Tag: mp3 encoding quality

Mp3 (an audio encoding method) Part 2


Free Download Mp4Gain
picture

Mp3 (an audio encoding method) Part 2

mp3 3ncoding

MPEG-1 Audio Layer 2 encoding began as a digital audio broadcast (DAB) managed by Egon Meier-Engelen at the German Deutsche Forschungs- und Versuchsanstalt für Luft- und Raumfahrt (later known as Deutsches Zentrum für Luft- und Raumfahrt, German Space Center). )draft.

mp3 encoding

This project is funded by the European Union as a EUREKA research project, and its name is commonly known as EU-147. The study period for EU-147 was from 1987 to 1994.
2. By 1991, two proposals had emerged: Musicam (called Layer 2) and ASPEC (Adaptive Spectrum Sensing Entropy Coding). The Musicam method proposed by Philips of the Netherlands, CCETT of France, and the Institut für Rundfunktechnik of Germany was chosen due to its simplicity, error robustness, and lower computational effort in high-quality compression. The Musicam format based on subband coding is a key factor in determining the MPEG audio compression format (sample rate, frame structure, header, sample points per frame). This technology and its design philosophy are fully integrated into the definition of ISO MPEG Audio Layer I, II and later Layer III (MP3) formats. The standard was developed by Leon van de Kerkhof (Layer I) and Gerhard Stoll (Layer II) under the auspices of Prof. Mussmann (University of Hannover).
3. A working group consisting of Leon Van de Kerkhof from the Netherlands, Gerhard Stoll from Germany, Yves-François Dehery from France and Karlheinz Brandenburg from Germany absorbed design ideas from Musicam and ASPEC and added their own design ideas to develop an MP3. MP3 can achieve MP2 sound quality from 192 kbit/s to 128 kbit/s.
4. All of these algorithms eventually became part of the first group of MPEG standards, MPEG-1, in 1992, resulting in the international standard ISO/IEC 11172-3 published in 1993. Further work on MPEG audio was eventually became part of the MPEG-2 standard, a second group of MPEG standards developed in 1994, officially known as ISO/IEC 13818-3, first published in 1995.
5. The compression efficiency of the encoder is generally defined by the bit rate, because the compression rate depends on the number of bits (: in: bit depth) and the sampling rate of the input signal. However, there are often products that use CD parameters (44.1 kHz, two channels, 16 bits per channel, or 2×16 bits) as the compression ratio reference, and the compression ratio using this reference is usually higher, which which also shows that the compression ratio is very important for lossy compression problems.
6. Karlheinz Brandenburg used Suzanne Vega’s song Tom’s Diner on CD to test MP3 compression algorithms. This song is used because the song’s smooth and simple melody makes it easier to hear glitches in the compressed format during playback. Some jokingly refer to Suzanne Vega as “the mother of MP3”. Some more serious and critical audio extracts (glockenspiel, triangle, accordion…) from the EBU V3/SQAM reference CD are used by professional audio engineers to assess the subjective perceived quality of the MPEG audio format.


Free Download Mp4Gain
picture


Mp4Gain Main Window
picture


Mp4Gain Features
picture


Free Download Mp4Gain
picture

Mp3 (an audio encoding method)

Mp3 (an audio encoding method)

Mp3 encxoding

MP3 is an audio compression technology, its full name is Moving Picture Experts Group Audio Layer III, called MP3.

mp3 encoding

It is designed to drastically reduce the amount of audio data. Using MPEG Audio Layer 3 technology, music is compressed into a smaller capacity file with a compression ratio of 1:10 or even 1:12, and for most users, playback quality is not as good as the original uncompressed. audio Significant decrease. It was invented and standardized in 1991 by a group of engineers at the Fraunhofer-Gesellschaft research organization in Erlangen, Germany. Music stored in the form of MP3 is called MP3 music, and a machine that can play MP3 music is called an MP3 player.

