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What is sound source compression?

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What is sound source compression?

digital sound wave

In the old days, you used to put a CD-sized Walkman and a CD case with your favorite CDs in a bag and go out.

Digital sound waves

Today, you can carry a large number of songs in the palm of your hand and enjoy music anytime, anywhere. The evolution of the times is amazing. Why is it possible to do this? It’s all thanks to “audio compression”!

Why compress in the first place? Doesn’t the sound quality get worse?
Music data can generally be burned to CD (700MB) for only 80 minutes without compression, but when compressed, the data size is reduced to about 1/10, which is 10 times larger in a place with the same capacity. capable of recording music. In other words, up to 10 music CDs can be burned onto one CD. Also, you can attach emails and download them from the Internet thanks to “audio compression”!

The sound quality is annoying, but it is not 100% the same, but it can maintain the same sound quality as an audio CD.

“Audio compression” has become an indispensable part of our musical life. There are various audio compression methods like MP3 and AAC, but I think there are many people who say, “I really don’t understand the difference …”.

First of all, from the basics of music files.
Music files are roughly classified into three types: “uncompressed”, “lossy compression”, and “lossless compression”. The nature of each format is as follows.

■ Uncompressed The
Sound quality is good because it is not compressed, but the file size is large.
<File format: WAVE (WAV) / AIFF>

■ Lossy compression When
compresses with lossy compression, it cannot be undone and the sound quality deteriorates, but the file size can be reduced.
<File format: MP3 / WMA / Ogg Vorbis / AAC / AC3 / ATRAC3 / ATRAC3plus / RealAudio / TwinVQ>

■ Lossless compression
Even if it is compressed once, it can be restored to the original data, so there is no deterioration in sound quality. The file size will be a bit larger, but recently the capacity of the PC hard drive has also increased, which is why it is popular.
<File format: Flac / Monkey’s Audio / TTA / WMA Lossless / RealAudio Lossless / Apple Lossless>

What are the MP3s that you see often on the Internet?
It uses a lossy compression method that achieves a high compression rate and is an audio compression method that compresses the amount of data to approximately 1/11 (128 kbps) while maintaining the sound quality of a music CD. In other words, the voice is compressed cutting the data in the part that is difficult for humans to perceive. The original data is compressed with the idea that “this part cannot be heard accurately by the human ear, so cut it out!”
What is the AAC you hear often these days?
AAC is a relatively new audio compression method. ITunes uses it in familiar places. AAC has a compression efficiency 1.4 times higher than MP3 and the sound quality is almost the same. The name itself may not be heard as much as MP3, but it is actually used in many places. For example, QuickTime, BS digital / 110 degree, CS digital / digital terrestrial / single segment broadcast, “Chaku-Uta” mobile phone, and so on.

Various audio compression methods
In addition to MP3 and AAC, there are several methods of audio compression.

■ Uncompressed
WAVE
(WAV)
.wav WAV or WAVE (RIFF waveform audio format). A format for writing audio data developed by Microsoft and IBM. A file format used primarily by Windows.
AIFF .aif A file format for audio data developed by Apple Computer, the standard audio format for Mac. Mac version of WAVE.

■ Lossy compression
MP3 .mp3 The compression format that has been overwhelmingly popular until now. It uses a lossy compression method that achieves a high compression rate by reducing data that is difficult for humans to perceive. Downloading music from the Internet, etc.
WMA .wma Abbreviation for Windows Media Audio. It has a copyright protection function and is used for broadcast distribution. Like other major compression methods, it uses a lossy compression method that achieves a high compression rate by reducing data that is difficult for humans to perceive.
Ogg Vorbis .ogg Popular as a next generation format for MP3. It is a lossy audio compression standard similar to MP3, WMA, AAC, etc., but differs from existing standards in that it is a free and open standard. One of the main features is that there are fewer rights restrictions than other codecs and formats.

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Digital sound and analog sound

Analog and Digital Sound

The “sound” can generally be expressed as a “waveform”.

Analog and Digital Sound

Analog recording is used to store this “waveform” as it is. Since the original waveform is stored as is, the sound quality appears to be good, but the analog data deteriorates in the playback process and the noise is mixed. However, since digital data does not deteriorate much in the playback process, it can be reproduced closer to the original sound without noise.

