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mp3 audio format, the most popular

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mp3 audio format, the most popular

With the rapid development of file compression technology, MP3 has become the most popular music format today.

High-quality music quickly spreads to all parts of the world with the arrangement of 0 and 1, shaking people’s hearts. What is MP3? The full name of MP3 is MPEG Audio Layer 3. It is an efficient computer audio coding scheme. It converts audio files into smaller files with .MP3 extension with a higher compression ratio and basically maintains the sound quality of the file. original. 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 and small quantity”, and now has formed MPEG Layer 1, Layer 2. Layer 3 three audio encoding and decoding schemes. The compression rate of MPEG Layer 3 can reach from 1:10 to 1:12. A 1M MP3 file can be played for 1 minute, while a 1 minute CD-quality WAV file (44100Hz, 16bit, 2ch, 60sec) occupies 10M of space, so Calculated, the time The playback time of a 650M MP3 disc should be more than 10 hours, while the playback time of a CD with the same capacity is about 70 minutes. The advantages of MP3 are unmatched by CD. 2 Analysis of the principle of MP3 2.1 MPEG audio standard MPEG (Moving 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, among which MPEG-1, MPEG-2, MPEG-2 AAC, 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 kits to reduce data traffic , and another feature is the multi-channel expansion kit, which increases the number of main channels to five. Fraunhofer IIS and AT&T released the MPEG-2 AAC (MPEG-2 Advanced Audio Coding) standard in 1997 to significantly reduce data traffic. The MDCT (Modified Discrete Cosine Transform) algorithm adopted by MPEG-2 AAC, The sampling frequency can be between 8 KHz and 96 KHz, and the number of channels can be between 1 and 48. MPEG Audio Layer 1, 2 and 3 use the same filter bank, bitstream structure, and header information, and the sample rate is either 32 KHz, 44.1 KHz, or 48 KHz. Layer 1 is designed for DCC (digital compact cassette) digital compression tape, the data rate is 384 kbps, and layer 2 has made a compromise between complexity and performance, and the data rate has been reduced to 256 kbps- 192kbps. Layer 3 was designed for low data rate from the beginning, and the data rate is 128Kbps-112Kbps. Layer 3 adds MDCT transform, which makes its frequency resolution 18 times higher than that of Layer 2. Layer 3 also uses information averaging similar to MPEG video entropy coding to reduce redundant information. The vast majority of MP3 uses the MPEG-1 standard. 2.2 The purpose of 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. Let’s look at an example: You want to sample a song you like that is about 4 minutes long, store it on a disc, and sample it in CD-quality WAV format at a sample rate of 44.1 kHz, which means receiving 44100 per second. , stereo, each sample data is 16 bits (2 bytes), so the space occupied by this song is: 44100×2 channels x2 bytes x60 seconds x4 minutes=40.4MB If you download this song from the Internet, assume the transmission speed is of 56kbps, the download time is: 40.4x106x8/56x103x60=96 minutes. Even a 1M broadband network takes more than 5 minutes. It can be seen that audio compression is especially important to reduce the storage space of audio data. 2.3 MP3 encoding and decoding MP3 audio compression involves encoding and decoding in two parts. Encoding is turning the data in a WAV file into a highly compressed bitstream, and decoding is taking the bitstream and reconstructing it into a WAV file. MP3 uses a distortion algorithm called Perceptual Audio Coding. The frequency range of sound perceived by the human ear is from 20 Hz to 20 kHz. MP3 cuts out a lot of redundant and irrelevant signals.

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MP3 encoder

1. MP3 Encoder FAQ

: what is an MP3 encoder?
An MP3 encoder is a piece of software that uses the MP3 codec algorithm (compression/decompression) to create mp3 files. Most encoders only convert
a WAV file to an MP3 file, although many can convert other formats such as WMA, Real Audio, Ogg, etc.

