What is MP3?

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What is MP3?

MP3 is a type of audio file that is compressed using a specific algorithm to reduce its size while maintaining its sound quality. MP3 stands for MPEG-1 Audio Layer 3 and was developed in the 1990s. Since then, it has become one of the most popular audio file formats in the world.

How does MP3 compression work?

MP3 compression works by removing sounds that are less important to the human ear. This process is called psychoacoustic analysis. The MP3 algorithm uses this analysis to determine which sounds to keep and which ones to discard.

For example, when you listen to music, you may not be able to hear sounds that are below a certain volume level. The MP3 algorithm takes advantage of this fact by removing these quieter sounds from the audio file.

In addition to removing quiet sounds, the MP3 algorithm also removes sounds that are masked by louder sounds. For example, if a loud drumbeat is playing at the same time as a quieter guitar solo, the algorithm will remove the guitar sounds that are masked by the drumbeat.

What are the benefits of using MP3 files?

There are several benefits to using MP3 files. One of the main benefits is that they take up less space than other audio file formats. This makes them easier to store and share.

Another benefit of using MP3 files is that they can be played on a wide variety of devices. Many music players, smartphones, and computers can play MP3 files without the need for additional software.

How are MP3 files encoded?

To create an MP3 file, you need to encode it using an MP3 encoder. The encoder takes the raw audio data and applies the psychoacoustic analysis and compression algorithms to create the final MP3 file.

There are several software programs that can be used to encode MP3 files, including iTunes and Audacity. Many music players also include built-in MP3 encoders.

What are the drawbacks of using MP3 files?

One of the main drawbacks of using MP3 files is that the compression process can result in a loss of sound quality. While the human ear may not be able to hear the sounds that are removed during the compression process, some people may be able to notice a difference in sound quality between an MP3 file and a file that has not been compressed.

Another potential drawback of using MP3 files is that they may not be suitable for all types of music. The psychoacoustic analysis used by the MP3 algorithm is based on the characteristics of typical music. As a result, some types of music, such as classical music, may not be compressed as effectively as other types of music.

What is MP3 Decoding?

MP3 decoding is the process of converting an MP3 file back into a digital audio signal that can be played by speakers or headphones. This process is the reverse of MP3 encoding and involves decompressing the audio data that was removed during the compression process.

To decode an MP3 file, you need to use an MP3 decoder. Many music players and software programs include built-in MP3 decoders that can be used to play MP3 files.

MP4Gain for Converting and Normalizing Audio and Video Formats

If you are looking to convert and normalize various audio and video formats, including MP3, you may want to consider using MP4Gain. MP4Gain is a software program that allows you to adjust the volume and normalize the loudness levels of your audio and video files.

In addition to MP3, MP4Gain supports a variety of other audio and video formats, including MP4, AAC, FLAC, and more. With MP4Gain, you can ensure that your audio and video files are all at a consistent volume level, making it easier to listen to and enjoy your media collection.

Conclusion

MP3 is a popular audio file format that is widely used for storing and sharing music. The MP3 algorithm uses psychoacoustic analysis to compress audio data and remove sounds that are less important to the human ear. While MP3 files have several benefits, they may not be suitable for all types of music, and the compression process can result in a loss of sound quality.

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Structure of an mp3

audio compression

The MP3 format began in the mid-1980s and the Fraunhofer Institute in Erlangen, Germany, was committed to high-quality, low-data-rate audio coding.

MP3 audio compression includes encoding and decoding in two parts. Encoding is converting the data in the WAV file into a highly compressed bitstream format, and decoding is accepting the bitstream and reconstructing it into the WAV file.

MP3 uses the distortion algorithm of Perceptual Audio Coding (PerceptualAudioCoding). The frequency range of sound perceived by the human ear is from 20 Hz to 220 kHz. MP3 cuts out a lot of redundant signals and irrelevant signals. The encoder transforms the original sound into the frequency domain through a hybrid filter bank. Using the psychoacoustic model, it is estimated that it may simply be The perceived noise level is quantized and converted to Huffman coding to form an MP3 bitstream. The decoder is much simpler and its task is to extract the sound signal from the encoded spectral line components through inverse quantization and inverse transformation.

