The working principle of audio equipment

(1) Sampling

The signal waveform whose amplitude changes continuously with time is sampled at a certain time interval to form a pulse sequence that is not continuous in time, which is called sampling. This time interval is called the sampling period, denoted as Ts, and the corresponding sampling frequency fs=1/Ts.

(2) Quantification

The amplitude dynamic range of the analog signal is divided into several levels at equal intervals, and the signal level of the sampled output is classified into a close value according to the principle of rounding, which is called quantization.

(3) Encoding

The process of transforming the value obtained by sampling and quantization into binary digits is called encoding. In digital audio, 16-bit (bit) numbers are usually used to represent a value, that is, the number of quantization bits n=16.

(4) Error correction coding

Since CDs and cassette tapes are damaged beyond the allowable range during the production and use process, the digital signal read is different from the original recorded signal. Therefore, measures to correct the error code are taken.

(5) Modulation

The analog audio signal is sampled, quantized, coded and CIRC error correction coded to form a digital signal, and it is not suitable to be directly recorded on a record or tape. Because all 16 bits may be 0 or 1 in the data stream, the signal will be extremely unstable when read from the record or tape, and the servo system will also be unstable.

(6) Frame structure

The digital signal is based on a character. If it is shifted by 1 bit, the signal level represented by the character will change. For this reason, the recording signal is divided into small blocks, and try to determine the dividing line between the blocks, such blocks are called frames.