There are only a few standalone encoders, and a lot of software also only uses 4 main encoding engines, largely due to
to Fraunhofer Gesellschaft patents and various companies helping with ISO sources. Although no company owns the license, the
Developers must pay expensive license fees no matter what proprietary MP3 encoder they use. Major MP3 encoding engines include: LAME (
non-ISO source), BladeEnc, Fraunhofer, and Real Networks’ Xing encoder.

– How does the MP3 encoder work?
The core technology under MPEG-Layer 3 is included in the MP3 encoder. The decoding process uses a series of algorithms and rules to compress audio.
The encoder also detect sounds that occur at the same time
and they try to rule out any that might be “masked” or “inaudible” by other sounds.

– What is a good MP3 encoder?
Xing is the fastest encoder in terms of speed, but the worst in quality. For smaller file sizes, Fraunhofer FastEnc
offers the best quality. LAME is a very good encoder, and one version is faster than the previous one, BladeEnc
it is the best quality for large files, but very slow.

2. Dissection of MP3 files
In addition to proficiency in using the basic features of the MP3 encoder, ordinary users do not need to know how the internal structure of the MP3 file is encoded, just like the situation when
face JPEG or DOC files. Out of morbid curiosity, here’s an X-ray view of an MP3 file:

– Box header
As mentioned above, MP3 files are made up of thousands of “frame frames”, each frame containing a part (second part) of valuable audio data.
for the decoder to reconstruct the audio data. The first part above is the box header. (Frame Header), which consists of 32-bit metadata related to the
later data, see the figure below. The MP3 header begins with an 11-bit “sync timing” block, which allows the player to seek and lock the first
legal framework available, which is useful in MP3 streaming, which can quickly move or jump ID3 from the playback source block to a normal one.
position . However, simply detecting synchronized blocks is theoretically not enough, so it is necessary to check the header.

– transmission lock
MP3 was originally designed for broadcast, and as a result it became important that the MP3 receiver could be synchronized with the signal at any part of the broadcast,
so the frame header is placed at the beginning of any frame transmission, so when an MP3 receiver “tunes” to a data stream, it picks up the
signal instantly and you can play it immediately. Interestingly, this fact makes it possible to cut MPEG files into small segments, each of which can be played independently. But unfortunately
not possible in 3-layer (MP3) files, where frames often depend on other frames, so you can’t just
Edit .

– Frames per second
Just as the movie industry has a standard for the number of frames per second in film to ensure proper viewing on any projector,
A similar standard is used in the MP3 standard, regardless of the file’s bitrate, MPEG-1 A frame in the file is 26 ms, approximately 38 fps frames per second. If the bit rate
is , the frame size is correspondingly larger, and vice versa. Also, the number of samples contained in an MP3 frame is constant, 1152 samples per frame.

The total size of any given frame can be calculated with the following formula:

FrameSize = 144 * BitRate / (SampleRate + Padding).

MP3 file format analysis

MP3 encoding

MP3 file format analysis

Signal Description:

(1) MP3 encoding input signal: PCM (Pulse Code Modulation) sound signal, some audio files in .wav format are PCM signals.

( 2 ) MP3 encoded output signal: transmission in MP3 format

WAV format file capacity = (sampling frequency X quantization number of bits X channel) X time / 8 (byte = 8 bits). When the 2 bytes of 14H~15H have a value of 1 , it indicates the PCM encoding format of the data bit , which can be used as the input of the MP3 encoder .

3. Analysis of the SHINE program
SHINE is a MP coding program written in C language, consisting of 11 source files in total. Add the source file to the newly created VC purchase mode console application to run, but it must use command line mode when running.

1. File data structure
A config_t structure type is defined in types.h , and a global variable configuration is initialized with it, which is equivalent to an ” object ” in an object-oriented language, and is used to encode data and parameters throughout the process. coding Save and manage

Define a wave_t structure type to store PCM pulse format file information, and use wave_t to define the wave variable in config_t , which stores the MP3 encoded source information as the input to the MP3 encoder.

