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The compression algorithm of an Mp3.

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The compression algorithm of an Mp3.

In addition to the physiological structural properties of the human ear, the function of the brain also plays a very important role.

Mp3 compression algorithm

The pitch in the sound is determined by the fundamental tone, while the timbre is determined by the harmonics, and the human brain will automatically complete the fundamental tone, even if the fundamental tone does not exist. For example, the bandwidth of a telephone is only 300~3200 Hz, but when we listen to a man with a base tone of 120 Hz talking on the telephone, we can still hear his correct tone and will not confuse a boy with a girl. . .

We still don’t know how the brain uses complex calculations to reconstruct this non-existent tone.

PS Add a little visual easter egg, can you see what’s weird about this image?

(Please read the answer to the end)

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_ Your vision~ amazing! The human body still has too many unknown magic eggs waiting to be excavated~~

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The compression algorithm of an Mp3.

The birth of the MP3 compression algorithm is nonsense of human organs in the digital age. The whole algorithm is not improved around the math, but rather optimized around how to fool the human hearing organ.

Mp3 compression algorithm

So this algorithm is very curious, Baidu finally found information after a long time, and has a little understanding of the principle of it, so please record it.

basic principle
There is a special effect of shading effect on the human hearing model.
The role of the cochlea is as a spectrum analyzer, converting sound waves into signals of different frequencies. The villous cells at each specific location will be stimulated by a specific frequency, but when the basilar membrane leads to fluctuations, the villous cells around it will also be stimulated. That is, if there is a frequency with a high volume, and at the same time there is a relatively weak frequency near it, the sound of the relatively weak frequency will be covered by the relatively loud sound, and our human ears have no way to distinguish the sound There is another sound of a weaker frequency.

To the human ear, the perception characteristics of sound do not change on a linear frequency scale (human hearing is not that good), but can be expressed in a series of limited frequency bands called critical frequency bands. Simply put, the entire frequency band is divided into several segments, and in each frequency band the auditory perception of the human ear is the same, that is, the psychoacoustic characteristics are the same.
Then, according to this principle, the mp3 compression work can be simply divided into two parts:

The first step: dividing the original audio data into several subcritical frequency bands according to certain principles;

Step 2: Analyze the frequency spectrum according to the psychoacoustic model to find the masking effect curve. Then, according to this curve, each sub-frequency band is quantized separately, and finally the compression of the audio is below the masking effect curve.

In this way, mp3 compression is done. And it is surprising that mp3 is really compressed in the digital world, but it belongs to compression without distortion for human perception.

MP3 file format

Mp3 file format

Introduction:
MP3 file format

1. Overview:
MP3 files are made up of frames, and frames are the smallest unit of MP3 files. The full name of MP3 must be MPEG1 Layer 3 audio files. MPEG
(Motion Picture Experts Group) translates into Chinese as Moving Picture Experts Group, and refers specifically to moving video and audio compression standards.
MPEG1 standard, also known as MPEG audio layer, which is divided into three layers based on compression quality and encoding complexity, namely,
Layer-1, Layer2 and Layer3, which correspond to the three sound files of MP1, MP2 and MP3 respectively, and use different
levels of audio files according to different purposes. The higher the MPEG audio encoding level, the more complex the encoder and the higher the compression ratio. The compression ratios of MP1 and MP2 are 4:1 and
6:1-8:1 respectively, while the compression ratio of MP3 is as high as 10:1-8:1. 12:1, meaning one minute of CD-quality music requires 10MB
of storage space without compression, but only about 1 MB after MP3 compression encoding. However, MP3 uses a lossy compression method for audio signals. To reduce
sound distortion, MP3 adopts “sensory coding technology”, that is, it first analyzes the frequency spectrum of audio files during encoding, and then uses filters to filter the
noise . levels. Then the remaining bits are spread and arranged by means of quantization, and finally an MP3 file with a higher compression ratio is formed, and the
compressed file can achieve a sound effect closer to the original sound source during playback.
2. The whole structure of
MP3 files: MP3 files are roughly divided into three parts: TAG_V2 (ID3V2), Frame, TAG_V1 (ID3V1)
ID3V2 contains information like author, composer, album, etc. The length is not fixed, which expands the information volume of ID3V1.
A series of frames, the number is determined by the size of the file and the length of the frame. The length of each frame of the
frame
may not be fixed or fixed, and is determined by the bitrate
.
Each table is divided into two parts: table header and data entity Header of data.
frame
Record the bit rate, sample rate, version and other information of mp3, and each frame is independent of each other The frame
ID3V1 contains information like author, composer, album, etc., and the length is 128BYTE . 3. MP3 FRAME format: each FRAME has a FRAMEHEADER frame header, the length is 4BYTE (32 bits), there may be two CRC check bytes after the frame header, the existence of these two bytes depends on the FRAMEHEADER information If bit 16 is 0, there is no checksum after the frame header, and if it is 1, there is a checksum. The checksum length is 2 bytes, followed by the FRAMEHEADER, followed by the frame entity data. The format is as follows: FRAMEHEADER CRC (free) MAIN_DATA 4 BYTE 0 OR 2 BYTE The length is calculated from frame header 1. The format of the FRAMEHEADER frame header is as follows: AAAAAAAA AAABCCD EEEEFFGH IIJJKLMM

