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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.

Why are MP3 bitrates often multiples of 32? (power of 2) part 2

Depending on the resource, VBR can be encoded by changing the bitrate between a fixed rate above each frame, or by sharing the available bits in adjacent frames (effectively producing a non-standard bitrate for the two frames combined).

the fixed frame depends on the sampling rate, 1152 samples per frame. There is no limit to the size of the frame itself, nor to the base 2 size of the frame (ie 417 bytes for a 128 kbit/s MP3 sampled at 44.1 kHz).

In the end, a file encoded at 126kbps will sound worse than a file encoded at 128kbps, and similarly a file encoded at 131kbps will sound better. However, MP3s are encoded according to the compression psychoacoustic model of a specific encoder. The amount by which a file sounds “better” or “worse” at a given bitrate largely depends on the algorithm used to implement the model, but in general higher bitrates allow for more data, presumably for rebuild a more accurate original transmission. audio signal

Why are MP3 bitrates often multiples of 32? (power of 2)

Some people say:

I understand why multiples of 2 often show up on computers since they are binary, but I can’t figure out how the most common mp3 bitrates (64kbps, 128kbps, 160kbps, 192kbps, 256kbps, 320kbps, etc.) also tend to follow this rule.

Since MP3 is just a sequential encoding of sound waves, why is it important to represent each second in kilobits divisible by 2?
Does a music player like iTunes continue to read the file and play the encoded sound regardless of the second limit, or does it read the file every second?
In the latter case, reading a 256kbps file requires reading slightly fewer memory pages than a 257kbps file, but the player can always read 256KB chunks, regardless of their bitrate, and just process them automatically. incremental, right, Bar?
Are 128kbps MP3 songs popular simply because it’s a generally accepted bitrate, or do they really have any advantages over 126kbps and 131kbps files, apart from a very slight difference in quality/file size?

For constant bit rate (CBR) encoding, the MPEG-1 Audio Layer III standard specifies standard bit rates of 32, 40, 48, 56, 64, 80, 96, 112, 128, 160, 192, 224, 256 and 320 kbit/second. There are a few others defined in the MPEG-2 standard, but they are also multiples of 2 (actually all multiples of 8 in the range 8 to 160 – see the table called “Bitrate Index” in the link above) .

Technically, there is nothing that limits the MP3 bitrate to a multiple of 2, since variable bitrate encoding can be used, or a custom bitrate can be achieved using some flags not used in the MPEG specification ( although this must be implemented manually). . In order for MP3 to be MPEG-compliant, and therefore compatible with most MP3 decoders, it must have a bitrate defined by the specification, so all CBR-encoded MP3 files have a bitrate of two.

Audio bit rate, bit rate

Bit rate refers to the number of bits (bits) transmitted per second. The unit is bps (Bit per second) The higher the bit rate, the faster the data transmission speed.

Bitrate in sound refers to the amount of binary data per unit of time after converting an analog sound signal to a digital sound signal, which is an indirect measure of audio quality.

Bitrate refers to the sampling rate at which digital sound is converted from analog to digital format. The higher the sampling rate, the better the quality of the restored sound. As a benchmark for the efficiency of digital music compression, bit rate indicates the rate of the number of bits bps (bit per second, bits per second) transmitted per unit of time (1 second). Kbps (in layman’s terms is 1000 bits per second) is usually used as the unit. The bit rate of digital music on CD is 1411.2 kbps (that is, to burn 1 second of CD music, 1411.2 × 1024 data bits are required), the high BIT RATE of the digital music file music means that it should be processed in a unit of time (1 second) The amount of data (BIT) is large, which means that the sound quality of the music file is good. However, when the BITRATE is high, the file size increases, which will take up a lot of memory capacity. The most commonly used bitrate for music files is 128 kbps, and MP3 files can generally use 8 to 320 kbps. In the same way, most of them are 32-256 Kbps. Of course, the wider the rate, the better, but 320 Kbps is the highest level at the moment.

