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: 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 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: 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: 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: An ACELP speech coding method according to ITU-T Recommendation G.729. When coding a random component vector, each of random component vector forming together the random codebook is formed of three or less pulses having a unit amplitude for each 6f a pair of subframes which form together a frame. The positions of the pulses are determined from a plurality of predetermined positions which a pulse can assume in a subframe so that distortion is minimized. The method allows speech coding at a lower bit rate.

Abstract: Representative vectors z.sub.1i and z.sub.2j are selected from codebooks CB1 and CB1, respetively, and multiplied by weighting coefficient vectors w.sub.1 and w.sub.2 of the same number of dimensions as those of the representative vectors, whereby weighted representative vectors z.sub.1i w.sub.1 and z.sub.2j w.sub.2 are generated. These weighted representative vectors are vector combined into a combined vector y.sub.ij, and a combination of the representative vectors is selected by a control part in such a manner as to minimize the distance between the combined vector y.sub.ij and an input vector X. The weighting coefficient vectors w.sub.1 and w.sub.2 each have a maximum component in a different dimension and are selected so that the sum of diagonal matrixes W.sub.1 and W.sub.2 using components of the weighting coefficient vectors as their diagonal elements becomes a constant multiple of the unit matrix.

Abstract: In a speech coding method of the present invention, initially, a plurality of samples of speech data are analyzed by a linear prediction analysis and thereby prediction coefficients are calculated. Then, the prediction coefficients are quantized, and the quantized prediction coefficients are set in a synthesis filter. Moreover, a pitch period vector is selected from an adaptive codebook in which a plurality of pitch period vectors are stored, and the selected pitch period vector is multiplied by a first gain which is obtained, at the same time, with a second gain. In addition, a noise waveform vector is selected from a random codebook in which a plurality of the noise waveform vectors are stored, and is multiplied by a predicted gain and the second gain. Then, the speech vector is synthesized by exciting the synthesis filter with the pitch period vector multiplied by the first gain, and with the noise waveform vector multiplied by the predicted gain and the second gain.

Abstract: Representative vectors Z1i and Z2j are selected from code-books codebooks CB1 and CB1 CB2, respectively, and multiplied by weighting coefficient vectors w1 and w2 of the same number of dimensions as those of the representative vectors, whereby weighted representative vectors Z1iw1 and Z2jw2 are generated. These weighted representative vectors are vector combined into a combined vector yij, and a combination of the representative vectors is selected by a control part in such a manner as to minimize the distance between the combined vector yij and an input vector X. The weighting coefficient vectors w1 and w2 each have a maximum component in a different dimension and are selected so that the sum of diagonal matrixes W1 and W2 using components of the weighting coefficient vectors as their diagonal elements becomes a constant multiple of the unit matrix.