculated masking threshold, corrects the spectrum power of the detected segment, and encodes both the spectrum power of the corrected segment and the calculated parameter.

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: In a voice packet communication system, a voice packet loss concealment device compensates for the deterioration of voice quality due to voice packet loss. In the device, a detecting section detects a loss of a voice packet and outputting information; an estimating section estimates the voice characteristics of the lost segment using a pre-loss voice packet received before the lost segment or a post-loss voice packet received after the lost segment; a pitch signal generating section generates a pitch signal having the voice characteristics; and a lost packet generating section outputs the pitch signal generated by the pitch signal generating section, with the voice characteristics estimated by the estimating section, which allows abnormal noise and feeling of mute, subjective deterioration of naturalness and continuity to be improved, and the voice packet loss concealment to be further improved.

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: According to an aspect of an embodiment, a method for regenerating an audio signal including a low frequency component and a high frequency component by decoding a coded data including a first coded data and a second coded data, the method comprising the steps of: generating the low frequency component; generating the high frequency component; determining whether the low frequency component has transient characteristics or not; generating a low frequency correction component by removing a stationary component when the audio signal has the transient characteristics; generating a corrected high frequency component by correcting the high-frequency component on the basis of the duration of the low frequency correction component when the audio signal has the transient characteristics; and regenerating the audio signal by synthesizing the low frequency component with the corrected high-frequency component.

Abstract: A decoding apparatus that decodes a first encoded data that is encoded into a first time range from a low-frequency component of an audio signal, and a second encoded data that is used when creating a high-frequency component of the audio signal from the low-frequency component and encoded into a second time range, into the audio signal. In the decoding apparatus, a high-frequency component compensating unit that compensates the high-frequency component created from the second encoded data based on the first time range. A decoding unit that decodes into the audio signal by synthesizing the high-frequency component compensated by the high-frequency component compensating unit, and the low-frequency component decoded from the first encoded data.

Abstract: When creating SBR data in a the low-resolution mode, an encoding device divides a high-frequency component of input audio data being encoded by SBR method into a high-frequency band and a low-frequency band, and calculates an average high-frequency power value that indicates the average value of the power in the high-frequency band of the audio data, as well as an average low-frequency power value that indicates the average value of the power in the low-frequency band of the audio data. The encoding device then compares the average high-frequency power value and the average low-frequency power value, selecting the smaller of the two. The encoding device then corrects the power of the high-frequency component of the signal being encoded by the SBR method so that it equals the selected average power value.

Abstract: To alleviate degradation of sound quality which may be caused by pre-echoes and bit starvation. An acoustic analyzer analyzes an audio signal to calculate perceptual entropy indicating how many bits are required for quantization. A coded bit count monitor monitors the number of coded bits produced from the audio signal and calculates the number of available bits for the current frame. Based on the combination of the perceptual entropy and the number of available bits, a frame division number determiner determines a division number N for dividing a frame of the audio signal into N blocks. An orthogonal transform processor divides a frame by the determined division number and subjects each divided block of the audio signal to an orthogonal transform process, thereby obtaining orthogonal transform coefficients. A quantizer quantizes the orthogonal transform coefficients on a divided block basis.