MP3 digital audio format

MP3 digital audio format

MP3 File Format

High-quality digitized audio requires a large amount of disk space.

mp3 file

Attempts to reduce the size of files using standard archivers (RAR, GZIP, etc.) do not generate significant gains due to the specificity of the sound data. However, it is possible to achieve a fairly significant level of compression of the audio information using special methods based on the analysis of the data structure and subsequent compression with some loss.

The real possibility of sound processing comparable in quality to existing analog examples did not appear until the late 1980s.

In 1988, the International Organization for Standardization (ISO) formed the MPEG (Moving Picture Experts Group) committee, whose main task is to develop standards for the encoding of moving pictures, sound and their combination. During the ten years of its existence, the committee has developed a series of norms on this subject. As a result, summarizing the extensive research in this area, several specific formats were recommended for storing data, which are excellent in quality of results and data flow.

There are currently three video storage standards: MPEG-1, MPEG-2, and MPEG-4.

Within the first two formats, there are also formats for storing audio information: Layer-1, Layer-2 and Layer-3. These three audio formats are defined for MPEG-1 and minor extensions are used in MPEG-2. The three formats are similar to each other, but use different levels of trade-off between compression and complexity.

Layer-1 is the simplest, it does not require significant compression costs, but it also provides a negligible compression ratio.

Layer-3 is the most time consuming and provides the best compression. Recently, this format has gained immense popularity. It is often called MP3. This name is associated with the extension of the audio files stored in this format.

The underlying idea behind all lossy audio compression techniques is to neglect the subtle details of the original sound that are beyond the reach of the human ear. Here several points can be highlighted.

Noise level . Sound compression is based on a simple fact: if a person is near a loud siren, they are unlikely to hear the conversation of the people who are nearby. And this happens not because a person pays close attention to a loud sound, but to a greater extent because the human ear actually misses out sounds that are in the same frequency range as a louder sound. This effect is called masking, it changes with the difference in volume and frequency of the sound.

The second point is the division of the audio frequency band into subbands, each of which is further processed separately. The encoding program extracts the loudest sounds in each band and uses this information to determine an acceptable noise level for that band. The best encoding programs also take into account the influence of adjacent bands. A very loud sound in one band can affect the masking effect and nearby bands.

Another point of the codification is the use of a psychoacoustic model based on the peculiarities of the human perception of sound. The compression used by this model is based on removing frequencies known to be inaudible, while more carefully preserving sounds that can be easily heard by the human ear. Unfortunately, there can be no exact mathematical formulas here.

The human perception of sound is a complex process, not fully understood, so the choice of compression methods is based on analyzing listening and comparing compressed sounds differently by teams of experts. But here there are practically limitless possibilities in the field of improving psychoacoustic models. Most of the existing algorithms to encode the human voice are based on the high predictability of said signal; Universal MPEG compression algorithms have tried to apply this technique with variable success.

Another compression technique is the use of so-called joint stereo. It is known that the human hearing aid can only determine the direction of the mid frequencies, the high and low sound, so to speak, separately from the source. This means that these background frequencies can be encoded into a mono signal. In addition to all this, compression uses the difference in the complexity of the flows in the channels.

