MP3 File Structure Analysis Part 2


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MP3 File Structure Analysis Part 2

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Sounds in nature are very complex and waveforms are extremely complex.

Mp3

Usually we use pulse code modulation coding, that is, PCM coding. PCM converts continuously changing analog signals into digital codes through three steps of sampling, quantizing, and encoding.

u Decode:

Reverse encoding process

1.1.2 Brief introduction of 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 a .mp3 extension with a higher compression ratio, essentially maintaining the sound quality of the source file. 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 low quality”. Three audio codec schemes, MPEG Layer1, Layer2 and Layer3, have been formed, respectively, corresponding to the three sound files MP1, MP2 and MP3

MPEG (Moving Picture Experts Group) is a group of moving picture experts under ISO. The MPEG standard it specifies is widely used in various multimedia. The MPEG standard includes video and audio standards. Audio standards have developed MPEG-1, MPEG -2, MPEG-2 ACC, MPEG-4. The MPEG-1 and MPEG-2 standards use the same family of Layer1, 2, 3 audio codecs, and most MP3s use the MPEG1 standard.

MP3 audio compression consists of two parts: encoding and decoding. Encoding is the process of converting the original signal to a level signal, and decoding is the reverse process. MP3 uses the PerceptualAudio Coding distortion algorithm. The frequency range of sound perceived by the human ear is 20 Hz to 20 kHz. MP3 cuts out a lot of redundant signals and irrelevant signals. The encoder transforms the original sound into the frequency domain through a mixed filter bank and uses a psychoacoustic model. to estimate that it may be only The perceived noise level is quantized and converted to Huffman coding to form an MP3 bit stream. The decoder is much simpler, its task is to extract the sound signal from the encoded spectral line components through inverse quantization and inverse transformation.

MP3 file data consists of multiple frames, and a frame is the smallest unit of an MP3 file. Each frame, in turn, consists of a frame header, additional information, and sound data. The playback time of each frame is 0.026 seconds and its duration varies with the bit rate. Some MP3 files have extra bytes at the end that contain description information for non-audio data.


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MP3 file structure analysis

MP3 file structure analysis

MP3 FORMAT

ü ID3:

mp3 format

 

Usually located in several bytes at the beginning or end of an mp3 file, it records the singer, title, album name, era, style, and other mp3 file information.

ID3 is divided into two versions, the V1 ID3 version is fixed at the end of the 128-word file section, it begins with the TAG character, if there is no ID3V1 information, it is considered that there is no ID3V1 information, the V2 ID3 version is found. at the beginning of mp3 and the length is variable.

ü Sampling rate:

The number of samples extracted from a continuous signal to form a discrete signal per second. It is expressed in Hertz (Hz). Sampling rate refers to the sampling frequency when converting an analog signal to a digital signal, i.e. how many points are sampled per unit of time. The higher the sample rate, the more realistic and natural the sound will be. On today’s major capture cards, the sample rate is generally divided into three levels: 22.05 KHz, 44.1 KHz, and 48 KHz. 22.05 KHz can only achieve the sound quality of FM radio, and 44.1 KHz is the theoretical limit of CD sound quality, and 48 KHz is more accurate.

ü 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 more information transmitted. In the audio and video fields, bit rate often translates to bit rate. The bit rate indicates how many bits per second the encoded (compressed) audio and video data should represent, and a bit is the smallest unit in binary. 0 or 1. The relationship between bitrate and audio and video compression is simply that the higher the bitrate, the better the quality of the audio and video, but the larger the encoded file; if the bitrate is lower, the situation is just the opposite.

Bit rate = sample rate * number of samples * number of channels

ü Bitrate/Stream/Bitrate:

It refers to the data stream used by audio and video files in a unit of time. The popular understanding is the sample rate, which is the most important part of quality control in audio and video encoding. Generally, the units we use are Kb/s and Mb/s. . Generally speaking, the higher the code stream, the lower the compression ratio and the higher the quality. The higher the code stream, the higher the sampling rate per unit time, the higher the data stream, the higher the accuracy, and the closer the processed file is to the original file.

ü Code:

From the point of view of information theory, the data that describes the source of information is the sum of the redundancy of information and data, namely: data = information + data redundancy. The audio signal has correlation in the time domain and the frequency domain, that is, there is data redundancy. Taking audio as the source, the essence of audio encoding is to reduce redundancy in the audio.

Audio Formats: Everything musicians should know to choose the right file

What is the best audio format? It is a very frequent question. Surely you’ve already raised it.

The answer is simple. It all depends on your needs. Whether you’re sending demos, building your digital music distribution, or archiving your songs, the file format is very important.

So, to help you choose the best file format for your music, we have collected all the essential information about the audio formats.

And even more important, which one is better in each situation.

Compression: the first impression

Audio formats depend on compression.

I don’t mean the compression you apply to a song in your DAW software. I am talking about file compression.

Compression makes a file smaller, to save space when streaming, downloading or storing.

But what happens when you compress?

