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What are MP3 files?

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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.

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Mp3 is still the best audio format

Does this phrase remind you of the fear of being dominated by “storage space” in those years?

This also shows from the side that in the years when storage technology was underdeveloped, MP3, which could compress the file size, was like a “black technology” for users at that time.

But after MP3 became popular, controversy also followed. Since MP3 allows anyone to rip CDs, make, copy and distribute music files to anyone, to a certain extent, it can be said that MP3’s dominance of the “global music format” is based on rampant pirated music. . However, from another perspective, these problems that MP3 brings with it are also forcing the music industry to start thinking about legalizing digital music.

What is the origin of the AAC?

AAC (Advanced Audio Coding) and MP3 are lossy compressed audio data formats. In 1997, it was jointly developed by Fraunhofer IIS, Dolby Laboratories, AT&T, Sony, and other companies to replace the MP3 format.

AAC is much younger than MP3, and the technology it was born into has far surpassed MP3. According to public information, AAC uses a new algorithm for encoding, which is more efficient and has a higher “cost performance”. As Fraunhofer IIS says, AAC looks better than MP3 in every way.

Although AAC is better than MP3, but MP3’s market share is too high. It is almost impossible for AAC to change the “MP3 inertia” of the user.

Fortunately, not everyone refuses to support AAC. Apple is one of the supporters of AAC. It started to provide AAC format for users to download as soon as possible. Li Chujie, founder of MOZIK music player, said:

In 2006, I bought my first iPod from Apple and it supported AAC files. All MP3 formats that I download from the Internet are converted to AAC files. It was also through this that I noticed the difference in sound quality and byte size between MP3 and AAC.

In the early days, apart from Apple, there were devices like Sony Walkman (NWZ-A, NWZ-S, NWZ-E, NWZ-X series), Nintendo NDSi, and Meizu that supported AAC. Later, as more and more music lovers paid attention to AAC, users gradually recognized its advantages. Both the number and category of hardware devices that support AAC have grown rapidly.

Mp3 is still the best audio format

Mp3 Audio

MP3, the king of music formats, has retired from the stage of history and been replaced by AAC. What is the origin?

Even someone who knows nothing about audio encoding may not have listened to MP3.

This audio format emerged in the 1990s. For most post-80s and post-90s generations, having an MP3 player capable of playing this audio format is one of the best childhood memories.

But now it’s time to say goodbye to MP3s. Recently Fraunhofer IIS, which invented this audio format, announced that it had rescinded some patents related to MP3 and officially took it off the stage of history.

Furthermore, Matthias Rose, director of the institute, said:

The Advanced Audio Coding (AAC) format has become the standard for mobile music and video downloads. This format is more efficient, more functional, and has a lower bit rate than MP3, and is used in both broadcast TV and radio to deliver high-quality audio.

This means that Fraunhofer IIS wants to replace MP3 with the AAC format below.

How did MP3 become the king of music formats?
There are generally two main types of audio file formats, one is a lossless compressed data format and the other is a lossy compressed data format.

MP3 is the most popular digital audio encoding and lossy compression format, and is supported by almost all terminals and software.

The birth of MP3, but also for 30 years. In 1987, Fraunhofer IIS in Germany began the development of the music compression format. They spent 4 years to improve the algorithm and sound quality, and finally developed this very advanced lossy compression audio format in 1993 and determined the extension .mp3.

The purpose of developing this audio compression technology is to greatly reduce the amount of audio data. Knowing that hard drives at the time were relatively small (like 500 MB), compressing all compressible data was essential for any computer user.

How does Fraunhofer IIS do it? During the invention of MP3, developers studied psychoacoustics and determined that some of the audio data is not important to the human ear. MP3 achieves the purpose of compressing the file size by discarding this part of the data, and can successfully compress the audio file to 1/10 of the original size.

After that, from the first half of 1995 to the end of the 1990s, MP3 began to flourish on the Internet. The “much smaller file but also good sounding” MP3 began its era of dominating audio formats. There were even many people at the time who believed that all digital music files were MP3s.

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