When compressing audio data, the original sound data is first divided into fixed blocks, and then direct MDCT is performed. MDCT itself does not perform data compression, but only converts a set of time-domain data to frequency-domain data to obtain time-domain data. In case of change, the direct MDCT converts the value of each block into 512 MDCT coefficients. Quantization compresses data, and when bits are allocated to transformed samples after quantization, it is necessary to consider making the entire quantized block the smallest, which becomes lossy compression. When decompressing, the 512 coefficients are restored to the original sound data by reverse MDCT, and the original sound data before and after are inconsistent, because redundant and irrelevant data are removed during the compression process.

MP3 file structure
MP3 files are roughly divided into three parts: TAG_V2(ID3V2), Frame, TAG_V1(ID3V1)

ID3V2 Contains information such as author, composer, album, etc., the duration is not fixed, expanding the amount of information of ID3V1
framework

A series of frames, the number is determined by the file size and frame length

The length of each frame can be variable or fixed, determined by the bit rate.

Each FRAME is divided into two parts: frame header and data entity

The frame header records the bitrate, sample rate, version, and other mp3 information, and each frame is independent of each other.

ID3V1　　　 Contains author, composer, album and other information, length is 128BYTE

Structure of an mp3

The full name of MP3 is MPEG Audio Layer3, which is an efficient computer audio coding scheme.

Structure of an mp3

It converts audio files into smaller files with .MP3 extension with a higher compression ratio and basically keeps the sound quality of the original file. MP3 is part of the ISO/MPEG standard. The ISO/MPEG standard describes audio compression using a high-performance perceptual coding scheme. This standard has been continuously updated to meet the pursuit of “high quality, low volume”. MPEGLayer1, Layer2 , and Layer 3 have now formed three audio codec schemes. The compression rate of MPEGLayer3 can reach from 1:10 to 1:12. A 1M MP3 file can play for 1 minute, while a 1 minute CD-quality WAV file (44100 Hz, 16-bit, two channels, 60 seconds) will take up 10M of space. , A 650M MP3 disc should play for more than 10 hours, while a CD with the same capacity should play for about 70 minutes. The advantages of MP3 are unmatched by CD.

MPEG audio standard
MPEG (Motion Picture Experts Group) is a moving picture expert group under ISO, and the MPEG standard it creates is widely used in various multimedia. MPEG standards include video and audio standards, among which MPEG-1, MPEG-2, MPEG-2AAC, and MPEG-4 audio standards have been developed.

The MPEG-1 and MPEG-2 standards use the same family of audio codecs: Layer 1, 2, and 3. A new feature of MPEG-2 is the use of low sample rate expansion to reduce data traffic, and another feature is multi-channel expansion, which increases the number of main channels to five. The MPEG-2AAC (MPEG-2 Advanced Audio Coding) standard was released by FraunhoferIIS and AT&T in 1997, with the goal of significantly reducing data traffic. MPEG22AAC adopts the Modified Discrete Cosine Transform (MDCT) algorithm and the sampling rate can be between 8 KHz and 96 KHz. The number of channels can be between 1 and 48.

MPEG Audio Layer1, 2, and 3 use the same filter bank, bitstream structure, and header information, and the sample rate is either 32 KHz, 4411 KHz, or 48 KHz. Layer1 is designed for DCC (DigitalCompactCassette) digital compression tape, the data rate is 384kbps, Layer2 has made a compromise between complexity and performance, and the data rate has been reduced to 256kbps-192kbps. Layer 3 was designed for low data traffic from the start, and data traffic ranges from 128 kbps to 112 kbps. Layer 3 adds MDCT transform, making its frequency resolution 18 times higher than Layer 2. Layer 3 also uses EntropyCoding similar to MPEGVid2eo, reducing redundant information. The vast majority of MP3s use the MPEG21 standard.

What are MP3 files?

The audio format is directly related to the quality and purpose of the audio track, i.e. where and on which device it will be played and what is its purpose.

What are MP3 files?

But before you can figure out the difference between them and choose the best audio format for your music, you need to know what categories they fall into. Let’s keep going!

Uncompressed audio is like a picture, and uncompressed audio is of better quality, larger file size, safer to copy, and nearly identical in detail to the original sound.