Also, if you copy an analog recording, the original waveform will gradually collapse and the sound quality will deteriorate. On the other hand, digital recording is recorded by dividing this waveform into small pieces and quantizing them. This value does not change even if you copy it, so the sound quality will not deteriorate. It can also be easily imported to a personal computer for processing.
Digital sound quality can be expressed in “bits” and “kHz”. “Bit” is the pitch of the sound volume. Also, “kHz” is the number of times a second is divided into digital data, and the higher the value, the higher the sound quality.
The must-have PCM recorder for musicians

“Linear PCM Recorder”.
It’s no exaggeration to say that no jazz musician doesn’t. When I go to a jam session or a live concert, I see all the members take out their PCM recorder and set it up.
I think this was MD a while ago, but why did musicians switch to PCM recorders? The first reason is that it is possible to record “good sound”. Let’s take a look at the history of music recorders and high-quality sound.
The Sony “Walkman” is the first most portable recording device.
It was initially used as a play-only cassette player, but after a few years it was equipped with a recording function.
After that, the era of MD arrived, and soon MD recorders were developed, and the clear digital sound quality, albeit compressed, gave a new step to recording. And DAT (Digital Audio Tape) is also a major player in the history of recording. DAT is a tape medium that enables digital recording. Immediately, a recorder using DAT was launched and its high-quality sound was well received. However, due to a copyright issue, DAT disappears.
After that, in the recording market, IC recorders focused on language study, lesson recording and conference proceedings appeared, but music recorders that sought high sound quality were never made.

However, with the recent band boom and the increasing music population, the need for music recorders has increased. In response to that, the PCM recorder appeared and attracted a lot of attention.
The biggest difference between a PCM recorder and a conventional recorder. It is in compression. Compression is the process of reducing capacity while preserving data content. In the past, most audio media was compressed and recorded. Sound quality deteriorated significantly due to compression, but there was a reason I kept using it.
This is because many semiconductor memories required for long-term recording were still expensive at the time.
However, in recent years, the price of semiconductor memory has fallen to the point that it can be kept at a reasonable price even if memory capacity is increased, leading to the birth of uncompressed linear PCM. By installing a high-capacity semiconductor, it was possible to record with good sound without compression.
24 bit / 96 kHz beyond CD
Some linear PCMs have a sound quality that exceeds that of a CD. That is the “24bit / 96kHz” format.
Of course, general CDs are not compressed. The format is “16 bit / 44 kHz”. What this means is that 16 bits for 44.1 kHz sampling, that is, 1 second is distinguished by 44,000 units, and from small to loud sounds in a unit of time it is a 16-bit gradation, that is, 56,000. It means that it is expressed in gradation. That is a CD.
On the other hand, there are several types of formats for linear PCM, but basically, the higher the sampling frequency and the higher the number of bits, the higher the sound quality and the better the sound can be recorded. It means that 96,000 times they can analyze sound more precisely than 44,000 times, and the analysis allows for more delicate expression of sound in 24-bit than 16-bit. Today, recorders with such amazing functionality have been marketed.

Techniques for enjoying compressed audio with high sound quality

MP3

I converted a music CD to MP3 or WMA, but the sound is not good … Many users will think so. So, let me introduce you to one of the methods to do high quality audio compression.

Sound quality is improved by lowering the volume of the extracted WAV file and then encoding it in MP3 or WMA.

There are various audio compression methods like WMA, AAC, Ogg Vorbis for MP3. They are very convenient because they can compress the file size to about 1/11 (at 128 kbps) compared to WAV. However, these are called lossy compression and sound quality degrades slightly because compression is done by reducing sound information to the point that it is difficult for the human ear to understand. If you really stick to the original sound of a music CD, you should use WAV as is, or use lossless (lossless) compression like WMA Lossless, Apple Lossless, FLAC.

However, even with MP3 and WMA, if it is compressed well, many people will not notice the difference in sound quality. Generally, a bit rate of 128 kbps is used, but if you increase it to 160 kbps or 192 kbps, the amount of data to be reduced will decrease and the sound quality will improve.

