Digital audio encoding
To represent the vibrations of sound in digital form, the amplitude of the sound signal is measured at each specific moment of the sound.
Since the waveform of sound is inherently continuous, for its accurate digital display it is necessary to measure the amplitude an infinite number of times per second and divide the amplitude scale by an infinite number of gradations. In reality, the number of measurements per second (sample rate) typically ranges from 10,000 to 96,000. Currently, the most common sample rates are 44100 Hz (the standard for CD-audio) and 48000 Hz (the main standard for CD-audio). DAT). The number of amplitude gradations (resolution) is generally taken equal to 28, 216, or 224 (depending on the number of bits allocated for this information).
Of course, distortion is unavoidable when sampling a continuous signal. The lower the sample rate and / or resolution, the closer the output waveform will be to rectangular. In this case, high-frequency distortions arise, which are partially suppressed by filters installed at the DAC output.
Digitized audio requires a large amount of memory. In fact, at a standard 44100 Hz sample rate and 16-bit resolution, the audio material (stereo) for one minute would be 10,584,000 bytes (approximately 10.09 MB). Also, the sound files are very poorly compressed by standard archive programs (zip, arj, etc.). Therefore, there are special compression algorithms for them. For example, a WAV file compressed with ADPCM takes about four times less space. However, distortion may occur. Therefore, it is better not to use audio compression algorithms in professional work.
