Computer sound processing
MP3 format
The audio compression method as well as the compressed audio file format proposed by the international organization MPEG (Moving Pictures Experts Group) is based on perceptual audio encoding. Work on creating efficient audio coding algorithms (originally for digital broadcasting) started in 1987 as part of the European project EUREKA (code EU147). The result is an extremely powerful algorithm, standardized as ISO-MPEG Audio Layer-3 (IS11172-3 and IS13818-3). It allows you to achieve a compression ratio of up to 12 (sometimes even more) and without a noticeable loss in sound quality. There are three levels of compression for stereo signals:
• MPEG Layer-1 – 1: 4 compression ratio with 384 Kbps data stream;
• MPEG Layer-2 – 1: 6.1: 8 at 256–192 Kbps;
• MPEG Layer-3 (MP3) – 1: 10.1: 12 @ 128-112 Kbps.
The files corresponding to these three levels usually have extensions. mp1, mp2 and. mp3 respectively.
Initially, MPEG audio compression and restoration methods were intended only for hardware implementation using digital signal processors (DSP), but the performance of modern processors is sufficient to reproduce compressed audio in real time. So, to play files recorded in MP3 format, a Pentium processor with a clock frequency of 75 MHz is sufficient.
Such processors are not powerful enough to compress audio in real time, but there are software converters that compress audio files out of the box. Another more convenient way to work with the MPEG Audio format in Windows 95 is to use ACM Codec, automatic format converters that work at the system level. When installing such a converter, MPEG audio files can be given the extension. wav and work with them using any program that supports ACM, for example the standard media player (phonograph).
A compressed MP3 audio file actually offers the audio quality of a CD. There are losses, of course, but they are insignificant and sometimes completely invisible (more, given the possibility of putting 10 to 12 albums on a CD using this format).
Technology is next. You must first create a WAV file from the tracks on the CD. There are two possibilities for this: digitize the sound with a sound card or obtain audio data directly from a CD.
In the former case, the signal from the sound card is sent to the CD or line (line) input and a “normal” audio recording is made.
In the second, special programs are used that allow you to obtain a high-quality WAV file by extracting digital audio data directly from a CD using special functions of the CD drive, namely the Read Long command. Such a program is called a grabber, and hobbyists call the process itself “grabbing an audio disc” (capture trace copy).
The main disadvantage of the first way is the noise of the sound card itself, which is the louder, the cheaper the card. So, this method leads to a loss of quality already in the first step and is often quite noticeable.