Bitrate calculation formula
The basic algorithm is: [Bit rate] (kbps)=[file size] (bytes) X8/[time] (seconds)/1000

Special algorithm for audio files: [bit rate] (kbps) = [quantization sample point] (kHz) × [bit depth] (bit/sampling point) × [number of channels] (typically 2)

For example, the D5 drive has a capacity of 4.3G, which takes into account different audio formats, so it is calculated as 600M (so the remaining capacity is 4.3*1000-600=3700M), so the video file should not be larger than 3.7G, in this example, take The capacity of the video file is 3.446G, and the length of the video is 100 minutes (6000 seconds). The calculation result: the bit rate is approximately equal to 4933kbps.

What is a good bitrate guide for mp3 files? Part 2

To produce high-quality MP3 files of classical and jazz music, the optimal bitrate depends on the characteristics of the song.

Smooth jazz can usually be copied at 192kbps to create a good balance between file size and diminishing returns, although 256kbps may sound better in a home entertainment center. A classical orchestra should be 256kbps for a portable player, but if you want to burn a CD at home or in your car, a 320kbps file might be a better option.

For saturated music such as hard rock, metal, arena, pop, electronic and house music, 320 kbps will provide the best results. The higher the number of bits per second, the more complex acoustic envelope will be preserved.

If possible, it’s best to create MP3 files with variable bit rates. This allows the encoding program to determine if a particular frame of music requires the full bit rate. Otherwise, the program will reduce data retention for that frame, resulting in a smaller file without sacrificing quality. Forcing the program to “oversample” frames can produce artifacts.

While this article is intended as a general guide, he or she may be equally satisfied with a lower bitrate for a particular song or songs in general. Many factors affect our ability to judge the quality of music, not only the devices we use but also our activities while listening to it. For example, for those who listen to MP3 files while exercising or taking a walk, external noise can make it more difficult to tell the difference in quality. Conversely, audiophiles may prefer to sample at 320kbps, regardless of their equipment, type of music, or listening habits.

If you create your own MP3 files, there are other settings that affect quality. LAME is an excellent MP3 encoder that is free and has many graphical interfaces as the interface for this popular command line program. LAME allows users to adjust many settings to generate high-quality MP3 files in seconds. You can also experiment with various bitrates in your source file to find the best subjective balance between quality and file size.

What is a good bitrate guide for mp3 files?

(a good bitrate guideline for mp3 files?)

MP3 files are compressed audio files created from audio formats such as wave (.wav). Wave files replicate analog recordings and digital sound files at the expense of large file size, while MP3 files sacrifice some quality for a smaller footprint. There are several factors that mitigate the quality sacrifice during the conversion process. With the correct bitrate and settings, MP3 files can provide very high quality results, making them very close to the original wave files when played on portable audio players.

An mp3 player.

The balance between file size and quality is somewhat subjective. For audiophiles, any difference is noticeable. Others may simply not be able to tell the difference between a high quality MP3 file and a raw wave source. In many cases, the nuances of the sound environment will only become clearer when played through a high-quality stereo system.

MP3s are compressed digital music files that sacrifice quality for file size.
MP3 files are primarily targeted at portable audio players. In this field, high-quality MP3 files are played with incredible sound due to their small file size. With the limited memory of portable players, it makes sense that one would want MP3 files to be as small as possible while maintaining the highest possible quality.

For this, one of the most important factors when creating MP3 files is the bit rate. In general, the more bits per second that are preserved from the original file, the higher the quality of the MP3 and the larger the file size. Lower bit rates reduce size and quality. The idea is to use the bitrate for maximum realism without saving unnecessary data, which just creates larger files with no noticeable difference to the ear.

For voice recordings such as lectures or language lessons saved to waveforms, a bit rate of 32 kilobits per second (kbps) is acceptable, although 64 kbps may offer better quality, depending on the source. At 32 kbps, the sound may sound “flat”, but that’s understandable. A 64 kbps MP3 file created from a voice recording should sound nearly identical to the original.