Motion Picture Expert Compression Standard Audio Layer 3 foreign name Moving Picture Expert Group Audio Layer III research organization Fraunhofer-Gesellschaft type audio coding advantage Drastically reduce the amount of audio data defect sound quality loss
content
1 Features
2 story
▪ origin
▪ go to the masses
3 audio quality
4 patent issues
transmission characteristics
MP3 converts the time-domain waveform signal to a frequency-domain signal by taking advantage of the human ear’s insensitivity to high-frequency sound signals and splits it into multiple frequency bands, using different compression rates. for different frequency bands and increasing the compression ratio for high frequencies (even ignoring the signal) Use a small compression ratio for low frequency signals to ensure that the signal is not distorted. In this way, it is equivalent to discarding the high-frequency sound that is basically inaudible to the human ear [1], keeping only the audible low-frequency part, thus compressing the sound with a compression ratio of 1:10 or even 1: 12. Because the full name of this compression method is called MPEG Audio Player3, people call it MP3 for short.
According to the MPEG specification, AAC (Advanced Audio Coding) in MPEG-4 will be the next generation of the MP3 format.
Compared to CD, FLAC and APE lossless compression formats, the sound quality of the highest parameter MP3 (320 Kbps) is not much different.
MP3 players are dying
When they first came out, MP3 players were at the forefront of the digital revolution. However, sales of iPods and other MP3 players in the UK fell sharply in 2012 as consumers turned to other digital products such as smartphones.
In 2012, sales of MP3 players in the UK market were £110m ($178m), just 29% of the £381m in 2011, according to market research firm Mintel. Mintel expects total MP3 player sales in the UK market to halve by 2017. In the worst case scenario, total MP3 player sales in the UK market will be just 25 million dollars five years later. [23]
1. MP3 is a data compression format;
2. Discards pulse code modulation (PCM) audio data that is not important to the human ear (similar to JPEG is a lossy image compression), resulting in a much smaller file size;
3. MP3 audio can be compressed according to different bit rates, providing a variety of trade-offs between data size and sound quality. The MP3 format uses a mixed conversion mechanism to convert audio domain signals. time in frequency domain signals;
4. 32 band polyphase integral filter (PQF);
Modified discrete cosine filter (MDCT) of 5, 36 or 12 taps; each subband size can be independently selected between 0…1 and 2…31;
6. MP3 not only has extensive client software support, but also has a lot of hardware support, such as portable media players (referring to MP3 players), DVD and CD players, outgoing calls

How to distinguish the sound quality of Mp3 songs?

How to distinguish the sound quality of Mp3 songs?

Mp3 quality

Factors that affect audio quality are the number of channels, the sampling rate, and the number of quantization bits.

Mp3 Quality

It’s not directly related to file size, I think friends who have used Audition or play more music will be more familiar with it.

-Number of channels

Channel count is easy to understand and is often referred to as channel count. Usually we talk about left and right channels, single and double channels, which refers to the number of channels.

The music that we listen to often in life is basically two-channel, that is, the left and right channels. Generally speaking, the higher the number of channels, the better the audio quality. Then the stereoscopic feeling of the sound will be stronger. It will feel more real. When a person speaks or an object makes a sound, the sound also spreads in all directions, and of course there are more than two channels. So, in fact, it is difficult for digital audio to achieve real sound realism.

-Sampling frequency

For example, when Audition exports audio files, there is a sample rate option. What exactly is this sample rate?

Sampling rate is explained in official words: the number of samples per unit of time (within 1S). The higher the sample rate, the more data it collects and the better the sound quality.

But you will find that music in real life is generally 44100HZ sampling rate, like the lossless music in the picture above. So there are so many miscellaneous sample rate options in the image below. What does this mean? The reason is that the audible sound range of the human ear is between 20 and 20,000 Hz. Even if you increase the sample rate, it will still sound the same to ordinary people, so there is no need.

-Quantization bits

This is also very understandable. It’s like the number of bits that people often say about the computer. Audio also has the concept of bits. A common number of bits for audio is 16 bits. Generally speaking, the higher the number of bits, the better the sound quality. The popular understanding of quantization is to digitize the sampled value, that is, in the binary form recognized by the computer.

The property display in Windows may not display these parameters intuitively, but you can see them with the help of tools. Sound quality is determined by the above three aspects. Instead of looking at the size of the file. Of course, the audio is basically compressed and transcoded when it is broadcast to the audience. After all, high volume digital audio is not conducive to broadcasting.

 

FAQ

How to distinguish the sound quality of an mp3?

It is important to look at several elements to distinguish its sound quality. Of course, first is the quality of the recording, then the bitrate and samplerate.

Your can improve the sound quality of an mp3?

