| 摘要 |
PURPOSE:To obtain a restored signal close to an original signal waveform, by setting the sampling period newly based on the result of detection in the state of change of the signal on a time axis. CONSTITUTION:A sound signal having <=4kHz frequency band by a low-pass filter LPEr is converted into a digital signal of a specified number of bits (taken as 8 bits for the explanation hereinafter) at an AD converter ADC. The specified signal processing is executed for the digital signal outputted from the AD converter ADC under the control of a control circuit CCT, and the data of amplitude is combined with a numeral relating to the zero point interval and stored in a memory M2. The data set comprising the set of information relating to the amplitude and the zero point interval is read out and given to a DA converter DAC in the stored order in the memory M2, the amplitude is converted into the amplitude li of analog quantity, the information relating to the zero point interval is converted into electric amoung gammai of analog quantity, and they are given to an interpolation circuit CP. |
