Abstract: To provide a bandwidth extension method which allows reduction of computation amount in bandwidth extension and suppression of deterioration of quality in the bandwidth to be extended. In the bandwidth extension method: a low frequency bandwidth signal is transformed into a QMF domain to generate a first low frequency QMF spectrum; pitch-shifted signals are generated by applying different shifting factors on the low frequency bandwidth signal; a high frequency QMF spectrum is generated by time-stretching the pitch-shifted signals in the QMF domain; the high frequency QMF spectrum is modified; and the modified high frequency QMF spectrum is combined with the first low frequency QMF spectrum.

Abstract: Provided is a multi-channel acoustic signal processing device by which loads of arithmetic operations are reduced. The multi-channel acoustic signal processing device includes: a decorrelated signal generation unit, and a matrix operation unit and a third arithmetic unit. The decorrelated signal generation unit generates a decorrelated signal w? indicating a sound which includes a sound indicated by an input signal x and reverberation, by performing reverberation processing on the input signal x. The matrix operation unit and the third arithmetic unit generate audio signals of m channels, by performing arithmetic operation on the input signal x and the decorrelated signal w? generated by the decorrelated signal generation unit, using a matrix R3 which indicates distribution of a signal intensity level and distribution of reverberation.

Abstract: An encoding device (200) includes an MDCT unit (202) that transforms an input signal in a time domain into a frequency spectrum including a lower frequency spectrum, a BWE encoding unit (204) that generates extension data which specifies a higher frequency spectrum at a higher frequency than the lower frequency spectrum, and an encoded data stream generating unit (205) that encodes to output the lower frequency spectrum obtained by the MDCT unit (202) and the extension data obtained by the BWE encoding unit (204). The BWE encoding unit (204) generates as the extension data (i) a first parameter which specifies a lower subband which is to be copied as the higher frequency spectrum from among a plurality of the lower subbands which form the lower frequency spectrum obtained by the MDCT unit (202) and (ii) a second parameter which specifies a gain of the lower subband after being copied.

Abstract: To provide an audio signal processing apparatus which can perform, with low operation amount, audio signal processing that is either time stretch and/or compression processing or frequency modulation processing. The audio signal processing apparatus is intended to transform an input audio signal sequence using a predetermined adjustment factor. The audio signal processing apparatus includes a filter bank (2601) which transforms the input audio signal sequence into Quadrature Mirror Filter (QMF) coefficients using a filter for Quadrature Mirror Filter analysis (a QMF analysis filter) and an adjusting unit (2602) which adjusts the QMF coefficients based on a predetermined adjustment factor.

Abstract: Provided is an audio decoder which can reduce an amount of arithmetic operations while suppressing occurrence of aliasing noise. The audio decoder includes: a decoder (102) and an analysis filter bank (110) which generate, from a coded down-mixed signal, the first frequency band signal (x) corresponding to a down-mixed signal (M); a channel expansion unit (130) which converts the first frequency band signal (x) generated by the analysis filter bank (110) into output signals (y) corresponding to respective audio signals of N channels, using BC information; an synthesis filter bank (140) which performs band synthesis for the output signals (y) generate by the channel expansion unit (130) and thereby converts the output signals (y) into the respective audio signals of the N channels on a time axis; and an aliasing noise detection unit (120) which detects occurrence of aliasing noise in the first frequency band signal (x).

Abstract: To provide an acoustic signal processing apparatus which can reduce the amount of calculation in matrix arithmetic. An acoustic signal processing apparatus converts down-mixed acoustic signals of NI channels to acoustic signals of NO channels, where NO>NI.

Abstract: Provided is an encoding device (1) including: a pitch contour analysis unit (101) which detects information, a dynamic time-warping unit (102) which generates, based on the information, pitch change ratios (Tw_ratio in FIG. 18) within a range (86) including a range (86a) of the pitch change ratios corresponding to absolute pitch differences of 42 cents or larger; a first lossless coding unit (103) which codes the generated pitch parameters (102x); a time-warping unit (104) which shifts a pitch of a signal according to the information; and a second encoding unit which codes a signal (104x) obtained by the shifting.

