摘要:
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.
摘要:
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.
摘要:
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.
摘要:
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.
摘要:
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.
摘要:
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.
摘要:
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.
摘要:
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.
摘要:
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. When the difference value is not positive, the quantization-candidate calculator 14 sets 0 as an quantization candidate corresponding to the sample. A vector quantizer 15 jointly vector-quantizes a plurality of quantization candidates corresponding to a plurality of samples to obtain a vector quantization index.
摘要:
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. When the difference value is not positive, the quantization-candidate calculator 14 sets 0 as an quantization candidate corresponding to the sample. A vector quantizer 15 jointly vector-quantizes a plurality of quantization candidates corresponding to a plurality of samples to obtain a vector quantization index.