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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.

The higher the bitrate, the higher the sound quality and the larger the file size.

but the quality of the source file determines the final quality.

From highest to lowest, the sound quality will be worse, but from lowest to highest, the sound quality will remain unchanged at most, but the file will be larger.Many

General mp3 are good with bit rate around 128, and also 3-4 BM in size.

The bitrate, choosing it, directly affects the size of your mp3 file and the listening experience. High compression ratio has high distortion, and low compression ratio has low distortion, but how do we find a balance point that we can accept on both counts? This requires careful exploration in the experiment. Considering that the sound quality of low bitrate files is not suitable for playing music, the minimum set is 128kbps, and four fixed bitrate files of 128, 192, 256 and 320 are used for comparison. and try.

The compression ratio of 128 kbps is still relatively rough, and the high-frequency part is highly distorted after compression. It sounds hollow, wrinkled, rough, and there are often flickering sounds. Misunderstanding, the compressed volume of a 3 minute 39 piece of music is 3414 Kb. Although the volume is not large, the sound is not satisfactory, and there are relatively large defects.

192kbps bit rate compression effect is much better than 128. First of all, the sound is solid, at least there is no empty feeling, the high-frequency distortion is also much less, the sound is compact, the noise is small and clean, and achieve relatively ideal listening The sound effect, just because the compression is still relatively strong, the detail performance is still not very good, the texture of musical instruments, especially instruments of wind, it is still very hard, unreal and lacks musicality. The compressed size is 5123kb, and I think the compression ratio is 128~ It is better to use it in a mp3 player with a capacity of ~256m, which can not only satisfy the basic sense of hearing, but also is suitable in size.128m can store about 95 minutes of music, and 256m can double to 190 minutes of music.

The 256 kbps compression rate is naturally a step higher than 192 in terms of sound quality. Take the first 10 seconds of the track, the low frequency of the cello is obviously less grainy, and the sound is more smooth and natural, with texture and texture. It is also clearer, with much more detail, the rendering of the atmosphere is more prominent, the rotation of parts in the following songs is also more expressive, the clarity of large and small signals is also improved, and the sound is more detailed and lasting. But at the same time, the file size has also increased to 6831kb, which is still affordable for a 256m mp3 player. It is not difficult to know by calculation. According to the bit rate of 256, about 135 minutes of music can be stored. Generally speaking, it is enough, 128m is a bit less and can only support a little over an hour, so it is recommended to use 192 bitrate for 128m.

320 kbps is the maximum bitrate that lame can provide. The final file generated is 8592kb which is about 8.4M. Compared to the 37M of the wav file the compression ratio is basically 4.5:1 but the generated mp3 file sounds very distorted Now on Compared with other 320 bit rate, the natural advantage is obvious, the tone, details, etc. are very delicate, basically achieve the sound quality of the original CD copy, especially in the CD player with playback function from mp3, the basic No difference, but I use relatively high-end earplugs with high resolution, plus my experience and skill with music and equipment, I can still hear a lot of differences compared to wav files, first Instead, the compressed mp3 sounds a bit The crunch feeling is relatively dry on the whole. Without the wav file, it sounds fresh and dynamic. In terms of final details, nuances and sense of space, the separation is not as high as the quality of the wav file, but it is quite close in terms of timbre, but the performance is poor and the digital flavor is relatively strong. So if you are using a miniature hard drive player like an iPod, I recommend you use 320kbps compression ratio, which can get the best listening experience. Of course listening to wav directly is the best~

The bit rate directly affects the sound quality.

High bitrate is good and low bitrate is bad.

The code rate is the number of data bits transmitted per unit of time during data transmission. Generally, the unit we use is kbps, that is, kilobits per second.

The popular understanding is the sampling rate. The higher the sampling rate per unit time, the higher the precision, and the processed file is closer to the original file, but the file size is proportional to the sampling rate, so almost all encoding formats pay attention. It’s about how to use the lowest code rate to achieve the least distortion. The cbr (fixed code rate) and vbr (variable code rate) derived from this core are all articles in this regard, but things are not absolute, in terms of audio, the higher the bit rate, the lower the compressed ratio, the smaller the sound quality loss and the closer it is to the sound quality of the audio source.
The information in the computer is represented by binary 0 and 1, and each 0 or 1 is called a bit, which is represented by lowercase b, that is, bit (bit); uppercase B represents byte, ie byte, one byte = Eight bits, ie 1B=8b; the capital K in front stands for thousand, that is, thousand bits (Kb) or kilobytes (KB). Indicates the size of the file, usually using bytes (KB) to indicate the size of the file.

Kbps: The first thing to understand is that ps refers to /s, which is every second. Kbps refers to the speed of the network, that is, how many thousands of bits of information are transmitted per second (K means thousands of bits, Kb means how many thousands of bits), it is expressed in kb (kilobit), and in the case KBps means how many kilobytes are transferred per second. 1KBps = 8Kbps. The Internet speed of ADSL is 512 Kbps. If converted to bytes, it is 512/8 = 64 KBps (that is, 64 kilobytes per second).

A frame is a still image, and continuous frames form an animation, like a television image.
We normally say the number of frames. Simply put, it is the number of image frames transmitted in 1 second. It can also be understood that the graphics processor can update several times per second, usually expressed in fps (Frames Per Second). Each frame is a still image, and showing frames in rapid succession creates the illusion of movement. Higher frame rates result in smoother, more realistic animations. The more frames per second (fps), the smoother the motion is displayed.

What is the bitrate of the music?
It can also be called bit rate, which is nothing more than the amount of data reproduced per second by a type of music, the unit is expressed in bits, that is, binary bits. bps is the bit rate. b is bit, s is second, p is per, and one byte is equal to 8 binary bits. That is, the file size of a 4-minute song at 128bps is calculated as (128/8)*4*60=3840kB=3.8MB, which means that the same song with the same bit rate (bps) will not no matter what format (such as mp3 wma) The capacity is basically the same, which can only represent a transmission rate, not the sound quality. Due to different compression engines, the sound quality of different formats varies a lot. However, for the same format, the higher the bitrate, the larger the file and the better the sound quality.

What is the sample rate of the music?
Sampling rate refers to the number of samples per unit of time. The sampling rate is 44KHz, which means the number of samples per second is 44K, which means that 44,000 pieces of data are used to describe the sound waveform in 1 second. That is, the higher the sample rate, the better the sound quality. But he and bitrate are two completely different concepts.