Methods used to compress digital audio.
Information compression methods when working with sound.
The larger the memory capacity of the WT card, the more realistic the sound will be (as more samples are stored in memory, they are recorded at a higher resolution). The General MIDI standard describes more than 200 instruments; To store your sound samples (tables), at least 8 MB of memory is required (at least 20 KB for each sample).
Known WF (Wave Form) method of sound generation, based on the transformation of sounds into complex mathematical formulas and the subsequent application of these formulas to control a powerful processor in order to reproduce the sound; from WF synthesis expect an even better reality (relative to FM and WT technologies) of musical instruments playing with limited volumes of sound files.
To reduce data flow, other analog (non-PCM) encoding methods are used. For example, a coding technique based on known characteristics of an analog signal is known to significantly reduce the amount of data stored; with the so-called -The encoding of the analog signal is converted into a digital code determined by the logarithm of the magnitude of the signal (and not by its linear transformation). The disadvantage of this method is the need to have a priori information about the characteristics of the original signal.
Conversion methods are known that do not require a priori information about the original signal. When differential pulse code modulation (DPCM, Differential Pulse Code Modulation) persists single signal difference between current and previous levels (the difference requires a digital representation of fewer bits than the full amplitude value). With delta modulation (DM, delta modulation), each sample consists of a single bit, which determines the sign of the change in the original signal (increase or decrease); Delta modulation requires a higher sample rate. Differential PCM technologies involve the accumulation of errors over time, so special measures are taken to periodically calibrate the ADC.
The most common when recording received audio is adaptive pulse code modulation (ADPCM, Adaptive Pulse Code Modulation), using 8- or 4-bit coding for the difference signals. The technology was first applied by Creative Labs and provides data compression up to 4: 1.
However, other audio information compression / decompression methods (software) are often used; Among them, the most popular lately is the MP3 format developed by Fraunhofer IIS (Fraunhofer Institute Integrierte Schaltungen, www.iis.fhg.de) and by THOMSON (the full specification of the MP3 format is published on the website www.mp3tech.org ). The full name of the MP3 standard sounds like MPEG-Audio Layer-3 (where MPEG is the essence of the Moving Picture Expert Group, not to be confused with the MPEG-3 standard designed for use in high definition television).
MP3 encoding of data occurs through the allocation of independent independent data blocks: frames. To do this, the original signal during encoding is divided into equal length parts, called frames, and encoded separately (to further reduce the amount of data, compression is applied using the Huffman algorithm); When decoding, the signal is formed from a sequence of decoded frames. The encoding process takes a significant amount of time; decoding (during playback) is done on the fly.
The MP3 format provides the best sound quality with the smallest file size. This is achieved by taking into account the peculiarities of human hearing, including the effect of masking a weak signal from one frequency range with a stronger signal from an adjacent range (when it occurs) or a strong signal from the previous frame, causing a temporary decrease in the ear’s sensitivity to the signal of the current frame (in other words, minor sounds are eliminated, which are not heard by the human ear due to the presence at this / previous moment of another – louder sound). It also takes into account the inability of most people to distinguish signals that are below a certain power level, different for different frequency ranges. This process is called adaptive coding, and it saves at least sound details that are meaningful from the point of view of human perception. The compression ratio (hence the quality) is not determined by the MP3 format, but by the width of the data stream during encoding.