WAV is the most widely used of these audio formats and plays music just as accurately as it records it.

compressed audio
When music is compressed, the files become smaller and can be easily stored on a device. Due to this advantage, users tend to choose compressed audio more.

However, it must be remembered that some audio formats in this category may lose quality depending on the option selected, just like MP3 and AAC.

What is the best audio format?
As we said before, the first step in deciding on an audio format is to know the final objective of the track. Whether it’s for music lessons, performances, karaoke, auditions, or recording versions, you need to understand the pros and cons of each option.

WAV
WAV (Waveform Audio File Format) is an uncompressed format and therefore requires ample storage space. This is suitable for those who already work with music, such as subject matter experts, or users who want to edit audio.

At high fidelity rates, WAV faithfully reproduces the elements and characteristics of the original soundtrack. Also, this format allows you to choose between different sample rates and bit rates and can be used on multiple platforms.

FLAC
FLAC (Free Lossless Audio Codec) is one of the most widely used compression formats by music lovers these days.

Digital audio encoding allows you to preserve its quality, but the resulting file will be smaller. Over the years, this format has become more widely used and compatible with different devices and platforms.

FLAC is free and open source, ready to use and can be easily played on smartphones and other devices.

MP3
Before deciding on the best audio format, it is worth taking a look at the most famous format in the world of music: MP3.

MP3 is one of the leading audio compression formats, and has become synonymous with the convenience and efficiency of producing files quickly, with smaller files, and at a certain level of quality.

Many devices and programs can play this format. But MP3 is difficult to use in professional audio processing and advanced audio editing.

As is known, this format exists on almost all platforms and is ideal for sharing audio.

Another interesting factor is its bitrate, although in a compressed format it can vary depending on the user’s objectives and quality improvements.

AAC Like MP3, Advanced Audio Coding (AAC) is a more efficient audio format than its predecessor.

If you need to create smaller files with less storage space, AAC is a great choice, reducing the file size for the user while maintaining a high-quality audio track.

Compatible with different platforms and devices, it is convenient to apply in different situations.

Analysis of the above audio formats leads to the conclusion that it is impossible to say which format is better than the other, just that each target has its own ideal format. So before downloading or uploading a file, check what platform the music will play on and what it is for.

What are MP3 files?

A file with the .mp3 extension is a digitally encoded file format for audio files, officially based on MPEG-1 Audio Layer III or MPEG-2 Audio Layer III.

It was developed by the Moving Picture Experts Group (MPEG) using Layer 3 audio compression. The compression achieved by the MP3 file format is 1/10 the size of a .WAV or .AIF file. This format offers the advantage of streaming such audio files over the Internet for online listening, which was previously not possible due to the large size of audio files. The sound quality of MP3 audio files can be controlled by setting parameters such as bit rate, sample rate, common or normal stereo.

A brief history of MP3

The MP3 format was invented and developed by a German company, Fraunhofer-Gesellshart. The algorithm has licensed patents for the compression techniques it uses. Here’s a helpful MP3 schedule:

• 1987 : The Fraunhofer Institute in Germany begins research on high-quality, low-bitrate audio coding. It’s called the EUREKA project EU147, Digital Audio Broadcasting.

• January 1988: The Moving Picture Experts Group (MPEG) is formed.

• **April 1989**: Fraunhofer patented the MP3 in Germany.

• 1992-Dieter Seitzer, who helped Fraunhofer with his research, integrated his audio encoding with MPEG-1.

• 1993 – Publication of the MPEG-1 standard.

• 1994 – The MPEG-2 standard was developed and released a year later.

• November 26, 1996 : US patent for MP3 is published.

• September 1998 – Fraunhofer begins to enforce the patent. People who used the MP3 audio codec paid Fraunhofer a license fee.

• February 1999 – SubPop, a record label, releases music in MP3 format, the first to do so.

• 1999 – The first portable MP3 player appears.

File format MP3##
MP3 files consist of MP3 frames, where each frame consists of a header and a data block. Frames are not independent and generally cannot be mined at arbitrary frame boundaries. The data blocks of a file contain frequency and amplitude information about the audio. The sync word in the header identifies the start of a valid frame. This is followed by 3 bits where the first bit indicates that it is an MPEG standard and the remaining 2 bits indicate that layer 3 is used; therefore, MPEG-1 Audio Layer 3 or MP3. After this, the value will vary depending on the MP3 file. ISO/IEC 11172-3 defines the range of values for each part of the header and the header specification. Most current MP3 files contain ID3 metadata, which precedes or follows the MP3 frame, as shown. Data streams may contain an optional checksum.