A type of structure mpeg_t is defined which is used to store MP3 encoded information, and the mpeg variable is defined in config_t with mpeg_t , and the information stored in this variable is output as MP3 encoded parameter information.

typedestruct {

time_tstart_time;

char*infile;

round_wave;

char* output file;

mpeg_tmpeg;

} config_t;

The above structure is mainly used to store the ” header ” information , and the byte stream entity information after encoding the output is stored in the bs structure ( defined in the bitstream.h file ), and the bs structure is defined as

static structure

MP3 file format analysis

1. Summary of MP3 and MPEG files

An MP3 file is made up of frames, and a frame is the smallest unit of an MP3 file. The full name of MP3 must be MPEG1 Layer-3 audio file.

MPEG (MovingPictureExperts Group), MPGE audio layer refers to the sound part of the MPGE file, which is divided into three layers based on the quality and complexity of the encoding, namely Layer-1, Layer2 and Layer3, corresponding to MP1, MP2 and MP3 format files.

2. Structure of MP3 files
MP3 files are divided into 3 parts : TAG_V2(ID3V2 ) , Frame, TAG_V1(ID3V1) .

( 1 ) Frame format

The frame header is 4 bytes and its structure is as follows

typedef FrameHeader

{

unsigned intsync: 11; // synchronization information

unsigned intversion: 2; // version

unsigned intlayer: 2; // layer

unsigned intprotection: 1; // CRC check

unsigned intbitrate: 4; // Bit rate

unsigned intfrequency: 2; // sample rate

unsigned intpadding: 1; // adjust frame length

unsigned intprivate: 1; // reserved word

unsigned intmode: 2; // channel mode

unsigned int mode extension: 2; // extended mode

unsigned intcopyright: 1; // Copyright

unsigned original: 1; // original logo

unsigned inemphasis: 2; // emphasis mode

}

HEADER, *LPHEADER;

Each frame takes 26 ms to play, regardless of the length of the frame. The length of MAIN_DATA is

Length(MAIN_DATA) = ((version == MPEG1)?144:72) * bitrate/frequency + padding;

( 2 ) ID3V1 format

ID3V1 is stored at the end of the MP3 file, a total of 128 Bytes, all information is stored sequentially and the insufficient part is filled with ‘\0’, which can be opened and viewed with UltraEdit.

typedef tagID3V1 structure

{

char header[3];

char Title[30];

artist char[30];

album char[30];

char Year[4];

char Comment[28];

coal reserve;

character track;;

charGenus;

}

ID3V1,*pID3V1;

( 3 ) ID3V2 format

ID3V2 is stored in the header of the MP3 file and consists of a tag header and several tag frames.

The tag header is 10 bytes,

char header[3];

char see;

character review;

char Flag;

character size [4];

Each tag frame consists of a 10-byte frame header and at least one byte of variable-length content. The frame header is defined as follows:

char frame ID[4];

character size [4];

char Flags[2];

How does an mp3 file work?

How does an mp3 file work

What is the structure of an mp3?

How does an mp3 file work

An MP3 song has three
versions: 96 Kbps (96 kilobits per second), 128 Kbps and 192 Kbps. Kbps (bit rate), which indicates the amount of music data per second,
the higher the Kbps value, the better the sound quality, and the larger the file, the MP3 standard stipulates that an MP3 file with a constant bit rate is called CBR, and most of the
MP3 files are CBR, and MP3 file with changing bit rate is called VBR, and the length of each FRAME can be changed. The following are
the differences between CBR and VBR:
1) CBR: The size of the FRAME with a fixed bitrate is fixed (the formula is as above), as long as the total length of the file and the length of the frame are known, mp3 can be calculated from the 26ms needed to play each frame. The total playback time can also be monitored by counting the number of frames to control operations such as fast forward, fast rewind, and slow playback. Note: Sometimes not all frames are the same length and some frames may be one or more bytes longer.
2) VBR: VBR is an algorithm released by XING company, so there will be “XING” keyword in the MP3 FRAME (many popular
Small software can also perform VBR compression (it is not known if they comply with this agreement), it is stored in the first valid FRAME in the MP3 file and identifies that the MP3 file is VBR. At the same time, the first FRAME stores the total number of FRAMES of the MP3 file, which makes it easy to get the total playing time, and at the same time, there are 100 bytes to store the FRAME INDEX of 100 times segments of the total playing time. . Suppose a 4 minute MP3 song 240S is divided into 100 segments, and the time difference between two adjacent INDEXes is 2.4S, so through this INDEX as long as some FRAMES are processed before and then we can quickly find the FRAME header we need to fast forward. Table 2 Explanation of the byte of
structure of the first frame of the VBR 1-4 file The same standard sound frame header as CBR 5-40 Save the VBR file logo “Xing” (58 69 6E 67), the specific location of this logo depends on the adopted standard MPEG and the sound depends on the channel mode. The leading and trailing bytes of the flag are not used. 36-39 MPEG-1 and non-mono (common) 21-24 MPEG-1 and mono 21-24 MPEG-2 and non-mono 13-16 MPEG-2 and mono 41-44 Flags, indicates whether the frame number, Se stores information about file length, directory table, and VBR scale, and if so, 01 02 04 08. 45-48 frame number (including first frame) 49-52 file length 53-152 file table directory, used for byte positioning according to time. 153-156 VBR scale for bit rate changes