It is possible, using Mp4Gain, to improve the perception of the quality of an mp3 or any other audio or video format. In addition to modifying the bitrate and sample rate, we can modify the “color” with an equalizer and even slightly modify the pitch and of course normalize the audio.

Encoding an mp3

Encoding an mp3

encoding mp3

What is masking

mp3 encoding

The lossy MP3 audio compression algorithm uses a limitation of human hearing perception called auditory masking. In 1894, the American physicist Alfred M. Mayer reported that a tone could be made inaudible by another tone of a lower frequency. In 1959, Richard Amer described a complete set of auditory curves related to this phenomenon. Between 1967 and 1974, Eberhard Zwicker worked on tuning and masking critical frequency bands, which in turn built on the fundamental research of Harvey Fletcher and his collaborators at Bell Labs in this area. Perceptual coding was first used to compress speech coding with Linear Prediction Coding (LPC), which has its origins in the works Fuminada Itakura (Nagoya University) and Shuji Saito (from Nippon Telegraph and Telephone) in 1966. In 1978, Bishnu S. Atal and Manfred R. Schroeder of Bell Labs proposed an LPC speech codec called adaptive predictive coding. , which used a psychoacoustic coding algorithm using the masking properties of the human ear. Schroeder and Atal’s further optimization with J.L. Hall was later described in a 1979 article. In the same year M.A. Krasner proposed a psychoacoustic masking codec, which published and produced hardware for speech (not used to compress musical bits), but the publication of its results in a relatively obscure technical report from the Lincoln Laboratory did not immediately influence the mainstream of the development of psychoacoustic codecs. The Discrete Cosine Transform (DCT), a type of transform coding for lossy compression, proposed by Nasir Ahmed in 1972, was developed by Ahmed with T. Natarajan and KR Rao in 1973; published their results in 1974. This led to the development of the Modified Discrete Cosine Transform (MDCT) proposed by JP Princen, AW Johnson, and AB Bradley in 1987 after earlier work by Princen and Bradley in 1986. MDCT later became the main body of the MP3 algorithm. Ernst Terhardt et al. Built an algorithm that describes auditory masking with high precision in 1982. This work adds to many reports by authors dating back to Fletcher, as well as work that originally defined critical ratios and critical bandwidth. In 1985, Atal and Schroeder introduced Code Excited Linear Prediction (CELP), an LPC-based perceptual speech coding auditory masking algorithm that achieved a significant degree of data compression for its time. IEEE peer-reviewed journal “Favorite Communications” reported on a wide variety of audio compression algorithms (mainly perceptual) in 1988. The February 1988 issue of Voice Coding for Communication reported on a wide range of audio compression algorithms bit-based established and operational. technologies, some of which use auditory masking as part of their core design, and some of which show real-time hardware implementations. – https://ru.qaz.wiki/wiki/MP3

ENCODING PRINCIPLES OF THE MP3 FORMAT.

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.

MP3 ENCODING

MP3 ENCODING

Mp3 encoding

The first step in encoding by the user is to specify a bit rate. This indicates the quality and at the same time the storage requirement of an MP3 file.

MP3 encoding

COMPRESSION RATES

With most recording programs, the quality of an MP3 file can be freely selected before recording begins. According to the Fraunhofer Institute, the CD quality of an MP3 file is a bit rate of 112 to 128 kbit per second, other measurements put CD quality at up to 160 kbit per second. However, the most used and sufficient for most listeners is 128 kbit.

In comparison, a corresponding CD quality for Layer 1 is 384 kbit / s and 256 kbit / s for Layer 2. A wave file works with a 1.4 Mbit / s bit rate and therefore works with roughly the same space requirements. as a CD audio track (CDA).

74 or 80 minutes of music can be put on a CD (depending on the size of the sound carrier), in MP3 format with a bit rate of 128 kbit / s, 11.5 or 12.4 hours would be possible.

PSYCHOACOUSTICS

MP3 audio compression relies on filtering out unnecessary information. Psychoacoustics is a science that deals with the perception of sound by the human ear.

Eg: You are in a disco. Loud music blasts through huge speakers and you try to talk to each other. This is almost impossible unless you yell. In acoustics, this is called masking. To eliminate masking, the sound level of speech should be raised to such an extent that the interfering signal (in this case music) no longer covers it.

