Abstract: A plurality of samples are vector-quantized to obtain a vector quantization index and quantized values; bits are assigned in a predetermined order of priority based on auditory perceptual characteristics to one or more sets of sample positions among a plurality of sets of sample positions, each set having a plurality of sample positions and being given an order of priority based on the auditory perceptual characteristics, the number of bits not being larger than the number of bits obtained by subtracting the number of bits used for a code corresponding to the vector quantization index from the number of bits assigned for the code corresponding to the vector quantization index; and index information indicating a group of coefficients that minimizes the sum of the error between the value of each sample included in each of the sets of sample positions to which the bits are assigned and the value obtained by multiplying the quantized value of each sample included in the set of sample positions by a coefficient co

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: In encoding, the number of bits to be assigned to codes corresponding to noise or a pulse sequence obtained according to prediction analysis applied to time series signals included in a predetermined time interval is switched according to whether an index that indicates a level of periodicity and/or stationarity of input time series signals satisfies a condition that indicates high periodicity and/or high stationarity or a condition that indicates low periodicity and/or low stationarity, to acquire the codes corresponding to the noise and the pulse sequence. In decoding, a decoding mode for codes corresponding to noise or a pulse sequence included in codes corresponding to a predetermined time interval is switched according to the same criterion as that described above to decode the codes corresponding to the noise or the pulse sequence to acquire noise or a pulse sequence corresponding to the predetermined time interval.

Abstract: An encoding technique encoding a sound signal at a low bit rate with reduced processing. The technique includes: an interval determination determining an interval T between samples corresponding to periodicity of an audio signal or an integer multiple of a fundamental frequency of the audio signal from a set S of candidates for the interval T; and a side information generating encoding the determined interval T to obtain side information. The interval determining determines the interval T from a set S of Y candidates (Y<Z) including Z2 candidates (Z2<Z) selected from among Z candidates for the interval T representable with the side information without depending on a candidate subjected to the interval determination in a previous frame a predetermined number of frames before the current frame and including a candidate subjected to the interval determination in the previous frame the predetermined number of frames before the current frame.

Abstract: When a number of samples which are less than a first reference value is a second reference value or less, a second encoding mode is selected. In the second encoding mode, when a difference value that is obtained by subtracting a value corresponding to the quantized normalization value from a value corresponding to the magnitude of each sample is positive and the sample is positive, the difference value is set as a quantization candidate corresponding to the sample; when the difference value is positive and the sample is negative, the sign of the difference value is reversed and the result is set as the quantization candidate corresponding to the sample; and a plurality of quantization candidates are jointly vector-quantized to obtain a vector quantization index. When the second encoding mode is not selected, a first encoding mode other than the second encoding mode is selected.

Abstract: A plurality of samples are vector-quantized to obtain a vector quantization index and quantized values; bits are assigned in a predetermined order of priority based on auditory perceptual characteristics to one or more sets of sample positions among a plurality of sets of sample positions, each set having a plurality of sample positions and being given an order of priority based on the auditory perceptual characteristics, the number of bits not being larger than the number of bits obtained by subtracting the number of bits used for a code corresponding to the vector quantization index from the number of bits assigned for the code corresponding to the vector quantization index; and index information indicating a group of coefficients that minimizes the sum of the error between the value of each sample included in each of the sets of sample positions to which the bits are assigned and the value obtained by multiplying the quantized value of each sample included in the set of sample positions by a coefficient co

Abstract: A quantized normalization value and a normalization-value quantization index corresponding to the quantized normalization value are obtained, the quantized normalization value being obtained by quantizing a normalization value that is a value representative of samples. If a difference value that is obtained by subtracting a value corresponding to the quantized normalization value from a value corresponding to a magnitude of a value of each sample is positive and if the value of each sample is positive, the difference value is set as a quantization candidate. If the difference value is positive and if the value of each sample is negative, a value obtained by inverting positive/negative of the difference value is set as the quantization candidate. The plurality of quantization candidates respectively corresponding to the plurality of samples are collectively vector-quantized, and a vector quantization index is thus obtained and output.

