Abstract: Provided is a measurement technology of an optical phase modulation amount due to sound without being affected by noise included in average light intensity.

Abstract: A sound signal downmixing method includes a step of obtaining, for each of two channels, a signal obtained by adding an input sound signal of one channel to a signal obtained by delaying an input sound signal of the other channel and multiplying the delayed input sound signal by a weight value, as a delayed crosstalk-added signal of the one channel, a step of obtaining preceding channel information and a left-right correlation value, and step of obtaining a downmix signal by performing weighted addition on the input sound signals of the two channels based on the left-right correlation value and the preceding channel information such that more of a signal derived from an input sound signal of a preceding channel among the signals derived from the input sound signals of the two channels is included as the left-right correlation value becomes larger.

Abstract: A technique is provided for measuring acoustic characteristics of a parametric speaker having any shape and characteristics. An acoustic characteristics calculation device includes a measurement data acquisition unit that acquires a set of a phase change amount ?q of light due to radiated sound of a parametric speaker obtained in q-th measurement by an optical measurement device and a parameter ?q that characterizes an optical path Lq in the measurement (q=1, 2, . . . , Q), and a sound field calculation unit that calculates a sound pressure at a point in a sound field that is generated by the radiated sound of the parametric speaker from the set of the phase change amount ?q and the parameter ?q (q=1, 2, . . . , Q).

Abstract: An envelope sequence is provided that can improve approximation accuracy near peaks caused by the pitch period of an audio signal. A periodic-combined-envelope-sequence generation device according to the present invention takes, as an input audio signal, a time-domain audio digital signal in each frame, which is a predetermined time segment, and generates a periodic combined envelope sequence as an envelope sequence. The periodic-combined-envelope-sequence generation device according to the present invention comprises at least a spectral-envelope-sequence calculating part and a periodic-combined-envelope generating part. The spectral-envelope-sequence calculating part calculates a spectral envelope sequence of the input audio signal on the basis of time-domain linear prediction of the input audio signal.

Abstract: A sound signal processing device that obtains a two-channel stereo encoding target signal to be subjected to stereo encoding from a two-channel stereo input sound signal, the sound signal processing device including a signal mixing unit that obtains, as an encoding target signal, for each channel, a signal obtained by performing weighted addition on an input sound signal of the channel and an input sound signal of the other channel, the signal being closer to the input sound signal of the channel as an absolute value of an inter-channel time difference of the two-channel stereo input sound signal is smaller.

Abstract: A sound signal processing device that obtains a two-channel stereo encoding target signal to be subjected to stereo encoding from a two-channel stereo input sound signal, the sound signal processing device including a signal mixing unit that obtains, as an encoding target signal, for each channel, a signal obtained by performing weighted addition on an input sound signal of the channel and an input sound signal of the other channel, the signal being closer to the input sound signal of the channel as the two-channel stereo input sound signal is likely to be a single sound source.

Abstract: A sound signal processing device that obtains a two-channel stereo encoding target signal to be subjected to stereo encoding from a two-channel stereo input sound signal, the sound signal processing device including a signal mixing unit that obtains, as an encoding target signal, for each channel, a signal obtained by performing weighted addition on an input sound signal of the channel and an input sound signal of the other channel, the signal being closer to the input sound signal of the channel as a bit rate of the stereo encoding is higher.

Abstract: A technique for accurately measuring acoustic characteristics of a parametric array is provided. A first calculation unit that calculates a complex amplitude d of an optical phase at a frequency fd from an amount of change ?s of the optical phase caused by demodulated sound S, assuming that qdiff(??, ??) is a function defined using a Gaussian beam expansion method, a second calculation unit that calculates a function value qdiff(?, ?) at a point X and a line integral value ?Lqdiff(??, ?)d?? of the function qdiff(??, ?) along the optical path L, and a third calculation unit that calculates a complex amplitude p of the demodulated sound with the frequency fd at the point X using the complex amplitude d, the function value qdiff(?, ?), and the line integral value ?Lqdiff(??, ?)d?? are included.