Abstract: A temporal processing apparatus includes: a splitter splitting an audio signal, included in the sub-band domain, into diffuse signals indicating reverberating components and direct signals indicating non-reverberating components; a downmix unit generating a downmix signal by downmixing the direct signals; BPFs respectively generating a bandpass downmix signal and bandpass diffuse signals; normalization processing units respectively generating a normalized downmix signal and normalized diffuse signals; a scale computation processing unit computing, on a predetermined time slot basis, a scale factor indicating the magnitude of energy of the normalized downmix signal with respect to energy of the normalized diffuse signals; a calculating unit generating scale diffuse signals; a HPF generating high-pass diffuse signals; an adding unit generating addition signals; and a synthesis filter bank performing synthesis filter processing on the addition signals and transforming the addition signals into the time domains.

Abstract: A coding apparatus which suppresses an extreme increase in a bit rate, includes: a downmixing and coding unit (301) that downmixes audio signals that have been provided, to reduce the number of channels to be fewer than the number of the provided audio signals, and to code the downmix signals; an object parameter extracting unit (304) that extracts parameters indicating correlation between the audio signals; and a multiplexing circuit (309) that multiplexes the extracted parameters with the generated downmix coded signals. The object parameter extracting unit (304) includes: an object classifying unit (305) that classifies each of the provided audio signals into a predetermined one of types based on audio characteristics; and an object parameter extracting circuit (308) that extracts parameters using a temporal granularity and a frequency granularity each of which is determined for a corresponding one of the types.

Abstract: An energy corrector (105) for correcting a target energy for high-frequency components and a corrective coefficient calculator (106) for calculating an energy corrective coefficient from low-frequency subband signals are newly provided. These processors perform a process for correcting a target energy that is required when a band expanding process is performed on a real number only. Thus, a real subband combining filter and a real band expander which require a smaller amount of calculations can be used instead of a complex subband combining filter and a complex band expander, while maintaining a high sound-quality level, and the required amount of calculations and the apparatus scale can be reduced.

Abstract: There was no method of positioning virtual sound sources of object signals obtained from received coded object information, in a listening space on a receiving site side.

Abstract: A combination device (305) according to the present invention includes: a detection unit (501) that detects active coded bitstreams that are effective coded bitstreams from a plurality of coded bitstreams (116) within a predetermined time period; a first combining unit (504) that combines, from a plurality of downmix sub-streams (115) included in the coded bitstreams (116), only downmix sub-streams (115) included in the active coded bitstreams so as to generate a combined downmix sub-stream (121); and a second combining unit (506) that combines, from a plurality of parameter sub-streams (113) included in the coded bitstreams (116), only parameter sub-streams (113) included In the active coded bitstreams so as to generate a combined parameter sub-stream (122).

Abstract: An encoding device (200) includes an MDCT unit (202) that transforms an input signal in a time domain into a frequency spectrum including a lower frequency spectrum, a BWE encoding unit (204) that generates extension data which specifies a higher frequency spectrum at a higher frequency than the lower frequency spectrum, and an encoded data stream generating unit (205) that encodes to output the lower frequency spectrum obtained by the MDCT unit (202) and the extension data obtained by the BWE encoding unit (204). The BWE encoding unit (204) generates as the extension data (i) a first parameter which specifies a lower subband which is to be copied as the higher frequency spectrum from among a plurality of the lower subbands which form the lower frequency spectrum obtained by the MDCT unit (202) and (ii) a second parameter which specifies a gain of the lower subband after being copied.