Why mp3 is enough for you, but Lossless is not necessary

Why mp3 is enough for you, but Lossless is not necessary

mp3

Why mp3 is enough for you, but Lossless is not necessary
Did you finish the greenhouse? So you don’t need to lose, listen to high quality mp3.

Very often there are people who, in principle, despise compressed formats. You should not be guided by your opinion. The following mods that in the studio with a 90% probability will not hear the differences between compressed and uncompressed audio.

MP3 wasn’t invented just to reduce quality. It was developed by the Fraunchhofer Society, an association of applied research institutes in Germany. Later they came up with AAC, which could become the main compressed audio format … But it didn’t work.

Did you know that MP3 comes with variable (VBR) and constant (CBR) bit rate? The constant bit rate, due to the operation of the algorithm, is encoded each time as the first. Therefore, it can produce uneven quality, which means that not all sounds in this situation will be recorded in high quality.

Since MP3 has been around for a long time, it has many limitations. Bit width is 16-24 bits. The sample rate is represented by the following set of options: 8; 11,025; 12; sixteen; 22.05; 24; 32; 44.1; 48. The maximum bit rate does not exceed 320 kbps. The maximum number of channels is 2. But we are still talking about music, we still have to search for multi-channel recordings.

Now let’s see how MP3 is encoded. The illustration shows the time-frequency distribution of sound. Same recording: Audio CD, OGG file, MP3 well encoded. What we observe is that the pieces on the right and left almost completely coincide. This means that the MP3 file sounds almost the same as the original CD recording.

Human hearing and its limits – psychoacoustics

The fact is that the main task of the Fraunchhofer Society is the development of psychoacoustic models of human perception of sound. And here are many subtleties. The main thing is that we are not dolphins.

Second, there are certain restrictions on the number of sounds perceived simultaneously. A person cannot simultaneously hear more than 250 sounds of 24 ranges (in addition, the number of simultaneous sounds in the range is also quite small).

Third, the audible range is 16 Hz to 20 kHz and at the age of 60 it is reduced by almost half. Ideally, and during training (yes, you have to train it!).

All frequencies below 100 Hz are perceived not by the hearing cells, but … by the skin. Then the low waves are reflected in the ear canal; these waves are perceived as infrabass. (This is from the bone conduction area).
mp3_7_resize
Also, the number of cells that register acoustic waves is different for each one. But what is there? For each individual, their number in the right and left ear is different.

By the way, the perception of each ear is different. Change channels of your favorite song – get a new sound.

If you dig deeper, it turns out that each sound frequency is perceived only at a certain volume. When it is reached, the silence is replaced by a sharp and quite different sound. After that, a person can hear a lower sound of this frequency.

What you need to know about MP3

What you need to know about MP3

Mp3

What is MP3?

MP3 is short for MPEG Layer3. It is one of the transmission formats for storing and transmitting audio in digital form, developed by Fraunhofer IIS and THOMSON, and later approved as part of the MPEG1 and MPEG2 compressed video and audio standards. This scheme is the most complex scheme in the MPEG Layer 1/2/3 family. It requires the most amount of machine time to encode compared to the other two and provides higher encoding quality. It is mainly used for audio CD encoding.

The high degree of compactness of MP3 compared to other formats such as PCM (i.e. normal WAV- file) and similar formats while maintaining similar sound quality (considered 16-bit stereo at 44.1 kHz) is achieved using additional quantization according to a certain scheme, which minimizes the loss of quality. This is achieved by taking into account the peculiarities of human hearing, including the masking effect of 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, which causes a temporary decrease in the ear’s sensitivity to the current frame signal (simply, background sounds are eliminated, which are not heard by the human ear due to the presence at a given / previous moment of another – louder). It also takes into account the inability of most people to distinguish between signals that are below a certain power level,

This is called adaptive coding, and it allows you to save on the less perceptually significant sound details. The compression ratio (and therefore quality) is not determined by the format, but by the width of the data stream when encoded in MP3. The bit rate when encoding a signal similar to an audio CD (44.1 kHz 16 bit stereo) varies from the largest, 320 kbs (320 kilobits per second, also kbs, kbps or kb / s), up to 96 kbs and less.

Why MP3?

