MP3: features and alternatives


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The peculiarities of the MP3 format and some clues about other solutions of equal or even higher quality.

Impossible to deny, the MP3 format is the most common and most enjoyable to listen to music on the go or, as it has been for some years, streaming. We use it everywhere now and any device can play it today.

MP3 is part of the family of audio files called “lossy”, that is, the types of formats that can also reduce the amount of data that should contain a sound, in any case try to maintain at least an acceptable quality.

The peculiarities of the MP3 format and some clues about other solutions of equal or even higher quality.
The parameters that determine the quality level of an MP3 file are: the sampling rate, bit rate, encoder and of course the source. Now let’s move on to the order.
At the origin of everything is the source, that is, the support or source from which the MP3 file can be downloaded. The higher the quality of the source, the greater the end result: purchasing MP3s from particularly reliable sites or extracting them from compact discs in good condition is the basis for a successful MP3. What becomes crucial is the encoder (the most famous and free is LAME) or the software that takes care of creating the file after properly configuring its parameters.

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The sampling rate is measured in Herz and expresses the number of times per second. Second, as the analog signal is measured and digitized; for MP3 it must be as faithful as possible on a CD, ie 44 100 Hz (44.1 KHz).

Bitrate is the number of binary units flowing, measured every second. The value of the bit rate is not fixed: as it increases, the similarity to the original file will also increase proportionally. The higher the bit rate, the higher the quality, the larger the file size. The bit rate range ranges from 32 kbps to 320 kbps, the maximum that can be obtained from an MP3 file.

The ones we’ve just listed are an important part of the tricks that allow us to have an MP3 quality; however, be aware that a lost file is by no means faithful in all respects to the original source. The most famous lost alternatives are: AAC (the format Apple uses to sell music in the iTunes Store and since July to stream audio from the Apple Music service); WMA; MPC; OGG (excellent quality open source format).

If you are looking for maximum faith in digital audio, give up MP3 and its loss-free alternatives to switch to “loss-free” audio formats, ie loss-free quality. Overall, this file type compresses the original sound while keeping the number of bits intact. Needless to say, quality comes at a cost in terms of the space taken up: a lossless file takes about half of the original audio file, but “weighs” nearly three times as much as a 320Kbps MP3. Of these, the most famous and used are: FLAC; ALAC (Apple Lossless Format); BEE; WavPack. The “lossy” and “lossless” file distinctions are extremely applicable to images and videos as well, not just audio files.

On several occasions it has been said how absolutely difficult it is to distinguish an MP3 at 320 kbps, obtained under the best conditions, from its original version on CD or in lossless files; It is only possible to notice it with instruments at a certain level and with a good ear. When noted, the MP3 format is excellent for listening on the move, as highlighted above; On the other hand, to better preserve our music or listen to it on systems of a certain level, it is better to resort to lossless formats such as FLAC or ALAC.


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How is an mp3 analyzed inside?

How is an mp3 analyzed inside?

MP3 is the acronym for MPEG 1 Layer 3 and is a lossy digital audio format developed by MPEG (Moving Picture Experts Group) in conjunction with the Franunhofer Institute of Technology to include it as an audio format for the MPEG-video format. 1. It is currently an ISO (International Organization for Standardization) standard. The reason it has become so popular is that it allows for high sound quality in very little storage space: About 650 songs can be recorded on a 650MB CDROM, in instead of the 15 that we could store following the format of traditional CD-Audio. Furthermore, it is possible to adjust the quality of the output file by adjusting the bitrate (sampling rate and number of bits per sample), which will be proportional to the size of the output file. Thanks to its small size, high quality and versatility, it became a standard for streaming.

It was said at the beginning that MP3 is a lossy algorithm, this means that the original and encoded sound are not exactly the same. For this, the MP3 takes advantage of the “deficiencies” of the human ear, specifically 3 of them:

Limits of hearing in frequency: The human ear is only capable of hearing frequencies that are approximately between 20 and 20,000 KHz, with which the rest are filtered and discarded as they would not add relevant information to the encoded signal. Also, the closer you are to the 2-4 Khz range (and harder to hear as the frequency gets closer to the extremes of hearing), the more audible it will be.
Masking effect: When 2 signals of similar frequency overlap, human hatred is only able to hear the one with the highest power (volume), therefore, the rest can be eliminated without appreciable loss of quality.
Stereo redundancy: Sometimes there is redundancy between the 2 channels and, furthermore, below a certain frequency, the human ear is not able to distinguish the directionality of the sound with which a single channel can be encoded and add to the other certain complementary information to not lose the spatial sensation of the other channel.
To carry out the three previous proposals, a system based on subbands is used in which the signal is filtered using several filters in order to have the signal separated into sub-signals, each covering a frequency range. Each of these bands is compared to a psychoacoustic model that determines which bands are important and which can be removed.

Specifically, a hybrid polyphase / MDCT (Modified Discrete Cosine Transform) filter bank is used: A filter bank is a set of band-pass filters that aim to separate the original signal into several frequency bands; A multiphase / MDCT hybrid filter bank is nothing more than a normal filter bank together with a block capable of doing the discrete cosine transform (MDCT).

The choice of which bands are maintained and which are removed is made by calculating the masking threshold, that is, it analyzes each audio sub-signal and calculates the amount of noise that can be input (signal is replaced by noise to save storage space) in function of the frequency, taking into account that a frequency masks signals of a higher frequency than yours rather than lower, without being noticeable to the human ear.

The following figure outlines the process described above:

The following figure represents the structure of an mp3 file:

As can be seen, an Mp3 file is made up of different frames which in turn are made up of an Mp3 header and MP3 data. Each of the frames is independent, that is, a person can cut the frames of an MP3 file and then play them back. The graph shows that the header consists of a sync word that is used to indicate the beginning of a valid frame. Following are a series of bits that indicate that the analyzed file is a standard MPEG file and whether or not it uses layer 3.

MP3 undoubtedly owes its success to Internet music downloads and portable audio players capable of playing the format. First, Discman compatible with MP3 were born, which allowed transporting 175 songs per cd instead of the usual 6. Subsequently, MP3 players based on a (small back then) flash memory were born. These had the advantage of being much smaller and lighter than portable CD players, but with the initial disadvantage that flash memory was small and expensive. Initially these devices had 64 or 128 MB memory, which allowed them to store between 16 and 32 songs. Currently these devices are sold with a memory of 1,2,4 or even 8GB. This allows them to store between 256 (for the 1Gb model) and 2048 (for the 8GB model)