Abstract: Provided is an audio decoder which can reduce an amount of arithmetic operations while suppressing occurrence of aliasing noise. The audio decoder includes: a decoder (102) and an analysis filter bank (110) which generate, from a coded down-mixed signal, the first frequency band signal (x) corresponding to a down-mixed signal (M); a channel expansion unit (130) which converts the first frequency band signal (x) generated by the analysis filter bank (110) into output signals (y) corresponding to respective audio signals of N channels, using BC information; an synthesis filter bank (140) which performs band synthesis for the output signals (y) generate by the channel expansion unit (130) and thereby converts the output signals (y) into the respective audio signals of the N channels on a time axis; and an aliasing noise detection unit (120) which detects occurrence of aliasing noise in the first frequency band signal (x).

Abstract: An encoding device (200) includes an MDCT unit (202) that transforms an input signal in a time domain into a frequency spectrum including a lower frequency spectrum, a BWE encoding unit (204) that generates extension data which specifies a higher frequency spectrum at a higher frequency than the lower frequency spectrum, and an encoded data stream generating unit (205) that encodes to output the lower frequency spectrum obtained by the MDCT unit (202) and the extension data obtained by the BWE encoding unit (204). The BWE encoding unit (204) generates as the extension data (i) a first parameter which specifies a lower subband which is to be copied as the higher frequency spectrum from among a plurality of the lower subbands which form the lower frequency spectrum obtained by the MDCT unit (202) and (ii) a second parameter which specifies a gain of the lower subband after being copied.

Abstract: The delay in a multi-channel audio coding apparatus and a multi-channel audio decoding apparatus is reduced. The audio coding apparatus includes: a downmix signal generating unit (410) that generates, in a time domain, a first downmix signal that is one of a 1-channel audio signal and a 2-channel audio signal from an input multi-channel audio signal; a downmix signal coding unit (404) that codes the first downmix signal; a first t-f converting unit (401) that converts the input multi-channel audio signal into a multi-channel audio signal in a frequency domain; and a spatial information calculating unit (409) that generates spatial information for generating a multi-channel audio signal from a downmix signal.

Abstract: To provide an enhanced true-to-life atmosphere enjoyed in multipoint connecting, and reduce a calculation load at a multipoint connection unit, as well.

Abstract: According to the present invention, it is possible to calculate appropriate chirp factor and noise component amount with a little processing amount. Input subband signal is segmented into a plurality of ranges by a range segmentation unit 101. The range segmentation is performed for energy value calculation, chirp factor calculation, noise component calculation, and tone component calculation, respectively, and determined range segmentation information ei, bi, qi, and hi are outputted. Respective processing for the energy calculation, the chirp factor calculation, the tone component calculation, and the noise component calculation are performed sequentially for the respective corresponding ranges. By using linear prediction processing, it is possible to obtain an parameter having higher accuracy with a little operation amount.

Abstract: An audio decoding device of the present invention includes: a decoding unit decoding a stream to a spectrum coefficient, and outputting stream information when a frame included in the stream cannot be decoded; an orthogonal transformation unit transforming the spectrum coefficient to a time signal; a correction unit generating a correction time signal based on an output waveform within a reference section that is in a section that overlaps between an error frame section to which the stream information is outputted and an adjacent frame section and that is a section in the middle of the adjacent frame section, when the decoding unit outputs the stream information: and an output unit generating the output waveform by synthesizing the correction time signal and the time signal.

Abstract: Provided is a multi-channel acoustic signal processing device by which loads of arithmetic operations are reduced. The multi-channel acoustic signal processing device (100) includes: a decorrelated signal generation unit (181), and a matrix operation unit (187) and a third arithmetic unit (186). The decorrelated signal generation unit (181) generates a decorrelated signal w? indicating a sound which includes a sound indicated by an input signal x and reverberation, by performing reverberation processing on the input signal x. The matrix operation unit (187) and the third arithmetic unit (186) generate audio signals of m channels, by performing arithmetic operation on the input signal x and the decorrelated signal w? generated by the decorrelated signal generation unit (181), using a matrix R3 which indicates distribution of a signal intensity level and distribution of reverberation.