If you can’t tell the difference from the original sound by listening to the lossy compressed sound, that’s fine, but it is troublesome that there are cases where you can clearly see the deterioration depending on the song. Specifically, there is a clip-like beep, and the cymbal and hi-hat sound is clearly distorted. This is not just a problem to be solved by increasing the bit rate, it has nothing to do with the lack of high frequencies. This happens because the music data matches the weak points of the compression algorithm.

The weak point is that it is vulnerable to the sound applied by the compressor (equalizing the maximum volume). In other words, it is vulnerable to loud and continuous sounds. In particular, the recent J-Pop is made by applying a compressor to gain sound pressure and setting it to maximum volume, so it can be said that it is the music data that is vulnerable to compressed audio.

Therefore, if you process the WAV data a little before compressing it to MP3 or WMA, the sound quality will be drastically improved and clipping phenomenon can be avoided.

The process is simply to lower the volume level of the entire song. Of course, it’s burning, so you can’t mess with the volume of CD playback or squeeze the volume from encoding software like Windows Media Player. It is necessary to rewrite the waveform of the WAV data using waveform editing software and lower the volume level. Specifically, it is safe to lower the level by about -6dB. This will cut the volume in half, but due to the features of the compression software, it doesn’t appear to be as quiet. If possible, it is even more effective to apply an effect called an expander to eliminate the compressor effect.

Improve quality of mp3s Part 3

MP3

Be aware of the difference between bit rate and bit depth!
Bit rate: amount of information per second
Bit depth: amount of information per sample divided by the sample rate
In other words, the calculation is “bit rate = bit depth x sample rate”.
What values can be set when exporting MP3? What is the best export configuration?

MP3

~ What items can be set when exporting MP3? ~

When exporting MP3, you can set the following two items 💡

Bit rate: 16 kbit ~ 320 kbit
Sampling frequency: 32,000 Hz, 44,100 Hz, 48,000 Hz
For example, if there are no specifications in a competition and you want to export with good sound quality with mp3, let’s export with “320kbit, 48,000Hz” 💡

What is the best setting to reduce capacity and export with good sound quality?
The capacity is small!
Sound quality is good!
So if you want the capacity to be as small as possible but also the sound quality as best as possible, which setting is better to export?

The sample rate is generally 44,100 Hz.
Use ~ 44,100Hz! ~

In the video industry, 48,000Hz is mainstream, but in music, the sound quality is high enough if it is 44,100Hz, which is used for CD.

In the blind test, it is said that about half of the people can distinguish between them and in addition, there are two options, so even if you answer properly, there is a chance that you will win about half.

The bitrate is around 128 kbit, which is the limit between good and bad.
~ 128 kbit or more is recommended! ~

Even if the bit rate is reduced to around 128 kbit, the print does not change much and the roughness does not appear.

If you lower it to reduce capacity, “about 128 kbit is a guideline” 💡

If you want to reduce the capacity and stick to the sound quality, it is better to select 128 kbit ~ 192 kbit and 44,100 hz to export.
~ 44,100Hz 128kbit ~ is the best!

For those who want to reduce capacity and focus on sound quality, it is better to set the sample rate to 44,100hz and the bit rate to 128-192kbit 💡

MP3s also have export settings. What are the settings for exporting with even slightly better sound quality? summary of
MP3 stands for “MPEG 1 Layer 3” compression method
Depending on the compression settings, the capacity may be reduced by 1/10 or more.
MP3 removes sound components that are inaudible to humans, thus maintaining sound quality.
Set “bit rate” and “sample rate” when exporting
The sample rate is “number of samples per second”
Bit rate is “amount of information per second”
Bit depth is “amount of information per sample”
The best sound quality settings for MP3 export are 48,000hz, 320kbit
44,100Hz, 128 ~ 196kbit is the setting that balances capacity and sound quality when exporting MP3.
MP3s are often used to check demos and save space 💡

Even if you have exported it casually, put it in the corner of your head that there is a setting for MP3 export, and when you need it, remember it and use it

Improve quality of mp3s Part 2

MP3

In other words, MP3 can be said to be a sound source file that was originally created by removing only the sounds that are difficult for humans to hear and without modifying the other sounds.