MP3: complete analysis of the audio format

WMA: WMA is the file format encoded by Windows Media Audio, developed by Microsoft.

WMA is not aimed at the independent market, but at the network! The competitor is the well-known Real Networks in the online media market. Microsoft claims that at a bit rate of just 64 kbps, WMA can achieve sound quality close to CD. Unlike the previous encoding, WMA supports the anti-copy function. Supports adding protection via Windows Media Rights Manager, which can limit playback time, number of playback times, and even playback machine, etc. WMA supports streaming technology, that is, play while reading, so WMA can easily realize online streaming. Because it is a Microsoft masterpiece, Microsoft has added support for WMA in Windows. WMA has excellent technical characteristics. With vigorous promotion, this format has been accepted by more and more people.

WAV: This is an old audio file format developed by Microsoft. WAV is a file format that complies with the PIFF Resource Interchange File Format specification. All WAVs have a file header, the encoding parameters of this file header audio stream. WAV does not have a strict regulation on the encoding of audio streams. In addition to PCM, almost all encodings that support the ACM specification can encode WAV audio streams. Many friends do not have this concept. Let’s take AVI as an example, because AVI and WAV are very similar in file structure, but AVI has one more video stream. There are many types of AVIs we have come into contact with, so we often need to install some decoders to watch some AVIs. DivX, which we have come into contact with a lot, is a type of video encoding. AVI can use DivX encoding to compress video streams, and of course we can also use other code compression. Similarly, WAV can also use a variety of audio codecs to compress its audio stream, but we commonly use WAV whose audio stream is processed by PCM encoding, but this does not mean that WAV can only use PCM codec, it is also you can use MP3 codec. in WAV Just like AVI, as long as the corresponding Decode is installed, you can enjoy these WAVs. On the Windows platform, WAV based on PCM encoding is the best supported audio format. All audio software can support it perfectly. Because it can meet higher sound quality requirements, WAV is also the preferred format for music creation and editing. Suitable for storing musical material. Therefore, WAV based on PCM encoding is used as an intermediate format and is often used in the conversion of other encodings, such as MP3 to WMA.

Ogg Vorbis: The so-called MP3 killer! What is the origin of Ogg Vorbis? OGG is the project name of a large multimedia development program, which will involve coding development in aspects such as video and audio. The whole purpose of the OGG project plan is to provide a completely free media encoding solution for anyone! OGG’s belief is: OPEN! FREE! The word Vorbis is the name of a “playboy” character in the fantasy novel “Small Gods” by Terry Platjat. This term became the official name for audio encoding in the OGG project. At present, Vorbis has been successfully developed and an encoder has been developed. Ogg Vorbis is a high quality audio coding scheme. Official data shows that Ogg Vorbis can achieve better sound quality than MP3 at relatively low data rates. This Ogg Vorbis encoding is also much more advanced than MP3, which was successfully developed in the 1990s. It can support multiple channels. What does this mean? This means that Ogg Vorbis can encode all channels with the support of SACD, DTSCD, DVD AUDIO ripping software (currently there is no such software), instead of MP3 it can only encode 2 channels. The rise of multi-channel music has brought revolutionary changes in music appreciation, especially when appreciating the symphony, it will bring more sense of presence. This revolutionary change cannot be adapted to MP3. Like MP3, Ogg Vorbis is a flexible and open audio codec that allows for significant sound quality adjustments and further algorithm improvements once the codec has been fixed. Therefore, its sound quality will be better and better. Just like MP3.

MP3: complete analysis of the audio format

Although MP3 technology has been fully disclosed now, its specific technical details still have some depth.

Therefore, this editor will provide a detailed explanation of some technologies and hope that netizens will laugh at it.

Sampling rate:

A digital audio system reproduces the original sound by converting the waveform of the sound wave into a series of binary data. The equipment used to accomplish this step is an analog-to-digital (A/D) converter, which samples the sound wave at a rate of tens of thousands of times per second. , each sample records the state of the original analog sound wave at a given time, called a sample.