Desaturated acoustic music with simple arrangements should work fine at 192kbps bitrate. You can choose 256 kbps if the music will be played on a high quality device. Music that falls into this category includes folk, boy band songs, easy listening, and folk music. There are also works by many classic artists such as James Taylor, Linda Longstadt, Jonny Mitchell, and Simon Garfunkel.

Audio Bitrate Calculation

Generally, the files we read will know the following parameters:

Sampling frequency: the number of sampling points that pass one second; by
For example, the most common sample rate of 44.1k means that there are 44,100 sampled data points in one second;

Bit depth: Bit depth is also called sampling bit depth, and the bit depth of the audio determines the dynamic range.
Our common 16-bit (16-bit) can record a dynamic range of about 96 decibels. Well, roughly you can know that each bit can record about 6 decibels of sound. Similarly, the 20-bit recordable dynamic range is about 120 dB; 24 bits is about 144 dB.

If we define 0dB as the maximum value, then the amplitude of the sound is calculated by extending down, then the dynamic range of CD audio is “-96dB ~ 0dB”. And so on, the dynamic range of 24-bit HD-Audio is “-144dB~0dB”. It can be seen that at higher bit depths, a greater dynamic range is available and lower levels of detail can be recorded.

Number of channels: usually single channel or two channels

Through the above parameters, the bit rate of the audio can be calculated

\color{blue}{code rate = sample rate (44.1k) * bit depth (16) * number of channels (2) = 1411.2kbps}

Audio bitrate, also known as bitrate: Refers to the amount of data that can pass through an audio stream per second. For example, 128 kbps, where ps (per second) is per second and kb is thousand bits, then 128 kbps means that the amount of data that can be transmitted in one second is 128 thousand bits. For files of the same format, the higher the bitrate, the better the sound quality. But for audio files of different formats, the same bitrate does not mean the same sound quality.

CD quality sound, usually two channels, compressed at 128kbps MP3, one second of data is converted to 128kbits. Some details are lost, which saves a lot of disk space or bandwidth.

After knowing the audio code rate, you can get the size of the whole audio file = duration (300s) * code rate (1411.2)/1024/8=51.67M

3 Ways to Change Bitrate of MP3 Files Part 2

The second thing you want to read is that you should never convert to a lower bitrate. bitrate to a higher bitrate stream and I hope it sounds better. You cannot gain quality by increasing the bit rate. This is absolutely correct. If you try to convert the bitrate, you will actually lower the quality of the MP3 file.

If you want an MP3 with a higher bitrate than it currently has, you’ll need to go back to the source (CD, etc.) and extract the audio at full quality. You can then convert that file to a higher bitrate MP3 file.

The last thing you need to know is that converting between lossy formats is not recommended as you will still lose quality. However, it is possible to convert between formats losslessly while preserving quality.

On the other hand, many tests done even with recording engineers or professional musicians have shown that less than 1% of the population can distinguish between an mp3 with good bitrate (196 for example) and a samplerate of 44100 or more.

There is a sector of puritanism that defends tooth and nail the use of losless formats, because in theory they seem to be better, but the limitations and peculiarities of human hatred mean that a loosy file with a high bitrate and a high samplerate is enough for the human ear.

Mp4Gain is the most professional and polished program to help you in all these matters and you can achieve the highest sound quality with any format.

3 Ways to Change the Bitrate of MP3 Files

Do you want to change the bitrate of an MP3 file?

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

There is a drop in quality, but the difference is negligible for most listeners using standard speakers or headphones. If you’re an audiophile, you’ll probably never use the MP3 format, except for expensive audio equipment.

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

The MP3 format is a lossy format, which means sacrificing audio quality to keep the file size relatively small. Almost every site will tell you not to convert lossless audio files to MP3 unless you might lose some audio quality.

<Almost all the time. The only time it might make sense is if you have a high bitrate audio file in a low quality format like WAV. For example, it might make sense to convert a 96kbps WAV file to MP3, but only if you choose a bitrate of 192kbps or higher. A higher bitrate in an MP3 file will allow it to maintain the same quality as a WAV file, even if it has a lower bitrate.