It is a commercially useful file format because it sounds very nice even if the capacity is small 💡

About MP3 Export Settings

When exporting MP3, the sound quality changes greatly depending on the bit rate (amount of data per second) and the sample rate (number of samples per second).

The sample rate determines how many divisions per second
~ How many divisions per second? ~

The sample rate determines how many divisions of information are handled per second.

At 44,100Hz, which is common, one second of information is divided into 44,100 samples, and at 44,800Hz, it is divided into 48,000 samples 💡

If you divide the sample into smaller samples, the information will be seamlessly connected, and if the information is approximate, it will be staggered.

Bit rate (kbit) determines the amount of information per second
~ Amount of information per second ~

On the other hand, the bit rate (bit) determines “the amount of information a sample has in one second”.

soon,

Bit rate = bit depth (1 sample information) x sample rate (number of samples per second)

For example, the bit rate of a 16-bit 44,100 hz wav file is 705.6 kbit.

Note that the bit rate is sometimes called “bps (bits per second)” because it is the number of bits per second 💡

caution!

* Bit depth is used in WAV export settings, etc. which is very similar, but the bit depth is the amount of information per sample. Be careful because it is confusing!

Improve quality of mp3s

MP3 quality

MP3 is often used as a compressed sound source when the capacity of uncompressed files like WAV and AIFF is large and inconvenient, but in fact, you know that depending on the settings like bit rate and sample rate, it can “write with fairly high sound quality”. mosquito?

mp3 quality

This time about such MP3

Setting to export with high sound quality
Settings to export with good sound quality while saving space
Misleading Bitrate and Bit Depth Differences
I will also present on such things ^ – ^ No

1. What is MP3?
1.1 How do MP3s reduce their capacity?
2. About the MP3 export settings
2.1. The sample rate determines how many divisions per second
2.2. Bit rate (kbit) determines the amount of information per second
2.3 What values can be set when exporting MP3? What is the best export configuration?
3. What is the best setting to reduce capacity and export with good sound quality?
3.1. The sample rate is generally 44,100 Hz.
3.2 The bit rate is around 128 kbit, which is the limit between good and bad.
3.3. If you want to reduce the capacity and stick to the sound quality, it is better to select 128kbit ~ 192kbit and 44,100hz to export.
4. MP3 also has export settings. What are the settings for exporting with even slightly better sound quality? summary of

What is MP3?
What is MP3?
MP3 is a compression technology (file format) that can reduce the capacity by 1/10 or more compared to the uncompressed compression method called “MPEG 1 Layer 3”.

It is also used when you want to save data capacity or when you collect a large number of songs in contests.

How do MP3s reduce their capacity?
~ How do MP3s reduce their capacity? ~

There are three reasons why MP3s can compress the capacity of a sound source 💡

Eliminate data in the “ultra high range (16 kHz or higher)” that is not as audible to humans at all frequencies.
It removes the data of small sounds that cannot be heard because they are erased by loud sounds for each frequency.
Humans cannot hear the small sound that plays immediately after the loud sound, so they cut it off.

Compress mp3 without losing quality

Compress mp3 without losing quality

Mp3

On lossless music compression, theory, practice, conclusions.
With this material, I want to open a series of articles with everything related to listening to music on a computer. The time has come to share experiences and summarize disparate articles on the Internet in one, although they are not intended to be precise, but relatively brief. In the first part, we will see the audio formats. What is FLAC, WavPack, TAK, Monkey’s Audio, OptimFROG, ALAC, WMA, Shorten, LA, TTA, LPAC, MPEG-4 ALS, MPEG-4 SLS, Real Lossless? Do you know how many types of audio files are registered today? So far, we are dealing with lossless compression formats for audio materials, and the answer to the question about the number of audio extensions is at the end of the article. Happy reading!

mp3

So first, let’s define the terms:

“An algorithm is a precise prescription that defines the computational process that goes from variable inputs to the desired result.”

“Codec (codec in English, of encoder / decoder – encoder / decoder – encoder / decoder or compressor / decompressor) is a device or program capable of converting data or signals. Codecs can encode a stream / signal (often for transmission, storage, or encryption) or decode, to view or change into a more suitable format for these operations. Codecs are often used in digital video and audio processing. Most codecs for audio and visual data use lossy compression to obtain an acceptable final (compressed) file size. There are also lossless codecs ”.

