Abstract: An encoder and a decoder are provided that are capable of reproducing a frequency-domain envelope sequence that provides high approximation accuracy around peaks caused by the pitch period of an audio signal by using a small amount of code. An encoder of the present invention comprises a periodic-combined-envelope generating part and a variable-length coding part. The periodic-combined-envelope generating part generates a periodic combined envelope sequence which is a frequency-domain sequence based on a spectral envelope sequence which is a frequency-domain sequence corresponding to a linear predictive coefficient code obtained from an input audio signal and on a frequency-domain period. The variable-length coding part encodes a frequency-domain sequence derived from the input audio signal. A decoder of the present invention comprises a periodic-combined-envelope generating part and a variable-length decoding part.

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: Performance of an encoding process and a decoding process for a sound signal is enhanced. A representative value calculating part 110 calculates, for each frequency section by a plurality of samples fewer than the number of frequency samples of a sample sequence of a frequency domain signal corresponding to an input acoustic signal, from the sample sequence of the frequency domain signal, a representative value of the frequency section from sample values of samples included in the frequency section, for each of predetermined time sections. A signal companding part 120 obtains, for each of the predetermined time sections, a frequency domain sample sequence obtained by multiplying a weight according to a function value of the representative value by a companding function for which an inverse function can be defined and each of the samples corresponding to the representative value in the sample sequence of the frequency domain signal, as a sample sequence of a weighted frequency domain signal.

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: 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 sampling frequency required for symbol timing recovery is made smaller than that of the related art. A receiver 1 performs visible light communication with a transmitter 8. A received signal generating unit 112 measures an intensity of an electrical signal corresponding to an optical signal received from a transmitter 8 at a predetermined time interval to generate a sequence of received signals. A parameter estimation unit 12 uses a distribution of received signals estimated from the sequence of received signals to estimate any one or more parameters of a maximum luminance value, a synchronization shift, and a steady noise level, the one or more parameters including at least the maximum luminance value.

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: 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 log spectral envelope sequence L0, L1, . . . , LN?1 and an envelope code for the log spectral envelope sequence L0, L1, . . . , LN?1 are obtained. The log spectral envelope sequence L0, L1, . . . , LN?1 is an integer value sequence corresponding to binary logarithms of respective sample values of a spectral envelope sequence and is an integer value sequence whose total sum is 0. For a quantized spectral sequence {circumflex over (?)}X0, {circumflex over (?)}X1, . . . , {circumflex over (?)}XN?1, a smoothed spectral sequence ˜X0, ˜X1, . . . , ˜XN?1 is obtained by: for {circumflex over (?)}Xk with Lk being a positive value, adopting {circumflex over (?)}Xk with Lk digits from its least significant digit removed as ˜Xk; for {circumflex over (?)}Xk with Lk being a negative value, adopting {circumflex over (?)}Xk with ?Lk digits added to its least significant digit in accordance with a predefined rule as ˜Xk; and when Lk is 0, adopting {circumflex over (?)}Xk as ˜Xk.

Abstract: As pitch enhancement processing, a pitch enhancement apparatus obtains, for a time segment judged to be a time segment including a signal that is a consonant, for each time of the time segment, as an output signal, a signal including a signal obtained by adding a signal, which was obtained by multiplying a signal at a time that is an earlier time than the time by the number of samples T0 corresponding to a pitch period of the time segment, the pitch gain ?0 of the time segment, a predetermined constant B0, and a value that is greater than 0 and less than 1, and a signal at the time.

Abstract: In encoding, a frequency-domain sample sequence derived from an acoustic signal is divided by a weighted envelope and is then divided by a gain, the result obtained is quantized, and each sample is variable-length encoded. The error between the sample before quantization and the sample after quantization is quantized with information saved in this variable-length encoding. This quantization is performed under a rule that specifies, according to the number of saved bits, samples whose errors are to be quantized. In decoding, variable-length codes in an input sequence of codes are decoded to obtain a frequency-domain sample sequence; an error signal is further decoded under a rule that depends on the number of bits of the variable-length codes; and from the obtained sample sequence, the original sample sequence is obtained according to supplementary information.

