Coding sound information

Any processes of reality can be transformed into digital form. Thus, the coding of audio information by means of computers is performed according to the following scheme:

- air vibrations are recorded by sensitive instruments;

- they are converted into an electric current, in which the frequency (amplitude) changes accordingly;

- the received current is digitized, that is, it is sampled (sometimes it is said that binary encoding of audio information takes place).

The received electronic analogue of the initial sound stream is more qualitative, the higher the sampling frequency at sampling and the depth of coding.

In other words, encoding audio information is the process of converting a familiar analog signal into a digital one, intended for subsequent processing on the respective devices. Let us consider in more detail the stages and ways of digitizing the sound.

Time-domain sampling is the basis for digitization. According to Kotel'nikov's theorem, an analog electrical signal can be transformed into a digital form by reading with a certain step a continuous series of values of its amplitude. The frequency of such readings should at least double the limiting value of the frequency of the main signal. If it is necessary to digitize the analog "source" with an operating frequency of 0-20 kHz, the sample should be sampled no less than 40 thousand times per second (40 kHz). Sampling indicates the number of measurements per second of the original analog signal (sampling, sampling frequency). With the growth of samples, not only the quality, but also the volume of the received data stream increases.

Also, encoding of audio information can be performed in other ways. As, for example, digitization by means of non-uniform quantization, sometimes called logarithmic. When using it, the entire amplitude scale is conventionally divided into sections with high and low values. Further encoding of audio information occurs by applying a large number of quantization levels in areas with a small amplitude value (and vice versa). However, we note that the total number of levels remains the same as in the homogeneous quantization method (PCM).

A completely different approach is implemented in an alternative coding method. It is called "differential pulse-code modulation" (DPCM). In this way, the quantization is not the immediate amplitude of the signal, but its relative values. As a result, it is possible to reduce the amount of data occupied by the data, since the mechanism for predicting the subsequent samples of the original signal is involved.

Coding and processing of sound information, described in this paper, presupposes the need to perform an "analog-digit" transformation. This process is carried out using an ADC (analog-to-digital converter). With the operation of this device, every owner of a computer equipped with a sound card faces each day (in this case, there is a reverse process - obtaining an analog signal from the digital stream).

The functions of the ADC are as follows:

- In limiting the bandwidth of the transmitted frequencies. With the help of filters, the components of the signal whose frequency is more than half the sampling frequency are cut off (the reason was described earlier).

- Sampling of amplitude values at certain intervals. As a result, the analog signal is represented by a sequence of single bits of different intensity (sampling).

- Replacement of the values of the obtained bits by their nearest values from a fixed set (quantization).

- Conversion of each quantized value by a conditional number of the quantization level (each value has its own serial number). This is the last stage of digitization.