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MP3 file format

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MP3 file format

The full name of MP3 is MPEG-1 or MPEG-2 Audio Layer III, which is a popular format for digital audio coding and lossy compression of minor parts, to achieve the purpose of compressing into smaller files.

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The MP3 format was invented in the mid-1980s by a group of engineers at the Fraunhofer research organization in Erlangen, Germany, and standardized in 1991. The association is committed to research in low-rate, high-quality sound coding of data. Although MP3 is a lossy compression format, for the listening experience of most users, the sound quality of MP3 does not have a noticeable decrease compared to the original uncompressed audio.

Later, with the popularization of the MP3, it had an impact and influence in the music industry.

MPEG audio standard
MPEG (Motion Picture Experts Group) is a moving picture expert group under ISO, and the MPEG standard formulated by it is widely used in various multimedia. MPEG standards include video and audio standards, from 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: Layer1, 2, 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 launched by FraunhoferIIS and AT&T in 1997 to significantly reduce data traffic. The Modified Discrete Co2sine Transform (MDCT) algorithm adopted by MPEG22AAC, the sampling rate It can be between 8KHz and 96KHz, and the number of channels can be between 1-48.

All three layers of 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, with a data rate of 384kbps.
Layer2 balances complexity and performance, and data traffic drops to 256kbps-192kbps.
Layer3 was designed for low data traffic from the beginning, and the data traffic is 128Kbps-112Kbps. Layer3 adds MDCT transform, which makes its frequency resolution 18 times than Layer 2. Layer3 also uses EntropyCoding similar to MPEGVid2eo Redundant information is reduced.
Currently, most MP3s use the MPEG21 standard.

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Change the bit rate of an MP3 file

Do you want to change the bit rate of an MP3 file?

This can be useful, for example, if you need to reduce the size of an MP3 file. A 320 kbps MP3 file, the highest bit rate allowed for an MP3 file, can be lowered to 192 kbps to significantly reduce the size of the MP3 file.

There will be some loss in quality, but the difference will be negligible to most listeners using standard speakers or headphones. If you’re an audiophile, chances are you’ll never use the MP3 format outside of expensive audio equipment.

Most likely, you are using a lossless format, such as compressed or uncompressed PCM audio, WAV, AIFF, FLAC, ALAC, or APE. Uncompressed PCM audio files are approximately 10 times larger than CD-quality MP3 files.

The MP3 format is a lossy format, which means sacrificing audio quality to keep file sizes relatively small. Almost all sites will tell you that you shouldn’t convert lossless audio files to MP3 unless you can afford to lose some audio quality.

Almost all the time. The only time it might make sense is if you have a bitrate audio file in a low quality format like WAV. For example, it might make sense to convert a 96 kbps WAV file to MP3, but only if you choose a bit rate of 192 kbps or higher. A higher bit rate in an MP3 file will allow it to maintain the same quality as a WAV file even though it has a lower bit rate.
The second thing to read is that you should never switch to a lower bitrate. bitrate stream to a higher bitrate stream and hope it sounds better. You cannot gain quality by increasing the bit rate. This is absolutely true. If you try to convert the bitrate, it will actually reduce the quality of the MP3 file.

Bit rate

Bit rate refers to the number of bits (bit) transmitted per unit of time, in bps (bit per second).