Processes like this belong to the fundamental areas of psychoacoustics.

Tones below this threshold are not heard and therefore become noise during MP3 recording (skipped).

The overlays work as follows: you have, for example (picture 2) a tone with 1 kHz (1) and another tone with 1.1 kHz, which is approximately 18 dB lower (2). The second shade is completely superimposed on the first. This also works for other weaker tones (see Fig. 2). Another tone with a frequency of 2 kHz, which is also 18 dB quieter than the first, would not overlap because it is just outside the threshold of the first tone.

Noise can be another compression option for MP3 recording. The fact that when a sound is digitized it cannot be sampled at an infinite frequency, a noise imperceptible to the human ear (quantization noise) is generated. It is used as a model for the MPEG audio layer and thus increases the noise around a tone. Above all, loud and short tones mask a certain range in the frequency range before and after themselves where the weakest signals would not be audible. With MP3 encoding, the noise level increases in this area, as if digitized at a lower resolution.

There is also masking in the temporal area: hearing needs a so-called “recovery time” for loud and quiet noises until it is fully functional again. This is especially noticeable with strong, short, and rapidly rising tones. After a delay of about 5 ms, the hearing threshold drops again and after about 200 ms it reaches the normal level, the so-called resting hearing threshold. This effect is called post-masking. The effect of pre-masking is less important, but even more impressive: it is based on the fact that the brain processes loud sounds more quickly than soft ones. To some extent, the strong impulse outweighs the silent one on the way to the brain. This results in a pre-masking time of up to 20 ms.

The above psychoacoustic algorithm is used in the following steps:
– Audio information is divided into subbands
– Subbands are reduced
– 16-bit samples are generated
– Samples are compressed
– Compressed samples are combined into blocks
– Coding according to Huffmann Procedure
: summary in tables

DIVIDED INTO SUBBANDS

Depending on the frequency of the acoustic information, it is divided into 32 subbands. The bands are of different sizes due to adaptation to the human ear according to a psychoacoustic model.

The division is done with the help of a polyphase filter. This means that the samples are decimated and filtered simultaneously.

In layers 1 and 2, the bands were the same size with a bandwidth of 625 Hz each. The reason for this division is to provide the algorithm with a better target.

SUBBAND ​​REDUCTION

The MP3 encoder now examines each of the subbands according to the psychoacoustic model for expendable frequencies. Here, the masking threshold is determined, then the subbands whose level is below this masking function are removed. Another reason for dropping an entire sub-band could be that it is inaudible due to the pitch, similar to a dog’s whistle.

CONVERSION INTO 16-BIT SAMPLES

The frequency bands are sampled and converted to 16-bit samples. Tones are broken down into digital signals and further processed as numerical values. The sample rate determines the length of the sample intervals. However, neither the measurement of the amplitude nor the size of the sampling intervals can be infinitely precise. For this reason, with analog-digital conversion, a value is rounded between two sample points. This results in rounding errors that are noted in what is known as quantization noise. This can be kept inaudible using the highest possible resolution: with 8-bit, a maximum of 256 levels can be displayed, with 12-bit and 4096 and with 16-bit 65536 individual steps, so that noise is not heard.

However, some samples are also digitized with a lower sample rate. In the eighth subband, for example, there is a tone with 1 kHz and 60 dB. The MPEG audio encoder now calculates the masking threshold and recognizes that it is 36dB lower. The acceptable signal-to-noise ratio here is 24 dB, which corresponds to a 4-bit resolution, since the two values ​​are directly related. Leaving one bit out of resolution increases the noise level by 6dB. Since an audio CD is generally digitized with 16 bits, considerable data reduction can be applied here.

SAMPLE COMPRESSION

The next step is to compress the samples further. However, this process no longer has anything to do with the original shades. From here on, compression is only data-driven.

Each sample consists of 16 bits, but not all of them are absolutely necessary to represent a level. For example, leading zeros can be omitted. If, for example, the value 0000011101010101 is obtained for a sample, the algorithm truncates the result to 11101010101. To reconstruct the original 16 bits from this information, the decoder needs two pieces of information: the scale factor and the bit allocation. The scale factor indicates where the remaining bits of the sample were in their original state. The bit mapping contains the information about how many bits are left in the sample, since you can no longer calculate with a fixed 16-bit number. However, if you were to store these values ​​individually for each sample, you wouldn’t gain much,

GROUPING THE SAMPLES

The 16-bit samples that were just created are now combined into blocks. There are two different block lengths for this purpose: the short blocks with twelve samples and the long blocks with 36 samples.