Abstract: A sound signal downmixing method includes a step of obtaining, for each of two channels, a signal obtained by adding an input sound signal of one channel to a signal obtained by delaying an input sound signal of the other channel and multiplying the delayed input sound signal by a weight value as a delayed crosstalk-added signal of the one channel, a step of obtaining preceding channel information and a left-right correlation value, and step of obtaining a downmix signal by performing weighted addition on the input sound signals of the two channels based on the left-right correlation value and the preceding channel information such that more of a signal derived from an input sound signal of a preceding channel among the signals derived from the input sound signals of the two channels is included as the left-right correlation value becomes larger.

Abstract: The stereo decoding unit 220 performs steps S222-1 and S222-2 below (step S222). The stereo decoding unit 220 obtains a signal concatenating a sum signal (a signal configured by addition of sample values of corresponding samples) of the monaural decoded sound signal for the section Y and the additional decoded signal for the section Y and the additional decoded signal for the section X, as a decoded downmix signal for the section Y+X (step S222-1) instead of step S221-1 performed by the stereo decoding unit 220 of the first embodiment and obtains and outputs the decoded sound signals of the two channels from the decoded downmix signal obtained at step S222-1 by the upmix processing using the characteristic parameter obtained from the stereo code CS, using the decoded downmix signal obtained at step S222-1 instead of the decoded downmix signal obtained at step S221-1 (step S222-2).

Abstract: A sound signal purification method includes an n-th channel signal purification step of obtaining, for each frame and for each corresponding sample t with respect to each channel n, a sequence based on a value ˜xn(t)=(1??n)×{circumflex over (?)}xn(t)+?n×{circumflex over (?)}xM(t) obtained by adding a value ?n×{circumflex over (?)}xM(t) obtained by multiplying an n-th channel purification weight ?n by a sample value {circumflex over (?)}xM(t) of the monaural decoded sound signal {circumflex over (?)}XM and a value (1??n)×{circumflex over (?)}xn(t) obtained by multiplying a value (1??n) obtained by subtracting the n-th channel purification weight ?n from 1 by a sample value {circumflex over (?)}xn(t) of the n-th channel decoded sound signal {circumflex over (?)}Xn, as the n-th channel purified decoded sound signal ˜Xn.

Abstract: A coding method and a decoding method are provided which can use in combination a predictive coding and decoding method which is a coding and decoding method that can accurately express coefficients which are convertible into linear prediction coefficients with a small code amount and a coding and decoding method that can obtain correctly, by decoding, coefficients which are convertible into linear prediction coefficients of the present frame if a linear prediction coefficient code of the present frame is correctly input to a decoding device.

Abstract: There is provided a technology that improves, in a case where there is a sound signal obtained from a different code that is different from a code from which a decoded sound signal is obtained and that is derived from the same sound signal, the decoded sound signal by using the sound signal obtained from the different code. A signal (hereinafter, referred to as a decoded sound common signal) obtained by downmixing a decoded sound signal of each channel is subjected to signal purification using a monaural decoded sound signal to thereby generate a purified common signal, and in each channel, a signal obtained by upmixing the decoded sound common signal is subtracted from the decoded sound signal and a signal obtained by upmixing the purified common signal is added thereto, to thereby generate a purified decoded sound signal.

Abstract: There is provided a technology that improves, in a case where there is a sound signal obtained from a different code that is different from a code from which a decoded sound signal is obtained and that is derived from the same sound signal, the decoded sound signal by using the sound signal obtained from the different code. A signal (hereinafter, referred to as a decoded sound common signal) obtained by downmixing a decoded sound signal of each channel is subjected to signal purification using a monaural decoded sound signal to thereby generate a purified common signal, and in each channel, the decoded sound common signal is subtracted from the decoded sound signal and the purified common signal is added thereto, to thereby generate a purified decoded sound signal.