MP3 has two huge advantages over other formats available today. It is true that MicroSoft is trying to squeeze MP3 with its new WMA format, and there are also alternative VQF and AAC formats, but they have not yet received proper distribution and the quality is often a little worse. However, WMA is still, in fact, closed for free use, so you have problems with various encoding / listening / maintenance programs (although, who doubts MicroSoft’s mobilization capabilities :-).

The first advantage of MP3 is that none of the existing similar formats can yet be said to fully guarantee the stable preservation of sound quality at sufficiently high bit rates, except MP3, which has stood the test of time with dignity.
The second, no less important advantage: over the next few years, and perhaps the entire decade, MP3 has become the de facto standard, as the parties that use it (eg me 😉 have made a lot of investments in him, including digital radio stations. There are also many easy-to-use software programs written for MP3. Now the production of hardware MP3 players has been launched, both pocket and car. Thus, MP3 became the first massively recognized audio storage format after Audio CD (although it is often illegal).

The most famous encoders

Today there are 3 main sources that have created programs to encode MP3 music. These are Fraunhofer-IIS, Xing Technologies, and ISO itself, which adhere to the ISO MPEG standard developed by it.
Most of the encoders created to date use modified code from one of these organizations. Fraunhofer-IIS based encoders are not very fast, but very high quality, quality optimized for low bit rates.

128 kbps (11: 1)
The most popular bit rate today. The 11: 1 compression ratio is of course an argument, especially for the internet, where every kilobyte counts. However, the high frequencies are not very well preserved and there is some distortion in the sound. At the same time, I can safely say that on an ordinary computer, for example, using an ordinary sound card, computer speakers, albeit of good quality, or output through a simple recorder to your speakers (using the input for a External CD, like me), the difference will not be noticeable unless you are a sound expert.
However, in normal speakers (at least large and expensive), the lack of high frequencies is quite noticeable.

Digital audio formats: the MP3 phenomenon

Digital audio formats: the MP3 phenomenon

MP3 format

The MP3 music format (MPEG-1 Layer 3) is one of the most widely used digital audio formats in the world.

MP3 format MP3 format : An Overview

It is compatible with all portable and stationary audio devices. In May 2017, the developers of the format announced his “death”. On April 23, 2017, the Technicolor and Fraunhofer IIS licensed commercial program was canceled: the last patent included in the program expired, making the format standard in the public domain.
Can we say that the days of the most popular format are numbered? MP3 development began in the late 1980s at the Fraunhofer Institute for Integrated Circuits (IIS). In 1987, the University of Erlangen-Nuremberg and Fraunhofer IIS teamed up to work on the EU147 EUREKA Digital Audio Broadcasting (DAB) project. The first result of the alliance’s work was the LC-ATC codec, which made it possible to encode stereo music in real time.

The next step was the development of an optimal frequency domain (OCF) coding algorithm, which already had some of the characteristics of the future MP3 codec. For the first time, it is possible to encode music in good quality at 64 kbps for a mono signal. OCF was the beginning of the path towards standardization MPEG (Moving Picture Expert): an organization, responsible for the development and implementation of international standards for the compression and transmission of digital video and audio content.

In 1989, MPEG received 14 proposals for the implementation of an audio coding standard, so participants were invited to combine their developments. This led to the emergence of four potential candidates, including MUSICAM from the Institute for Broadcasting Technology IRT and Philips and ASPEC (Adaptive Spectral Perceptual Entropy Coding), which is the result of further enhancements to the OCF Fraunhofer IIS in addition to contributions from the University of Hannover in collaboration with AT&T and Thomson.

After extensive testing, MPEG proposed combining MUSICAM and ASPEC to create a family of three encoding methods: Level 1: a low-complexity version of MUSICAM; level 2 – MUSICAM codec; Level 3 (later called MP3): based on ASPEC. Technical development of the MPEG-1 standard was completed in December 1991. In 1994, Fraunhofer IIS introduced the world’s first MP3 encoder, the L3enc, and in 1995 the Fraunhofer researchers unanimously accepted “.mp3” as the file extension for MPEG Layer 3 [1].