Long blocks are used for low frequencies. However, long blocks would not allow sufficient resolution at higher frequencies; short blocks are used here. In the so-called mixed block mode, long blocks are used for the two frequency bands with the lowest frequencies. For the remaining 30 frequency bands, it is the turn of the short blocks. This mode allows better frequency resolution in the low frequencies without paying tribute to the sampling frequency in the high frequencies.

HUFFMANN CODING

The last step in MP3 compression is Huffmann encoding. This algorithm is also used, for example, in packaging programs such as WinZip. The frequency of certain values ​​is important here. However, the subbands are organized in advance. Subbands with lower frequencies tend to contain significantly more values ​​than those with high frequencies. The subbands are divided into three groups according to their frequency. Each area has its own Huffmann tree (Fig. 3) to achieve the optimal compression factor.

As a first step, the encoder excludes high frequencies; encoding is not necessary here, as its size can be derived from those of the other two regions. The mid-frequency range is treated as is, and the low frequencies are again divided into three regions, each of which is assigned its own Huffmann tree. The appearance of a Huffmann tree is stored in the MP3 file.

The structure of a Huffmann tree works as follows: frequently occurring values ​​are given a short sequence of bits, while rare values ​​are given a long one, so the algorithm first determines the distribution of values ​​within the data to be compressed.

To determine what is known as the Huffman tree, you start with the two rarest values. They are assigned a “0” or a “1”. The two values ​​are summarized, in the order that they are now represented by the sum of their frequency. The same is true for the next two rarer values. This process ends when only one value remains. The result of this procedure is a tree structure. The encoding is based on this structure. Each branch on the left receives a 0, each branch on the right is identified by a “1”. In our little example, the least common would be

Value 4 represented by the sequence of bits 010. The most common value 6, on the other hand, is assigned a simple 1.

FRAMEWORK SUMMARY

The result of the above compression is summarized in so-called frames. Each of these frames contains 1152 samples (32 subbands x 36 samples). A frame consists of a header, a checksum check, the actual audio data, and in certain circumstances a so-called bit repository. Such a deposit arises when the samples within the frame can be compressed in such a way that the full theoretical number of bits in a frame is not required. The encoder can fall back on these buckets if the available bits are insufficient for a subsequent frame. A distinction must be made between two terms: frame size and frame length.

The size of the frame is determined by the number of samples and is constant within a layer. In Layer 1 format, this is always 384 samples per frame, in Layers 2 and 3 1152 per frame. However, the length of the frame may differ at Layer 3 due to the change in bit rate or the pool of unfilled bits. The frame also contains the aforementioned information about the scale factor and bit allocation to be able to reconstruct all the samples again.

A file header, as it is known from other file formats, does not exist in an MP3 file. In the case of an image file, a header would contain information about the entire image (e.g. size, color depth, resolution

MP3: quality standard?

For many, dematerialized music rhymes with illegally downloaded MP3.

If this comment is often true, since illegal music sharing platforms have made mp3 the primary format for music playback, you obviously need not limit yourself to the single mp3 format.

Mp3

MP3: birth of a format

The MP3 was democratized on the music exchange platforms of the time like Napster, Kazaa, Emule … in the late 90’s and for good reason they allowed you to download an entire album in a few minutes by compressing the music and thus shrinking the files.

Therefore, it is the need to exchange files and shorten the download time (remember we were paying the internet at that time according to your connection time …) making the development of mp3 essential and for many synonymous with dematerialized music.

The MP3 principle is therefore simple and attractive on paper: enable file sharing by drastically reducing the weight of files (more than 90%), and only by keeping what the human ear can do. listen, that is, the frequencies between 20 Hz and 20 kHz.

MP3: bad reputation

Unfortunately, the main consequence of this thin race is that the quality deteriorates: every mp3 has a compression level. The higher it is, the more the musical signal is cut off: this is called destructive compression: we eliminate all information that is considered useless and impossible to return.