“Lossless data compress. – method of data compression, using encoded information that can be restored in one bit. This fully recovers the original data from the compressed state. As a rule, each type of digital information has its own lossless compression algorithms “.

Lossless data compression is used when the identity of the compressed data with the original is important. Common examples are executables, documents, and source code. Programs that use lossless compression formats are called archivers, everyone knows the most popular ZIP or RAR file formats, the Unix Gzip utility, etc. All these programs differ in the applied algorithms (one or more) and therefore in different compression properties of different files.

Part I. – THEORY:

Compression methods or lossless compression algorithms can be classified according to the type of data for which they were created. There are three main types of data: text, images, and sound. Basically any multipurpose lossless data compression algorithm (multipurpose means it can handle any type of binary data) can be used for any type of data, but most of them are inefficient for all basic types. Audio data, for example, cannot be compressed well with a text compression algorithm and vice versa.

Compression methods include the following: entropy compression, dictionary methods, statistical methods. Each method is good for a specific type of data and includes several algorithms.

Entropy compression: Huffman algorithm Adaptive Huffman algorithm Arithmetic coding (interval Shannon-Fano algorithm) Golomb codes Universal Delta code (Elias Fibonacci)

Dictionary methods: RLE Deflate LZ (LZ77 / LZ78 LZSS LZW LZWL LZO LZMA LZX LZRW LZJB LZT)

Statistical algorithm models for text (or textual binary data as executable) include: Burrows-Wheeler transform (block sort preprocessing that makes compression more efficient) LZ77 and LZ78 (used by DEFLATE) LZW.

Is it possible to improve the quality of an MP3?

Thanks to MP3 we can listen to our favorite music everywhere. When you put your MP3s on a USB stick, you can listen to your favorite music in the car, for example. But you can also put MP3 music on your smartphone. Allowing you to listen to music whenever possible.

MP3 quality

But sadly, it still happens that the quality of an MP3 is not really what it should be. In this article, we look at the options to solve that problem. So that you can not only listen to music everywhere, but also enjoy it everywhere.

Mp3 quality

What exactly is an MP3 music file?

Mp3 is a method of compressing digitally stored music. Uncompressed storage of a stereo digital music file takes up a lot of disk space. An average of 10MB of disk space per minute of recorded music.

However, compressing a music file and saving it as MP3 will leave only one-tenth the size of the original file.

Since the introduction of the CD, music has been recorded digitally in the form of samples or measurements. Sound is neither more nor less than vibrating air. These vibrations are also known as sound waves. Sound waves can be measured, recorded, and stored.

However, when sound waves are produced creatively, then it is music.

The number of vibrations per second determines the pitch of the sound. A large amount of vibrations creates a high tone, a small amount of vibrations for a low tone.

The number of vibrations per second is expressed in hertz. The human ear can perceive sounds between 20 Hz and 20,000 Hz.

It was once scientifically discovered that in order to record the highest pitch, a measurement must be taken 44,100 times per second. Therefore, the number 44,100 is the sample rate in hertz that is required for good quality recording.

In addition to high and low tones, a piece of music also contains high and soft passages. The difference between the loudest and the softest passages is called the dynamic range. For the dynamic range of a piece of music to be recorded digitally, you can choose 256 steps (8-bit) between the softest part and the hardest part or 65536 steps (16-bit).

The dynamic range is highest when recording with 16-bit samples or 65536 steps.

If we then do some math with this data, we see that 44,100 measurements are needed for one second of music. Each measurement (sample) is 16 bits (2 bytes) in size. That means 1 second of music takes up 88,200 bytes or 88Kb of disk space.

But since we like to listen to music in stereo, we can multiply that number by 2. For example, one second of music in stereo takes up 176 Kb of disk space and therefore 10 MB per minute.

When a compressed MP3 file is created from an original music file, this is done using a lossy compression method.

Lossy compression causes data loss. With an MP3 file, this means that information is omitted from the file that is beyond the reach of the human ear.

Humans are most sensitive to sounds between 2 kHz and 4 kHz. And we cannot hear loud and soft sounds simultaneously. Therefore, it is only necessary to keep the loud sound. In technical terms, this is called psychoacoustic masking.

