The algorithm used to compress and decompress files is called CODEC (acronym for compression / decompression). “Codec” is software that tells the computer which mathematical operations it must manipulate to compress them and which ones to perform to show them compressed.
Instead, the “format” is a kind of file that contains the codec and integrates it with the system.

Sounds are digitally recorded using a technique called “sampling”: the sound wave is divided into many pieces called samplers.

The quality of a digital audio track depends on:

– sampling frequency, measured Hertz (Hz, number of samples per second). A frequency at 11.025 Hz is suitable for recording voice, one at 22.050 Hz (medium quality) is suitable for recording a tape and one at 44,100 Hz for recording in CD quality. Reducing the sample rate leads to loss of quality.

– from termination, ie. the number of bits used (8.16, 24 to 32) for each ciampione (with 8 bits = 1 byte for 256 options, 16 bits = 2 bytes for 256 * 256 = 65,536 values ​​in the levels, and so on). Converting 16-bit to 8-bit samples cuts the original file in half, but also reduces the quality of the music.
– the number of channels: mono (1) or stereo (2).

bit rate: the product of these three elements: frequency, resolution, and number of channels are defined as bit rate, ie bits per second or bps. From this it can be deduced that every second there are 44,100 recorded values ​​which are then multiplied by the 2 stereo sound channels which are multiplied by 16 as the recording takes place in 16 bits (corresponding to 2 bytes). Then we get:

The bit rate for songs on audio CDs = 44,100 * 16 bit * 2 = 1,411.2 kbps (~ 10.6 MByte per minute 44,100 * 2 byte * 2 * 60)
The bit rate of an audio recording = 22,050 * 8 * 1 = 176.4 Kbps (~ 1.3 MByte per minute = 22,050 * 1 byte * 1 * 60)

Accordingly, compressing by reducing the total length of the file will reduce the average length of the subsequent ones, ie. it will reduce the average bit rate. Therefore, in these cases, the average bit rate becomes the index of the compression scope. For example, if the source file had a bit rate of 1,411 Kbit / if the compressed file had an average bit rate of 320 Kbit / s, we would have reduced by a factor of approx. 4.5.
Loss compression compromises the loss of information and the size of the final file, while a lossless compression must balance the size of the final file with the execution times of the algorithm.

The most popular lossless audio formats are:

-WAV sampling, Wave file (Waveform Extension), where wave means wave: standard format for audio files in the Windows audio sampling environment; It has large dimensions as it manages sampling frequencies of up to 44.1 kHz, 48 kHz and now also 96 and even 192 kHz, resolution of up to 32 linear bits and allows to store stereo or surround signals with a number
Unlimited in a single speaker file (equivalent to so many channels). The wave format is nothing more than digital recording of real sounds, sounds that have had
originates from a source external to the PC. In a WAV piece of music drums, piano, guitar, bass or
voice is heard in the same way, regardless of the PC to which the file is heard (to
obviously with the same acoustic quality of the hardware components).
-Aif (Apple Audio Interchange File Format or AIFF) similar to WAV format, is a format that generates good sound quality, is compatible with many browsers and does not require plugins. to Apple’s AIFF format. The Au format also manages more efficient quantization methods that allow a reduction in the amount of data by even 4 times the original value at the cost of a modest loss of quality.
-APE (Monkey Audio; .ape): Lost raw format that allows you to reduce the space occupied
our music about 50% (in some cases even more) without loss of quality. This way an album there
wav format has approx. 600 MB, has an average of 300 MB (much more than about 100 MB a
high bit rate and 60 mpc of an mp3, but the quality is identical to the original); I say, on average, because there is
certain types of music where the level of compression is even higher. To listen to songs in this format,
you can use plugins for WinAmp or, better yet, a player that integrates the native as
Foobar 2000. Right now it’s probably the best lossless codec considering a balance between
speed and compression (click here for lossless comparison table).