Abstract: There is provided such embedded encoding that the algorithmic delay of stereo coding/decoding is not larger than that of monaural coding/decoding. An encoding device (100) encodes a sound signal having a plurality of channels. A stereo encoding unit (110) obtains and outputs a stereo code representing a characteristic of difference between channels of the sound signal. A downmix unit (150) obtains a signal by mixing the sound signal as a downmix signal. A monaural encoding unit (120) encodes the downmix signal by an encoding scheme that includes processing of applying a window having overlap between frames to obtain and output a monaural code. An additional encoding unit (130) encodes a part of the downmix signal for a section corresponding to the overlap between a current frame and an immediately following frame to obtain and output an additional code.

Abstract: For each frame, an n-th channel compensated decoded sound signal ˜X?n is obtained that is a signal obtained by compensating a high frequency of an n-th channel purified decoded sound signal ˜Xn obtained by performing signal processing in a time domain on an n-th channel decoded sound signal {circumflex over (?)}Xn (n is each integer of 1 or more and N or less) that is a decoded sound signal of each channel of stereo obtained by decoding a stereo code CS.

Abstract: There is provided a technology that improves, in a case where there is a sound signal obtained from a different code that is different from a code from which a decoded sound signal is obtained and that is derived from the same sound signal, the decoded sound signal by using the sound signal obtained from the different code. A signal (hereinafter, referred to as an upmixed common signal) obtained by upmixing a decoded sound common signal obtained by downmixing a decoded sound signal of each channel is subjected to signal purification using a signal (hereinafter, referred to as an upmixed monaural decoded sound signal) obtained by upmixing a monaural decoded sound signal to thereby generate a purified upmixed signal, and in each channel, the upmixed common signal is subtracted from the decoded sound signal and the purified upmixed signal is added thereto, to thereby generate a purified decoded sound signal.

Abstract: An autocorrelation calculation unit 21 calculates an autocorrelation RO(i) from an input signal. A prediction coefficient calculation unit 23 performs linear prediction analysis by using a modified autocorrelation R?O(i) obtained by multiplying a coefficient wO(i) by the autocorrelation RO(i). It is assumed here, for each order i of some orders i at least, that the coefficient wO(i) corresponding to the order i is in a monotonically increasing relationship with an increase in a value that is negatively correlated with a fundamental frequency of the input signal of the current frame or a past frame.

Abstract: There is provided a technology that improves, in a case where there is a sound signal obtained from a different code that is different from a code from which a decoded sound signal is obtained and that is derived from the same sound signal, the decoded sound signal by using the sound signal obtained from the different code. A signal (hereinafter, referred to as an upmixed common signal) obtained by upmixing a decoded sound common signal obtained by downmixing a decoded sound signal of each channel is subjected to signal purification using a signal (hereinafter, referred to as an upmixed monaural decoded sound signal) obtained by upmixing a monaural decoded sound signal to thereby generate a purified upmixed signal, and in each channel, the upmixed common signal is subtracted from the decoded sound signal and the purified upmixed signal is added thereto, to thereby generate a purified decoded sound signal.

Abstract: A technology of accurately coding and decoding coefficients which are convertible into linear prediction coefficients even for a frame in which the spectrum variation is great while suppressing an increase in the code amount as a whole is provided. A coding device includes: a first coding unit that obtains a first code by coding coefficients which are convertible into linear prediction coefficients of more than one order; and a second coding unit that obtains a second code by coding at least quantization errors of the first coding unit if (A?1) an index Q commensurate with how high the peak-to-valley height of a spectral envelope is, the spectral envelope corresponding to the coefficients which are convertible into the linear prediction coefficients of more than one order, is larger than or equal to a predetermined threshold value Th1 and/or (B?1) an index Q? commensurate with how short the peak-to-valley height of the spectral envelope is, is smaller than or equal to a predetermined threshold value Th1?.