A series of samples can be connected to describe a sound wave. The number of samples per second is called the sample rate or sample rate, and the unit is HZ (Hertz). The higher the sample rate, the higher the frequency of the sound wave that can be described. For each sampling system, a certain storage bit (number of bits) is allocated to express the sound wave amplitude state of the sound wave, which is called the sampling resolution or sampling precision. Increase the dynamic range of 6db, that is, the dynamic range of 6db, a 2-bit digital audio system expresses thousands of states, that is, the dynamic range of 12db, and so on. If you keep increasing the number of bits, the sampling precision will increase at a very fast rate. It can be calculated that 16 bits can express 65536 states, corresponding to 96db, and 20 bits can express 1048576 states, corresponding to 120db. 24bit can express up to 16777216 states. Corresponding to the dynamic range of 144db, the higher the sampling precision, the more delicate the restoration of sound waves. (Note: dynamic range refers to the range of the sound from the weakest to the loudest.) The hearing range of the human ear is usually 20HZ~20KHZ.

According to the Nyquist sampling theorem, sampling at twice the frequency of a sine wave can completely restore the waveform, so the sampling frequency of a digital recording wave is directly related to its rate of sampling. higher refresh rate. For example, sampling with a sampling frequency of 44.1 KHZ can restore the highest frequency of 22.05 KHZ — this value is slightly higher than the hearing limit of the human ear (Note: MD can be recorded, please For example, the sampling rate of R900 is 44.1KHZ, it also has a sampling frequency converter, which can convert the input 32KHz/44.1KHZ/48KHZ into the machine’s standard sampling rate 44.1KHZ. The refresh rate is enough to truly record and reproduce the sound that everyone can distinguish, so the sampling specification of CD audio is defined as 16bit 44KHZ, even if 16bit recording is actually done with high precision electronic components that are almost impossible to manufacture in real life in the most ideal environment there will still be issues like sound filtering and positioning people can still detect some tiny distortions so much Many professional digital audio systems have used 18-bit or even 24-bit for recording and playback.

Existing sampling methods:

MP3: The full name of MP3 should be MPEG1 Layer 3 audio file. MPEG (Moving Picture Experts Group) translates into Chinese as Moving Picture Experts Group, which refers to the compression standard for moving video . The MPEG audio file is the sound part in the MPEG1 standard. It is called MPEG audio layer, which is divided into three layers according to compression quality and encoding complexity, namely Layer-1, Layer2, Layer3, and corresponds to the three sound files of MP1, MP2 and MP3 , respectively. , and use different layers according to different purposes. The higher the MPEG audio encoding level, the more complex the encoder and the higher the compression ratio. The compression ratio of MP1 and MP2 is 4:1 and 6:1-8:1 respectively, while the compression ratio MP3 compression ratio is up to 10:1-12:1, that is, one minute of CD-quality music requires 10 MB of storage space without compression, and only about 1 MB after encoding with MP3 compression. However, MP3 uses a lossy compression method for audio signals. To reduce the degree

When did mp3 music files first appear?

MPEG-1 Audio Layer 3, often referred to as MP3, is one of the most popular lossy compression and digital audio encoding formats today.

There is no noticeable drop in sound quality compared to the original uncompressed audio. It was invented and standardized in 1991 by a group of engineers at the Fraunhofer-Gesellschaft research organization in Erlangen, Germany.
MPEG-1 Audio Layer 3, often referred to as MP3, is one of the most popular lossy compression and digital audio encoding formats today. There is no noticeable drop in sound quality compared to the original uncompressed audio. It was invented and standardized in 1991 by a group of engineers at the Fraunhofer-Gesellschaft research organization in Erlangen, Germany.

The audio format supported by the MP3 player is not only MP3 format, but also WMA, WAV, MP3Pro, ASF, AAC and VQF, etc. The WMA format can reach CD quality when compressed to 64 kbps, and output is only half the size of the corresponding MP3 file. This is very important for models with only 32 MB of flash memory. WMA and RA formats are supported, which means FlashMemory space is almost doubled. If it’s hard, be sure to ask this question when purchasing.
Among all the music formats supported by MP3, the most common ones are MP3, WMA and WAV. Others are unpopular or too bulky to be practical.

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.