Abstract: In encoding, index information indicating a group of coefficients that minimizes the sum of the error between the value of each sample and the value obtained by multiplying the quantized value of the sample by a coefficient corresponding to the position of the sample, for all sample positions, among a plurality of groups of predetermined coefficients corresponding to the positions of the samples, is output. In decoding, a plurality of values corresponding to an input vector quantization index are obtained as decoded values corresponding to a plurality of sample positions; and, with the use of a group of predetermined coefficients corresponding to the plurality of sample positions and indicated by input index information, the values obtained by multiplying the decoded values and the coefficients, corresponding to the sample positions are output.

Abstract: A normalization value calculator 12 calculates a normalization value that is representative of a predetermined number of input samples. A normalization value quantizer 13 quantizes the normalization value to obtain a quantized normalization value and a normalization-value quantization index corresponding to the quantized normalization value. An quantization-candidate calculator 14 subtracts a value corresponding to the quantized normalization value from a value corresponding to the magnitude of each of the samples to obtain a difference value and, when the difference value is positive and the value of each of the samples is positive, sets the difference value as an quantization candidate corresponding to the sample. When the difference value is positive and the value of each of the samples is negative, the quantization-candidate calculator 14 reverses the sign of the difference value and setting the sign-reversed value as an quantization candidate corresponding to the sample.

Abstract: When acoustic signal packets are communicated over an IP communication network, data corresponding to an acoustic signal (acoustic signal corresponding data) has been included and transmitted in a packet different from a packet containing the acoustic signal. However, conventionally, a packet in which the acoustic signal corresponding data is to be included must be determined beforehand and cannot dynamically be changed. According to the present invention, the amount of delay of acoustic signal corresponding data with respect to an acoustic signal is contained in an acoustic signal packet as delay amount control information. Furthermore, the conditions of a communication network are detected from the number of packets lost in a burst loss or jitters and the number of the packets to be stored and the amount of delay at the receiving end are thereby determined.

Abstract: A high-quality decoded signal is synthesized. A coding method of the present invention includes a local decoding coefficient searching step. The local decoding coefficient searching step includes a replication determining sub-step, a candidate replication shift signal sequence generating sub-step, a distance calculating sub-step, and a minimum distance shift amount finding sub-step. The replication determining sub-step determines, for each source signal sequence to be coded, whether or not a candidate replication shift signal sequence is to be generated from a decoded signal sequence and outputs a replication determination flag. If the replication determination flag indicates that a candidate replication shift signal sequence is to be generated, the candidate replication shift signal sequence generating sub-step generates a candidate replication shift signal sequence for each predetermined candidate signal shift amounts.

Abstract: There is provided a coding technique capable of reducing the amount of computation in coding while maintaining the efficiency of the coding. The technique uses an input signal and one of a decoded signal decoded from a first code obtained by encoding the input signal and a decoded signal obtained during generation of the first code. A gain group set includes one or more gain groups including different numbers of values corresponding to gains. A gain group is allocated to each sample by using a predetermined method. The sample is multiplied by a gain identified by a value corresponding to each gain in the allocated gain group and a gain code indicating a gain that results in the smallest difference between the product and the input signal is output.

Abstract: A coding method with a small error is provided. In the coding method of the present invention, a normalization value obtained from an input signal is corrected for an error calculated from an input and output in vector quantization and is then quantized. The coding method includes a normalization stage of normalizing the input signal in accordance with the normalization value of the input signal, calculated in each frame; a dividing stage of dividing the normalized frame into divided input signal sequences in accordance with a predetermined rule; a vector quantization stage of applying vector quantization to the divided input signal sequences to generate a vector quantization index; and a normalization value correction stage of correcting the normalization value of the input signal for the error obtained from the input and output in the vector quantization stage.