What determines the quality of an MP3?
The MP3 format was developed by the German research institute Fraunhofer ISS. In addition to utilizing the limitations of human hearing just mentioned, the format consists of several mathematical formulas. This makes it possible to reduce the original file by a factor of 3 to 12.

The degree of compression is related to the bit rate. Bit rate is the amount of data that is processed per unit of time. This means, among other things, that the more data there is in one second of music, the larger the MP3 file will be. But also the better the sound quality of the MP3.

A bit rate of 64 to 96 kbps is enough to talk. A bit rate of 128 kbps is used for a good quality music file. Excellent quality can be achieved with a bit rate of 192 kbps or higher, with a maximum bit rate of 320 kbps.

A bit rate of 192 kbps or higher is only useful if the recording quality of the track is also excellent.

Obviously if you want the mp3 to sound even better, use Mp4Gain to mormalize mel volume, to correct the equalization and to make a series of changes or improvements.

MP3 quality – too compressed for hi-fi sound?

Audio quality

When it comes to the subject of “MP3 and sound quality”, one is entering a minefield. Hi-fi fundamentalists claim that many people no longer know what good sound really is because of MP3s. The accusation is not entirely unfounded, because MP3 is a lossy format. However, you shouldn’t make it too easy for him with judgment. After all, there is no uniform standard for MP3 quality. Another important question is: what about the sound quality of other formats?

audio quality

What “lossy” means for the sound quality of an MP3 file

MP3 and other lossy audio formats such as AAC may have been lost. to. designed with the aim of saving storage space. Because at the time of its development, the storage capacity of hard drives was much more limited than it is today, and the download and upload rates were also insufficient for large amounts of data. Today, the bandwidth for streaming and wireless transmission over Bluetooth are limiting factors. So compression still has to be. How is the amount of data reduced compared to the original recording?

On the one hand through compression and on the other hand through the omission of certain sound information. Because not everything that is captured in a recording also becomes the compressed file. To limit the effects of data loss on MP3 quality, only information that is acoustically insignificant is ignored. To be more precise: particularly low frequencies and particularly high tones are cut off. Because people can only perceive extreme highs and lows up to a certain point or not at all.

That’s how high MP3 quality really is

A general evaluation of the quality of MP3 sound is complicated by the fact that there are different levels of quality. They are the result of the respective bit rate (data rate, “bit rate”), specified in kilobits per second (“Kbit / s”). 64 Kbit / s as well as 128, 192, 256 or 320 Kbit / s can be implemented. The following applies: The higher the value, the less data loss will be compared to the source material.

A rule that is mentioned from time to time states that from a bit rate of 192 kbit / s data loss is no longer important for auditory impression. The file format alone says little about the quality of the audio signal.

But there is no clear limit. Factors like music genre, system, and last but not least individual hearing all play an important role when it comes to evaluating the quality of an MP3 file. There are also differences between the audio formats: a file encoded in AAC at 192 kbps tends to provide a better listening experience than an Ogg Vorbis file with the same data rate.

What is the sound quality on Spotify and other music streaming services?

Some 20 years after its invention, MP3 is still the most widely used audio format on the Internet. However, there are other formats that play an important role in music playback today. An example of this is the patent-free Ogg Vorbis format mentioned above. The streaming giant Spotify also relies on this.

Other audio formats used by streaming services are:

Apple Music: AAC
Spotify: Ogg Vorbis
Google Play Music: MP3
Deezer HiFi: FLAC

Streaming providers are quite reluctant to provide information on the respective data rates. When the service launched, Apple Music announced that the streams would be streamed at a bit rate of 256 kbps. With Spotify it is 320 Kbit / s with high sound quality, also with Google Play Music. At lower quality levels, the bit rate drops below 200 Kbit / s. However, providers of lossless transmission clearly exceed these values: Deezer, for example, announces its high fidelity subscription with 1,411 kbit / s. The stream here is in lossless FLAC format.

What exactly is an MP3 music file?

Mp3 is a method of compressing digitally stored music. Uncompressed storage of a stereo digital music file takes up a lot of disk space. An average of 10 MB of disk space per minute of recorded music.