How does an mp3 file work?

What is the structure of an mp3?

MP3 file is composed of frame (frame), frame is the smallest composition unit of MP3 file. MP3 full name should be MPEG1 Layer 3 audio files. MPEG
(Motion Picture Experts Group) translates into Chinese as Moving Picture Experts Group, and refers specifically to moving video and audio compression standards.
MPEG1 standard, also known as MPEG audio layer, which is divided into three layers based on compression quality and encoding complexity, namely,
Layer-1, Layer2 and Layer3, which correspond to the three sound files of MP1, MP2 and MP3 respectively, and use different
levels of audio files according to different purposes. The higher the MPEG audio encoding level, the more complex the encoder and the higher the compression ratio. The compression ratios of MP1 and MP2 are 4:1 and
6:1-8:1 respectively, while the compression ratio of MP3 is as high as 10:1-8:1. 12:1, meaning one minute of CD-quality music requires 10MB
of storage space without compression, but only about 1 MB after MP3 compression encoding. However, MP3 uses a lossy compression method for audio signals.

Low sound distortion, MP3 adopts “sensory coding technology”, that is, when encoding, the audio file is first analyzed for frequency spectrum, and then the noise level is filtered by a filter, and then each bit remaining is sparse and arranged by quantization, and finally form an MP3 file with a higher compression ratio, and make the file
compressed achieve a sound effect closer to the original sound source when played back.

2. The complete structure of the file
MP3 The MP3 file is roughly divided into three parts: TAG_V2 (ID3V2), Frame, TAG_V1 (ID3V1) ID3V2
contains
information like author, composer, album, etc. The length is not fixed, which expands the information volume of ID3V1.
frame
… Frame A series
of frames, the number is determined by the file size and the length of the frame
. The length of each FRAME may not be fixed, or it may be fixed. It is determined by the bit rate. Each FRAME is divided into two parts: frame header and data entity. The frame header records information such as bit rate, sample rate, and mp3 version, and each frame is independent of each other. ID3V1 contains information like author, composer, album, etc. ., and the length is 128BYTE. 3. MP3 Frame Format 1. Frame Header Format The frame header is 4 bytes long. For fixed bitrate MP3 files, the frame header format of all frames is the same. The data structure is as follows: typedef FrameHeader { unsigned int sync: 11; // unsigned synchronization information int version: 2; // version

mp3 audio normalizer

An mp3 audio normalizer is a tool to avoid differences in volume level in 2 or more audio files.

audio volume normalizer

There are various ways to do this, but the most common is to use software that can normalize the levels of the files automatically. This can be done with an audio editor, or with a dedicated mp3 normalizer program.

audio volume normalizer

Normalizing an audio file is checking each frame and adjusting it to be sure that each one is optimized to get the best possible volume without distortion.

An audio file can be normalized in two ways:

1. Peak Normalization:

This is where the audio file is analyzed and the highest peak is identified. The volume of the entire file is then increased or decreased so that this peak is at the maximum possible level without distortion.