Bit rate is also known as “binary bit rate”, commonly known as “code rate”. Indicates the number of bits transmitted per unit of time. It is used to measure the transmission speed of digital information, often written as bit/sec. According to the number of bits occupied by each image storage frame and the transmission bit rate, the digital image information transmission speed can be calculated [1].
In modern digital communication, the transmission volume of digitized video and other information is large, so it is often measured in kilobits per second or megabits per second, which are written as kbit/sec (or kbps) and Mbit/sec. (or Mbps respectively). ). For example, the amount of information digitized from an ordinary color TV signal can reach 216 Mbit/sec. A good digital broadcast channel can transmit dozens of color TV programs, and its capacity can reach several gigabits or gigabits per second (written as Gbit/sec or Gbps) [1] .
Bitrate is often used to measure the quality of video files.
Bitrate is often used to measure the quality of video files.
flexibility edit stream
Because each network is unique and each access line has different conditions (such as length, attenuation, crosstalk environment, etc.), access lines from different telephone companies must support different data rates. For ADSL and VDSL modems, it is best to set the data rate to one of many possible data rates. For example, DMT-based ADSL and VDSL can theoretically change the tariff at fine intervals, and CAP-based RADSL (Rate Adaptive ADSL) also provides some flexibility in tariff configuration [2].
However, telephone companies may want to limit xDSL service to a small set of rates sufficient to provide a variety of services. If a limited set of tariffs can be adapted to a wide range of services, then the management of the services in this case is simpler than in the case of variable tariffs. Telephone companies want the choice of modem speed to be under the control of the network, not the user [2] .
In this mode, the selection of the transmission rate set of the xDSL network must be prudent. In this case, there is a possibility that two adjacent systems receive traffic at very different rates and the system must be able to handle such a situation. The other model, the “best match” approach using adaptive rate ADSL (similar to a voiceband modem), is more beneficial to new network operators and Internet Service Providers (ISPs) [2] .
Transmission control method
Most bit rate control schemes consist of two parts. Part of the encoded bit stream output by the encoder is fed into a buffer. For a constant bitrate channel, the data in the buffer is fetched at a constant rate, and if the buffer is large enough, the bitrate variation caused by the MPEG picture type, etc. can be smoothed out. This is necessary for both constant bit rate transmission and variable bit rate transmission in general. However, in practice, the buffer size is always limited. The buffering process will bring a delay to the system, and this delay is proportional to the size of the buffer. Latency is often a serious issue for real-time image communication, so buffers should be kept as small as possible. That is, long-term fluctuations in bitrate due to changes in scene content or changes, etc. they cannot be softened in this way, so another part is needed. This is to send some measure of the output bitrate to the encoder to control the encoding process, thus changing the output bitrate [3] .

Quality (bit rate)

In multimedia technology, quality is often used to judge the effect of audio, and quality here is actually bitrate.

1. Introduction
2 sound control
3 encoding mode
Introductionedit transmission
The term quality is widely used.
In multimedia technology, quality is often used to judge the effect of audio, and quality here is actually bitrate.
On WINDOWS it is called “bit rate” and on some players it is described as ” bit rate “.
Quality refers to the bit rate at which digital sound is converted from analog to digital format. The higher the bitrate, the better the quality of the restored sound.
sound control edit stream
16 Kbps = phone quality
24 Kbps = increase phone quality, shortwave transmission, longwave transmission, European standard medium wave transmission
40 Kbps = American standard medium wave transmission
56Kbps=Voice
64 Kbps = boost voice (best bitrate setting for cell phone ringtones, best setting for cell phone mono MP3 players)
112 Kbps = FM stereo broadcast FM 128 Kbps = tape (best setting for mobile phone stereo MP3 player, best setting for low-end MP3 player)
160 Kbps = HIFI high fidelity (best setting for mid to high end MP3 players)
192Kbps=CD (best setting for high-end MP3 players)
256Kbps=Studio Music Studio (for music enthusiasts)
In fact, with the advancement of technology, the quality of music is also getting higher and higher, the highest quality of MP3 is 320Kbps, but some formats can achieve higher sound quality.
For example, the emerging APE audio format can provide real audiophile level lossless sound quality and smaller volume than WAV format, and its quality is usually 550kbps-950kbps.
encoding modeedit stream
VBR (Variable Bitrate) Dynamic Bitrate means there is no fixed bitrate. The compression software immediately determines which bitrate to use based on the audio data being compressed. This is a method that takes quality as a premise and takes file size into account The recommended encoding mode;
ABR Average Bit Rate (Average Bit Rate) is an interpolation parameter of VBR. LAME created this encoding mode in response to the low file volume ratio of CBR and the variable size of files generated by VBR. Within the specified file size, ABR takes every 50 frames (about 1 second for 30 frames) as a segment. High-frequency and insensitive frequencies use relatively low traffic, and low-frequency and large dynamic performance use high traffic, which can be used as VBR and CBR, a compromise option.
CBR (constant bitrate), constant bitrate means the file has one bitrate from start to finish. Compared to VBR and ABR, the compressed file size is very large and the sound quality will not improve significantly compared to VBR and ABR.