Nippon Telegraph and Telephone Corporation

Akitoshi Kataoka

Takeshi Mori

Yusuke Hiwasaki

Hitoshi Ohmuro

Shinji Hayashi

Takehiro Moriya

Kimitaka Tsutsumi

Shigeaki Sasaki

Kazunori Mano

Sachiko Kurihara

Satoshi Miki

Jotaro Ikedo

Hitoshi Omuro

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.

Coding method, decoding method, apparatuses thereof, programs thereof, and recording medium

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.

Acoustic signal packet communication method, transmission method, reception method, and device and program thereof

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.

Sound packet reproducing method, sound packet reproducing apparatus, sound packet reproducing program, and recording medium

Input speech is coded in an encoder (), the coded speech is decoded in a decoder (), compensatory speech which compensates the speech of the current frame is generated in a compensatory speech generating part () 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 (), and as many identical packets as the number specified by the duplication level is generated for the coded speech in a packet generating part (), and the packets are transmitted, thereby reducing the possibility that packet loss will occur at the receiving end.

Sound packet transmitting method, sound packet transmitting apparatus, sound packet transmitting program, and recording medium in which that program has been recorded

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.

Speech coding method

Vector coding method, encoder using the same and decoder therefor

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. Consequently, speech data comprising a plurality of samples are coded as a unit of a frame operation. Furthermore, the predicted gain multiplied by the noise waveform vector which is selected in a subsequent frame operation, is predicted based on the current noise waveform vector which is multiplied by the predicted gain and the second gain at the current frame operation, and also the previous waveform vector which is multiplied by the predicted gain and the second gain in the previous frame operation.

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Nerima-ku, Japan

Akitoshi Kataoka