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ENCODING PRINCIPLES OF THE MP3 FORMAT.

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ENCODING PRINCIPLES OF THE MP3 FORMAT.

Mp3 Encoding

Mp3, or fully MPEG-1, 2 and 2.5 Layer 3, is one of the most popular and widespread standards for storing audio data.

MP3 ENCODING

In this article, we will not delve into the history of creation and further development, but will consider the basic principles of the standard and examples of its implementation.

The mp3 standard does not establish a specific compression algorithm to “encode” the source data, but rather describes the essence of the possible methods.

The quality of the result obtained depends on the modification of the algorithm used, embedded in any encoding program of the “codec”, and on the quality of the original audio data.

There are 3 most common modifications of the mp3 format, which differ in the compression ratio parameters of the original audio data.

Name
Modification of the rule
Data rate per second (bit rate) Possible sample rates
MPEG-1 layer 3
32 – 320 kbps 32000 Hz
44100 Hz
48000 Hz
MPEG-2 Layer 3 16 – 160 kbps 16000 Hz
22050 Hz
24000 Hz
MPEG-2.5 Layer 3 8 – up to 160 kbps 8000 Hz
11025 Hz

Processing begins with dividing the original audio signal into equal time intervals: equal frames, for example 0.05 or 0.26 seconds, after which each frame is analyzed and compressed according to general or individual parameters based on the data of the previous and next frames.

Most of the compression algorithms used are based on the perceptual characteristics of the human ear. Let’s consider the main options, which, as a rule, are applied in a complex way.

It is worth starting with the fact that, by ear, the average person is capable of perceiving a frequency range of approximately 10 Hz to 20,000 Hz. With growth, changes occur in the hearing aid and, for most, the sensitivity the higher frequency range decreases, as a result of which, in some mp3 modifications, during compression, all frequencies above 16000 hertz are cut off, which can significantly reduce the amount of information.

Audio recordings can be encoded in stereo (a surround sound effect that uses separate channels for the left and right speakers) or mono (the opposite of stereo). In mp3 format, different tracks are not recorded for each of your speakers, but information about the differences between the left and right channels.

In acoustics, there is a concept like “harmonics”, these are the frequencies of the “sounds” that sound together with the main and most prominent tone. For example, when hitting a drum, the loudest sound will be the tone and the minor, weaker, will be the harmonics.

After such a loud sound, the so-called “period of deafness” occurs, during a period of duration in which a person’s hearing practically does not respond to changes.

If in the intervals of the “deafness period”, remove all frequencies, then the errors of perception, will practically not allow to notice their absence, because of this, during compression, the weakest harmonics are cut off, located close to the most sounds. strong: tones.

A method is used to replace the near peak values of the signal “peaks” (in terms of volume) with an average value.

There is a concept as bit rate: this is a value that characterizes the number of transmitted bits of information “units” during a period of time, usually one second.
The higher the bit rate, the better the audio detail will be, as long as the original, uncompressed audio data is of high quality.

As you can guess, digital formats consist of certain code sequences, in other words of sequences 0 and 1.
To save space, frequent joins within a file are assigned unique identifiers that replace long sequences.

Thanks to such complex influences, it is possible to compress the original audio signal into one of the popular formats with loss of quality – the mp3 format.

Various experiments have been carried out many times in order to reveal how significant the differences are before and after compression in mp3. As tests have shown, differences, some similar moments were not always possible, quickly and to distinguish, even when reproduced on equipment with higher fidelity.

For those who have never had the opportunity to directly compare the original and compressed audio recording, in most cases it will take some time or even find obvious differences.

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MP3 COMPRESSION

To achieve such a dramatic reduction in the number of bits required to transmit an MP audio signal, use different techniques. These techniques include those based on perceptual coding and others such as byte reservation, stereo assembly or Huffman codes. Percentage coding consists of removing all the information that goes into the audio signal that the human ear is not capable of detecting. We will now describe them:

PERCEPTUAL CODING

Minimum hearing threshold The ear’s minimum hearing threshold is the power below which a tone at a given frequency is not capable of being detected by the ear. This threshold is non-linear. As we see in the figure, which represents the Fletcher and Mundson law, the frequencies in which we hear best are those between 2 and 5 Khz. Therefore frequencies outside that band are not totally essential since they will hardly be perceived. Therefore it is possible to remove the content of the audio signal outside these frequencies.

As we can see in the drawing, the range in which a lower power is needed for the tone to be heard is between 2 and 4 Khz.

The masking effect This effect consists in that, when an audio signal has a tone at a given frequency, it produces a masking effect at the frequencies close to it, so that if at these nearby frequencies the signal does not exceed a certain power threshold cannot be heard and therefore it is not necessary to encode them. The form that this power threshold will take according to the position of the tone or the masking tones is what is called the psychoacoustic model, which as the name itself indicates is a perception model that tries to emulate the perception of the human ear.

In this graph we can see how if we put a tone at 1 Khz of 60 dB (masking tone) and then we put another tone at, for example 1.1 Khz and we vary the frequency of this, it is not possible to detect the presence of this second tone until its power exceeds the threshold presented in the figure.

In this case we see various masking tones and the resulting new hearing thresholds. In MP3, what is done is to divide the spectrum to be transmitted (that is, between 2 and 5 Khz) into frequency subbands, so that the power of the subband is evaluated and the masking threshold is created in the nearby subbands. Nearby subbands that exceed that power threshold are coded and those that do not exceed it are not coded.

Furthermore, the masking is not only in appearance but also in time as we can see in the figure.

The byte reserve: Often, some passages of a musical piece cannot be encoded at the same rate without altering the quality of the music. MP · then uses a small byte reservation that acts as a buffer using the capacity of passages that can be encoded at a lower rate in the given stream.
The stereo assembly In the case of a stereo signal, the MP3 format can use a few more tools to further compress the data.
Intensity stereo (IS) The human ear is not able to locate with complete certainty the spatial origin of sounds for very high or very low frequencies. This technique takes advantage of this, recording some frequencies as a monophonic signal, so that a minimum of spatial content is subtracted from the sound.
Mid / Side (M / S) Stereo When the left and right channels are similar then a middle channel (L + R) and a side channel (LR) are created, which are encoded instead of encoding the left channel on one side and the right for another. In this way it is possible to reduce the transmitted data using fewer bits for the lateral channel. Then during playback the MP3 decoder will reconstruct the left and right channels.

Huffman Coding: This coding technique is used at the end of the whole process. It works by creating variable-length codes, so that the symbols that appear in the bitstream most likely have shorter codes. The translation between symbols and codes is done using a table. Each code has a unique prefix so that the codes can be decoded correctly despite their variable length. This type of coding allows on average to reduce by 20% the amount of data to be transmitted. It is an ideal complement to perceptual coding since, during great polyphonies, perceptual coding is very efficient since many sounds are masked, but nevertheless little information is identical and Huffman’s algorithm becomes inefficient. During pure sounds there are few masking effects, but Huffman encoding is very efficient since digitized sound contains many repeating bytes.