Abstract: A coding technology that efficiently codes an input sound signal irrespective of the characteristics thereof and can obtain a decoded sound signal that sounds less artificial to a listener. A coding method codes an input sound signal frame by frame of a predetermined time segment by a selected coding processing from a plurality of types of coding processing in the frequency domain, the coding method makes it possible for a selection unit to select coding processing which is different from the coding processing of the preceding frame as coding processing of the present frame if at least one of the magnitude of the energy of high frequency components of the input sound signal of the preceding frame and the magnitude of the energy of high frequency components of the input sound signal of the present frame is smaller than or equal to a predetermined threshold value.

Abstract: A coding technology that efficiently codes an input sound signal irrespective of the characteristics thereof and can obtain a decoded sound signal that sounds less artificial to a listener. A coding method codes an input sound signal frame by frame of a predetermined time segment by a selected coding processing from a plurality of types of coding processing in the frequency domain, the coding method makes it possible for a selection unit to select coding processing which is different from the coding processing of the preceding frame as coding processing of the present frame if at least one of the magnitude of the energy of high frequency components of the input sound signal of the preceding frame and the magnitude of the energy of high frequency components of the input sound signal of the present frame is smaller than or equal to a predetermined threshold value.

Abstract: The present invention aims to encode and decode a sequence of integer values by substantially assigning the number of bits of a decimal fraction value per sample. An integer converter 11 selects M selected integer values from L input integer values for a set of the L input integer values and obtains J-value selection information that specifies which of the L input integer values the M selected integer values are. Furthermore, the integer converter 11 obtains one converted integer value by reversibly converting the M selected integer value and an integer value corresponding to the J-value selection information. An integer encoder 12 encodes the converted integer value to obtain a code.

Abstract: In encoding, a frequency-domain sample sequence derived from an acoustic signal is divided by a weighted envelope and is then divided by a gain, the result obtained is quantized, and each sample is variable-length encoded. The error between the sample before quantization and the sample after quantization is quantized with information saved in this variable-length encoding. This quantization is performed under a rule that specifies, according to the number of saved bits, samples whose errors are to be quantized. In decoding, variable-length codes in an input sequence of codes are decoded to obtain a frequency-domain sample sequence; an error signal is further decoded under a rule that depends on the number of bits of the variable-length codes; and from the obtained sample sequence, the original sample sequence is obtained according to supplementary information.

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: Provided is pitch enhancement processing having little unnaturalness even in time segments for consonants, and having little unnaturalness to listeners caused by discontinuities even when time segments for consonants and other time segments switch frequently. A pitch emphasis apparatus carries out the following as the pitch enhancement processing: for a time segment in which a spectral envelope of a signal has been determined to be flat, obtaining an output signal for each of times in the time segment, the output signal being a signal including a signal obtained by adding (1) a signal obtained by multiplying the signal of a time, further in the past than the time by a number of samples T0 corresponding to a pitch period of the time segment, a pitch gain ?0 of the time segment, a predetermined constant B0, and a value greater than 0 and less than 1, to (2) the signal of the time.

Abstract: Provided is pitch enhancement processing having little unnaturalness even in time segments for consonants, and having little unnaturalness to listeners caused by discontinuities even when time segments for consonants and other time segments switch frequently. A pitch emphasis apparatus obtains an output signal by executing pitch enhancement processing on each of time segments of a signal originating from an input audio 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: An encoding apparatus comprising an encoding part 14 performing encoding by an encoding process in which bits are preferentially assigned to a low side to obtain a spectrum code, the encoding apparatus further comprising: a fricative sound judging part 12 judging whether a sound signal is a hissing sound or not; and a fricative sound adjusting part 13 obtaining, if the fricative sound judging part 12 judges that the sound signal is a hissing sound, what is obtained by exchanging all or a part of a low-side frequency spectrum sequence existing on a lower side than a predetermined frequency in a frequency spectrum sequence of the sound signal for all or a part of a high-side frequency spectrum sequence existing on a higher side of the predetermined frequency in the frequency spectrum sequence, as an adjusted frequency spectrum sequence, the number of all or the part of the high-side frequency spectrum sequence being the same as the number of all or the part of the low-side frequency spectrum sequence; wherein t