2. RMS Normalization:

This is where the audio file is analyzed and the average volume level is identified. The volume of the entire file is then increased or decreased so that this average level is at the maximum possible level without distortion.

This behavior is similar to the compressor that keep the volume range of noisy and lower passages of the audio to keep both in a range that can sound fine.

The main difference is that the goal of an audio normalizer is not to change the sound of the audio, but to make sure that the volume levels of all the files are the same.

The most common format that is normalized is MP3, but other formats such as WAV can also be normalized.

In this times even video files can be normalized too.,

There are many reasons why you might want to normalize your audio files.

For example, if you have a number of files that were recorded at different levels, you may want to normalize them so that they all have the same volume level.

This can be useful if you want to create a playlist of files that are all the same volume, or if you want to make sure that all the files in a particular folder have the same volume level.

Another reason to normalize your audio files is if you want to make sure that they sound their best when played back on different devices.

For example, if you have an MP3 file that sounds great on your computer, but sounds terrible on your phone, you may want to normalize it so that it sounds its best on both devices.

Normalizing your audio files can also be useful if you want to make sure that they are suitable for use in different situations.

For example, if you have an MP3 file that you want to use as background music for a video, you may want to normalize it so that it is not too loud or too quiet.

Normalizing your audio files can also be useful if you want to make sure that they sound their best when played back at different speeds.

For example, if you have an MP3 file that sounds great when played back at normal speed, but sounds terrible when played back at double speed, you may want to normalize it so that it sounds its best at both speeds.

Normalizing your audio files can also be useful if you want to make sure that they sound their best when played back at different volumes.

For example, if you have an MP3 file that sounds great when played back at a low volume, but sounds terrible when played back at a high volume, you may want to normalize it so that it sounds its best at both volumes.

Normalizing your audio files can also be useful if you want to make sure that they are suitable for use in different environments.

For example, if you have an MP3 file that you want to use as background music for a party, you may want to normalize it so that it is not too loud or too quiet.

Normalizing your audio files can also be useful if you want to make sure that they are suitable for use in different places.

For example, if you have an MP3 file that you want to use as background music for a restaurant, you may want to normalize it so that it is not too loud or too quiet.

Normalizing your audio files can also be useful if you want to make sure that they are suitable for use at different times of the day.

For example, if you have an MP3 file that you want to use as background music for a morning show, you may want to normalize it so that it is.

What is a loudness normalizer or an audio volume normalizer?

A volume normalizer is used to make sure that audio files play at the best possible volume without clipping and also that all audio files play at a similar volume.

A volume normalizer analyzes an audio file and then adjusts it to sound at a specific volume level. This is often done with audio files that are uploaded to file sharing sites, so that all users can listen to the audio at a similar volume.

It is quite common to find volume differences in the files that are downloaded from the internet, since these have been created from a wav but different sampling and bit frequencies have been used to create them.

A volume normalizer can analyze an audio file and then apply gain or attenuation to adjust the volume of the file to a specified level. This is useful if you want to ensure that all audio files played on your website or in your application sound at a similar volume.

Because it’s frustrating to have a collection of audio or even video files and find that when you play them they play at different volumes.

For this reason, it is necessary to use a volume normalizer, with which you can make sure that all audio and video files are heard at a similar volume.

It is important to ask yourself if the bit rate is important for the quality of an audio or video file, the same for the sample rate.

The answer is no, not always.

Bit rate and sample rate refer to the amount of information that can be stored in an audio or video file.

The higher the bit rate, the higher the quality of the audio or video file.

However, sometimes a low-quality video or audio file can sound better than a high-quality file.

This is because the bit rate and sample rate are not always indicative of the quality of the audio or video file.

There are many factors that can affect the quality of an audio or video file, such as the encoder used to encode the file, the quality of the microphone used to record the file, the quality of the equipment used to play the file, etc.

In summary, the bit rate and the sample rate are not always indicative of the quality of the audio or video file.

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.

source
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.

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.