Thanks to the compression algorithm used in the MP3 audio format, the size of the data required to reproduce the recording and ensure the quality of sound reproduction is significantly reduced to 10-12 times the original, depending on the recording bit rate. . Bit rate refers to the encoding / decoding rate of a digital audio stream; sound quality improves with increasing bit rate. The MP3 format has the following bit rates: 32 kbps (very low quality, acceptable only for voice), 96 kbps, 128 kbps (medium quality), 160 kbps, 192 kbps, 256 kbps, 320 kbps (maximum optimal quality). The principle of the compression algorithm is as follows: during the compression process, the audio codecs analyze the signals, focusing on the audible fragments, which are saved for later playback or transmission.

This rules out sounds beyond the perception range of the human ear (20 to 20,000 Hz). That is why MP3 is called lossy. There are three ways to encode MP3 files: constant bit rate (CBR), variable bit rate (VBR), and medium bit rate (ABR). CBR is the default encryption mode. In this mode, the bit rate is constant for the entire file. This means that each part of the MP3 file uses the same number of bits. Regardless of the complexity of a piece of music, the encoder uses the same bit rate, so the quality of the final file is variable.

Complex parts will be of lower quality than simpler ones. The main advantage of this mode is that the size of the final files does not change and can be accurately predicted. When encoding in VBR mode, the user selects the desired quality on a scale of 9 (lowest quality, highest distortion) to 0 (highest quality / lowest distortion). The codec then tries to maintain a certain quality throughout the file by choosing the optimal number of bits for each part of the audio recording. The main advantage is the ability to specify the level of quality to be achieved, but the significant disadvantage is the unpredictability of the final file size.

How much compresses an MP3

MP3 compression was an engineering response to the problem of digital storage and its large memory resource requirements. A conventional digital signal called PCM (Pulse Code Modulation) could easily require up to 10 Megabytes of memory per minute. This would represent about 30 Mb for a three minute song.
That requirement for storage memory could be handled by any computer if it were a few files, but when talking about three thousand songs the numbers become worrying. As if this were not enough, there is the problem of the Internet and its current transmission speeds. In the case of telephone lines, they have a limitation on their transmission bandwidth, so very large or heavy files represent a problem for conventional network traffic.

MPEG3 compression is considered the sound part of the original MPEG1 format that was intended for cinematography. Its abbreviations, Moving Picture Experts Group come from the committee that was created by the ISO Organization (international Standards Organization) and IEC ((International Electrotechnical Commission) to develop this format. Its principle is based on the Psychoacoustic model.

The human ear is known to discriminate sound according to its limitations. According to subject matter expert Paul Sellars, “If you hear solitary applause in a room, it will surely sound loud, but if it is preceded by the sound of a gunshot, it will sound fainter. The same thing happens in a room when you record a rock band, at a certain moment the strongest sound guitar in the mix, until the moment the drummer plays a certain cymbal, at which point the guitar will seem to attenuate “This phenomenon is used by the MP3 algorithm to perform its compression . I once explained it in the article that talked about ATRAC compression of the Minidisc.

The MP3 format divides the sound into 32 sub-bands, which allows it, according to the Psychoacoustic model on which it is based, to give priority to one element over another. At a certain moment in the material we can have a predominant low frequency sound of the kick drum, a high frequency of the cymbal and the vocalist at the same time. The algorithm is not that it eliminates two of them, but that it dedicates less storage space to them.

The mathematical part used with MP3 compression goes through the Shannon-Nyquist theorem, which states that for a wave to be properly reproduced in PCM digital format, its frequency of takes (Sampléo) must be twice the highest that is want to reproduce. In this case if we want to reproduce the frequency of 22.5KHz, (The auditory range oscillates between 20Hz-20KHz), our sampling frequency should be 44.1KHz.

The Fast Fourier Transform (FFT) is also used, which as we know can decompose a complex wave (PCM material) into a fundamental wave with its harmonics, all from its amplitude. The Discrete Cosine Transform is also used, which is based on the FFT but only using the real numbers

UNTIL IT IS RECOMMENDED

These formats will continue to be perfected and emerge, but it should be understood that despite being disseminated there may be details that will not be perceived. In other words, for serious Audio work this format should not be used.

Some improvements can be made by looking for compressors that have a better ratio, such as 224, 256 and 320 Kbps. You can also consider using VBR (Variable Bit Rate) encoding where musical passages with greater dynamic complexity are treated with a higher rate. storage in contrast to the simplest. However, this will bring other complications because not all the reproducers can handle them.