Abstract: The present invention prevents a receiving buffer from becoming empty by: storing received packets in the receiving buffer; detecting the largest arrival delay jitter of the packets and the buffer level of the receiving buffer by a state detecting part; obtaining an optimum buffer level for the largest delay jitter using a predetermined table by a control part; determining, based on the detected buffer level and the optimum buffer level, the level of urgency about the need to adjust the buffer level; expanding or reducing the waveform of a decoded audio data stream of the current frame decoded from a packet read out of the receiving buffer by a consumption adjusting part to adjust the consumption of reproduction frames on the basis of the urgency level, the detected buffer level, and the optimum buffer level.

Abstract: Input speech is coded in an encoder (11), the coded speech is decoded in a decoder (12), compensatory speech which compensates the speech of the current frame is generated in a compensatory speech generating part (20) by using past decoded speech, the quality of the compensatory speech is evaluated by using the input speech and the compensatory speech and a duplication level is generated the value of which increases incrementally with decreasing speech quality evaluation value in a speech quality evaluating part (40), and as many identical packets as the number specified by the duplication level is generated for the coded speech in a packet generating part (15), and the packets are transmitted, thereby reducing the possibility that packet loss will occur at the receiving end.

Abstract: At the speech encoding end, upon generation of an fixed excitation vector, the shape of an excitation vector output from pulse excitation codebook 301 is identified in pulse excitation vector shape identifier 302, a dispersion vector used for excitation vectors of the shape is output from dispersion vector storage 304, and, in dispersion vector convolution processor 303, dispersion vector convolution processing of the excitation vector is performed. In particular, when a pulse excitation vector having a specific shape of high frequency of use is output from pulse excitation codebook 301, pulse excitation vector shape identifier 302 controls dispersion vector storage 304 in such a way that an additional dispersion vector prepared dedicated to the pulse excitation vector is output. By this means, it is possible to provide a technology that improves the quality of decoded speech and that decodes speech more natural and audible to the user.

Abstract: When acoustic signal packets are communicated over an IP communication network, data corresponding to an acoustic signal (acoustic signal corresponding data) has been included and transmitted in a packet different from a packet containing the acoustic signal. However, conventionally, a packet in which the acoustic signal corresponding data is to be included must be determined beforehand and cannot dynamically be changed. According to the present invention, the amount of delay of acoustic signal corresponding data with respect to an acoustic signal is contained in an acoustic signal packet as delay amount control information. Furthermore, the conditions of a communication network are detected from the number of packets lost in a burst loss or jitters and the number of the packets to be stored and the amount of delay at the receiving end are thereby determined.

Abstract: A first determiner 121 tentatively determines whether the current processing unit represents a stationary noise period, based on stationary properties of a decoded signal. Based on the tentative determination result and a determination result of the periodicity of the decoded signal, a second determiner 124 determines whether the current processing unit represents a stationary noise period, thereby distinguishing a decoded signal including a stationary speech signal such as a stationary vowel from stationary noise and correctly identifying the stationary noise period.

Abstract: The present invention carries out pre-selection on many LPC codevectors stored in an LSF codebook 101 using a weighted Euclidean distortion as a measure and carries out a full-code selection on the LPC codevectors left after the pre-selection using an amount of distortion in a spectral space as a measure. This makes it possible to improve the quantization performance of the LPC parameter vector quantizer and improve the quality of synthesized speech of the speech coder/decoder.

Abstract: The present invention prevents a receiving buffer from becoming empty by: storing received packets in the receiving buffer; detecting the largest arrival delay jitter of the packets and the buffer level of the receiving buffer by a state detecting part; obtaining an optimum buffer level for the largest delay jitter using a predetermined table by a control part; determining, based on the detected buffer level and the optimum buffer level, the level of urgency about the need to adjust the buffer level; expanding or reducing the waveform of a decoded audio data stream of the current frame decoded from a packet read out of the receiving buffer by a consumption adjusting part to adjust the consumption of reproduction frames on the basis of the urgency level, the detected buffer level, and the optimum buffer level.