Mp3

Therefore, during this period there was the spread of the famous 128 kbps MP3: this figure indicates the amount of information in the file and therefore its quality, the higher and better the sound will be. Therefore, some sort of standard has gradually been established around this bit rate of 128 kbps (kilobits per second), since it is inseparable as the quality of the CD at 1411 kbps.

mp3 – Napster

The Napster interface, one of the first illegal music sharing programs. Note the music ratio at 128 kbps (see less …)

It is clear that we do not reduce the amount of musical information by 90% with impunity and the results are often poor, the quality of the mp3 with 128 kbps is much lower and perfectly noticeable from the original CD. Then the new bit rates of 160 kbps, 192 kbps, 256 kbps and 320 kbps came to maximum, then “VBR” formats for “Variable Bit Rate” and against CBR (constant bit rate)) used earlier: we decreased the bit rate at rest and increased if necessary.

Therefore, we can see that it is difficult to pronounce the MP3 format in general: the results will be very different between a 128 kbps CBR mp3 and a 320 kbps VBR and, to a large extent, for the latter, for the price of one double weight.

Alternatives to MP3

MP3 is not the only dematerialized music format, it is first necessary to divide the compression formats into two categories:

Destructive Compression Formats: We remove content to reduce its size

MP3: the readable standard for 100% of music devices released in 20 years

AAC: Used by Apple in the iTunes music store (Apple Audio Codec), almost as universal as MP3

Ogg Vorbis: a free, efficient yet non-standard format

WMA: Microsoft format, not very standard either, except on Windows PC (can Microsoft have anything to do with it?)

Non-destructive compression formats: we compress the data for storage and decompress it when reading, therefore the sound reproduction is lossless, but it generates files 3 times larger:

FLAC: Free Lossless Audio Codec, it is somewhat the equivalent of Ogg in that it is royalty free and has established itself as the current standard for non-destructive formats (lossless in English)

ALAC: The Apple version of FLAC, which has also been in free form for some time, has the advantage of being compatible with the brand’s products and computers and offers the same benefits as FLAC.

Mp3: Advantages of MP3

MP3 refers to an encoding format that is formally referred to as MPEG-1 Audio Layer III for digital audio. Designated MP3 or MP3 data that stores basic MPEG-2 audio data or MPEG-1 audio data that is encoded. They do not contain any completely different complexity than the format. Find out more about the MP3 file format and its benefits.

As worrying as audio compression is, MP3 is a lossy compression module for encoding data with the expectation of partial rejection and inaccurate approximations of the data. And this ends with a noticeable reduction in file size, now no longer like with uncompressed audio.

The small size and excellent audio quality led to the provision of song data on Highway Records in the 1990s. MP3 served as expanded storage capacity and bandwidth in these hardly expensive times.

In about a year, the MP3 format has addressed controversy surrounding song piracy and copyright infringement. Instead, this file format became a custom format with the advent of avid portable gamers, including smartphones.

How does this compression work? This compression reduces the precision of the explicit parts of the sound that people can no longer hear. This method is ceaselessly referred to as perceptual coding or psychoacoustic modeling.

Then the free time of audio knowledge is recorded with the extraction of the allocated memory. FFT and MDCT algorithms are used here. Unlike CD audio, this audio compression design can reduce file size by up to 95%. Every time you document a conversion at a fixed bit rate of 128 kb / s, the file size is 9% more realistic than the audio on the actual CD.

Advantages of MP3

Here’s a hint about one of the benefits the MP3 audio file format offers. For these benefits, you can also lift whenever you can safely move around in this format, or opt for a lossless option.

One of the main advantages of this file format is that you can document songs, speeches or conversations for hours without affecting the allocated storage space. Basically, up to 95% of the allocated storage space is allocated. And the most important thing is that fine audio is identical with completely different codecs that take a lot of assignment.

The small size of such audio data allows you to map hundreds of data onto a small memory card or memory card. In other words, you can also save more than 170 songs on a CD with a storage capacity of 700 MB. On the other hand, the CDDA option does not mean that more than 15 tracks can be saved.

Instead, due to the logic that the file size is extremely small, you don’t have to spend a lot of bandwidth every time you get a lot of songs. Therefore, MP3 is a great wish for all types of users.

The best thing about MP3 is that you can change the audio level you want based on the available memory allocation. It can also rise between 32 kbps and 320 kbps. On the other hand, indicate that the higher the bit price, the larger the file size.

MP3 COMPRESSION

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.