There are 3 types of file compression:

Uncompressed (I know that “uncompressed” is not a type of compression, but I add it to make everything clearer), without loss and loss.

Uncompressed and lossless files retain the original data intact. But files with loss delete certain data from the original file to reduce the file size.

So the more compressed a file is lost, the more information is lost.

Compressing with loss does not mean that all your drums are going to be deleted. It simply removes the audio that the human ear is not able to hear. Maybe only dogs notice the difference …?

In any case, if you really want to hear what disappears when you compress a file, watch this MP3 conversion experiment.

About compression types
Here is a simple way to understand each type of compression:

An uncompressed file is an exact copy of the original. No information is lost. Uncompressed files are like an original picture.
Lossless files are slightly smaller files, but they keep the original information intact. A lossless file is like an original painting, but it is folded in two until you look directly at it.
Files with loss are the most compressed. Some of the original information is lost during compression. Files with loss are smaller versions than the original — the photo is still there, but some details have disappeared.


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Now that you know what compression is, you may be wondering how each type of file is compressed.

Do not worry. Here we go.

How each type of file is compressed

Uncompressed Formats

Uncompressed formats are not compressed (obviously). The most common uncompressed formats are WAV and AIFF.

These are the formats that you usually export from your DAW. If you duplicate a song to WAV, it is an exact and uncompressed copy of the original.

Lossless Formats
LANDR: A space to create. More details
Lossless files are compressed. But although they are compressed, they retain all the original information as a WAV. They simply unfold at the time of opening.

The most common lossless format is FLAC. Apple also has its own lossless format, called ALAC, used in iTunes.

The FLAC format makes the files lighter than WAVs, but they retain all the original information. Although the size of these files is usually very large.

Formats with loss

Lossy files are the most common audio format. The most used is the MP3. But there are other types, such as OGG, WMA and AAC.

The drawback of files with lossy compression is that it deletes some data from the original file.

But the benefit is that they are smaller, open faster and take up less space.

Files with loss can be high and low resolution, depending on the amount of compression. The higher the quality, the less information will be lost.

The truth about bitrates

The quality of an audio file is determined by its bitrate (bit rate).

The bitrate corresponds to the information processed per second. And that is what 320 or 192 means in MP3 files.

Thus, an MP3 with a bitrate of 320 has 320 kilobits per second — or kbps.

WAV and AIFF usually have 1411kbps.

A higher bitrate means more information per second. And more information per second means better sound. Simple, right? Now you understand the basic points of compression, file types and bitrate, right?

Perfect. Let’s continue.

Now comes the million dollar question …

 

In what situation do I use each format?
If I talked about each of the audio formats, we would be here for days. Surely you have other responsibilities, and a lot of music to produce. So I will be brief and concise. These are the best uses for each of these formats. We talk about WAV, MP3 320 and MP · 192.

WAV
The WAV is at the top of the podium. It is the Ferrari of audio formats. The WAV offers a cleaner and sharper sound than the other compressed formats. If you share demos with a record label, show your work for a possible audiovisual project or send your music to a blog, you need a mastered WAV.

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The WAV is a guarantee that your best sound represents you.

When you master your music, always use the WAV as the delivery format.

WAVs can also be converted to other formats later, so it is the right format for conversion later.

The only drawback of WAVs is the large size of the files. They take up a lot of hard disk space. So your computer, your phone, your iPod or your Dropbox will fill up very quickly if you only use WAV.

But when it comes to your own music, it is important to always have a WAV copy of each of your tracks.

Most platforms require WAV to upload your music for distribution. For example: iTunes and Amazon ask for high-quality WAV to upload music to their services.

The 320 MP3
The MP3 of 320 is the most frequent type of file. For one simple reason: It has the best of both worlds.

They are compressed, so they are easy to handle in regards to their size. But they also offer a pleasant and rich sound.

If you listen to music in streaming, it is very likely that it is 320. For example: everything you hear in high quality on Spotify is at 320kbps.

The MP3 of 320 is a good way to share your best sound saving space on the hard drive and avoiding long waits during the upload and upload.

MP3 192
The 192kbps MP3 is the draft horse. They are fast and dirty MP3, for when you have to share something easily and quickly. They are useful when transferring a handful of files at once, check your entire catalog or share and reference tracks quickly.

A lower bitrate causes more degradation than an MP3 of 320 with loss, but sometimes it is difficult to feel the difference. Take the test and judge for yourself.

The MP3s of 192 are the perfect tool for musicians who need an efficient and fast way to share or listen to their music in streaming.

Useful tip: if you use your own streaming player on your web page, an MP3 of 192 will make your page load faster.

Don’t forget any format by the way
Each format has its uses. Choosing the right format depends on each context.

So think about what sound you share and where you do it. Are you using the right format?

Mastering in WAV format is the best bet to share your music. Once you have the mastered WAV, you can convert it in any other format into a periquete.

Formats are important in the era of streaming. So make a smart choice and use the right format.