Digital audio encoding

Digital audio encoding

Digital audio encoding

In fact, one or another digital form of representation of analog audio signals is already a coding method – a sequence of numbers that describes an analog audio signal is itself a digital code.

Digital Audio Encoding

However, the encoding that we are going to talk about now is something else. Now let’s look at the methods of encoding digital audio signals.

A digitized audio signal “in its pure form” is a fairly accurate, but not the most compact, way of recording the original analog signal.

Judge for yourself. To obtain complete information about the original analog signal in the frequency range 0-20 kHz (in the audible frequency range), the analog signal must be sampled at a frequency of at least 40 kHz. Therefore, the CD – DA standard (the standard for recording data on audio CDs familiar to all) establishes the following encoding parameters: recording of two or one channel in PCM format with a sampling frequency of 44.1 kHz and a 16-bit quantization bit depth. One hour of music in this format takes up approximately 600 MB of space (60 minutes * 60 seconds * 2 channels * 44100 samples per second * 2 bytes per sample = approximately 605 MB). Taking into account that, for example, the music collection of an ordinary music lover may have 5,000 tracks with an average length of about 3 minutes each, the amount of memory required to store it in its original digital form is quite significant. Awesome. Therefore, storing relatively large amounts of audio data, ensuring fairly good sound quality, requires the use of various “tricks” to compress the data.

In general, all existing methods for encoding audio information can be conditionally divided into only two types.

1. Lossless data compression (“Lossless Encoding”) is a method of encoding (compacting) digital audio information, which enables one hundred percent recovery of the original data from the compressed transmission (the term ” original data “here means the original form of the digitized audio data). This method of data compression is used in cases where one hundred percent absolute preservation of the quality of the original audio data is required. Lossless compression algorithms that exist today can reduce the volume of data occupied by 20-50% and at the same time guarantee a 100% recovery of the original digital material from the compressed data. The operating mechanisms of such encoders are similar to the operating mechanisms of general data archivers, such as ZIP or RAR, but at the same time they are specially adapted to compress audio data …. Lossless encoding While it is ideal in terms of preserving the quality of audio materials, it cannot provide a high level of compression.

2. There is another more modern way to compact data. This so-called lossy data compression (Engl. “Lossy encoding”) The purpose of encoding is to achieve the highest data compression rate by all means while keeping sound quality at an acceptable level. The idea behind lossy encoding is based on two simple underlying considerations:

original digital audio data is redundant: it contains a lot of unnecessary information that is useless to the ear, which can be removed, thereby increasing the compression ratio;
Requirements for the sound quality of audio material may vary and depend on specific purposes and areas of use.
Lossy encoding is therefore called “lossy”, which results in the loss of some of the audio information. Such encoding leads to the fact that the decoded signal, when reproduced, sounds similar to the original, but in reality it is no longer identical to it. Most lossy coding methods rely on the use of the psychoacoustic properties of the human auditory system, as well as various tricks associated with resampling and resampling the signal. In frequency, during the compression process, the encoder analyzes the audio data to identify various details of the sound that can be ignored. Disguised frequencies, inaudible and inaudible sound details can be sacrificed for a higher compression ratio. Where intelligibility is only important in sound (for example, in telephony, where the presence of frequencies above 4 kHz is not necessary), the audio information during the encoding process undergoes a serious “simplification”, which, together with the use of successful “smart” quantifiers and “greedy” data compression algorithms.

Basics of digital audio

Basics of digital audio:

Before the computer can record, manipulate, and reproduce sound, sound must be transformed from an audible analog form to a computer-acceptable digital form, using a process called analog-to-digital conversion (ADC). Once the sound data has been stored as bytes in the computer, the power of the computer’s CPU can be used to transform this sound in thousands of ways. Finally, when you are ready to listen to the result, the digital-to-analog conversion (DAC) process transforms the sound bytes back into an analog electrical signal from the speakers.

Sampling: Analog to Digital Conversion

Given an analog signal, discrete values ​​of its amplitude are taken at small time intervals, obviously the more reliable the reproduction the more samples per second are taken. These obtained values ​​are assigned a digital value that the computer can understand and process as required. We can use 8 or 16 bit words, thus obtaining 256 or 65536 different combinations and obtaining higher resolution.

 

SAMPLE FREQUENCY: According to the Nyquist theorem, it is possible to accurately repeat a waveform if the sampling frequency is at least twice the frequency of the component with the highest frequency. The highest frequency that the human ear can perceive is close to 20 kHz, so the 44.1 kHz sampling rate of sound cards is more than enough. This value is the one used today by CD audio players.

SAMPLE SIZE: The sample size controls the dynamic range that can be recorded. For example, 8-bit samples limit the dynamic range to 256 steps (50 dB range). In contrast, a 16-bit sample has a dynamic range of 65,536 steps (90 dB range) a substantial improvement. The human ear perceives a whole world of differences between these two sample sizes. Ears are more sensitive to detecting differences in pitch than intensity, but are even more sensitive to the strength of sound.

From the previous processes we can get an audio file, such as (and since it is the best known), a WAV audio file. It is the own format of Windows. They can be 8 or 16 bit with sampling rates of 11,025 kHz, 22.05 kHz, or 44.1 kHz and generally have good sound quality.

Digital audio compression

It could be assumed that all you have to do to get good sound is to record at the 44.1 kHz speed limit with 16-bit (2-byte) samples. The only problem that appears if recording in stereo, sampling simultaneously on the left and right channels at 44.1 kHz, a one minute sound sample needs a 10.58MB storage space. This involves using large disk spaces to store these sound files. Many compressed file formats (codecs) have been developed that enable high-quality recording without the need for so much disk space.

Most common audio formats:

With the simple objective of listing a series of codecs used by different operating systems to perform audio compression. Later, a more complete description of the most used is made: MP3.

Therefore, some of the most used are:

Advanced Audio Coding (AAC): used by Apple computers. More efficient than MP3.

Audio for Unix (AU): Acoustic standard for the JAVA programming language.

Windows Media Audio (WMA)

Ogg Vorbis: It is free, open and not patented.

Atrac: compression and playback technology for minidisc.

 

The codec par excellence: the MP3

Its origin and current

The abbreviations MP3 respond to the abbreviation of MPEG (Moving Picture Expert Group) 1 Layer 3, which is a perceptual coding algorithm. This among others was developed by the Moving Picture Expert Group (MPEG) (http://www.cselt.it/mpeg/) together with the Fraunhofer Institute of Technology (http://www.ipa.fhg.de/english/ ).

Moving Picture Expert Group is an ISO / IEC research committee. MPEG is in charge of the international development of compression, decompression, processing and encoded rendering standards for movies, audio and the combination of both. It is a non-profit institution created in 1988, which brings together 300 experts from 20 countries three times a year.