However, if you compress a music file and save it as MP3, only a tenth of the original file size remains.

mp3 quality

Since the introduction of the CD, music has been digitally recorded in the form of samples or measurements. Sound is no more or less than vibrating air. These vibrations are also known as sound waves. Sound waves can be measured, recorded and stored.

However, when creative sound waves are produced, there is music.

The number of vibrations per second determines the pitch of the sound. A large amount of vibration produces a high tone, a small amount of vibration produces a low tone.

The number of vibrations per second is expressed in Hertz. Human hearing can perceive sounds between 20 Hz and 20,000 Hz.

Once it has been scientifically established that to capture the highest tone, a measurement must be taken 44,100 times per second. Therefore, the number 44,100 is the sampling frequency expressed in hertz needed for a good quality recording.

In addition to the high and low tones, a piece of music also contains hard and smooth passages. The difference between the highest and smoothest passage is called the dynamic range. For dynamic range on a digitally recordable track, you can choose 256 steps (8 bits) between the softest and loudest part, or 65536 (16 bits).

The dynamic range is highest when recording with 16-bit samples or 65536 steps.

If we then add a calculation to this data, we see that it takes 44,100 measurements for a second of music. Each measurement (sample) is 16 bits (2 bytes) in size. That means that 1 second of music takes up 88,200 bytes or 88 KB of disk space.

But since we like to listen to music in stereo, we can multiply that number by 2. For example, a second of music in stereo already takes up 176 Kb of disk space and, as said, 10 MB per minute.

When a compressed MP3 file is made from an original music file, it is done with a lossy compression method.

Data is lost on lossy compression. With an MP3 file, this means that the information is outside the file that is beyond human hearing range.

For example, people are more sensitive to sounds between 2 kHz and 4 kHz. And we can’t hear loud, soft sounds at the same time. Therefore, only loud sound needs to be preserved. In technical terms, this is called psychoacoustic masking.

What determines the quality of an MP3?

The MP3 format was developed by the German research institute Fraunhofer ISS. In addition to taking advantage of the limitations of human hearing just mentioned, the format consists of a series of mathematical formulas. This allows you to reduce the original file by a factor of 3 to 12.

The amount of compression is related to the bit rate. Bit rate is the amount of data processed per unit time. This means, among other things, that the more data there is in a second, the larger the MP3 file will be. But also the sound quality of the mp3 will be better.

For speech, a bit rate of 64 to 96 kbps is sufficient. A bit rate of 128 kbps is used for a good quality music file. Excellent quality can be achieved with a bit rate of 192 kbps or higher, with a maximum bit rate of 320 kbps.

A bit rate of 192 kbps or higher is useful only if the recording quality of the track is also excellent.

Check the quality of an MP3

Unfortunately, the quality of MP3 music is not always good. This applies, for example, when an MP3 comes from a somewhat unknown source. But of course you can also make an MP3 from a recording that is not very good in itself.

Generally, if you create MP3s from music on your own CDs or other sound media, you can guarantee the quality of the MP3s simply by choosing the appropriate settings in the software you are using.

However, if you get MP3 music in other ways like downloading from the internet for free, it will be a slightly different story.

Then you have to settle for what you get. With the knowledge of this article, it is already much easier to distinguish a low quality MP3 from a good quality MP3.

Something that can be useful. Because there is not much good to do of poor quality.

In summary, we can say that a good MP3 meets the following requirements:

-The MP3 file must have a bit rate of 128 kbps.
A higher bit rate is only desirable for excellent recordings.
-The recording quality must be good.
-The recording quality can be checked by listening to each MP3 before buying and / or downloading it. Preferably with headphones. This gives you the best impression of sound quality.

The MP3s that you buy online, for example at the Apple Store, are usually of good quality. Usually, it is the MP3 files you download from other sources that you should carefully check and listen before using them.

Music you download from sources other than online stores will definitely end up in the Downloads folder.

You can check the MP3 music downloaded from the Internet as follows:

Launch File Explorer and navigate to the Downloads folder.
To display only MP3 files in File Explorer, type: * .mp3 in the search box. This search will show you all the files in the Downloads folder with the extension .mp3.
Right-click on the MP3 file you want to check and click Properties in the context menu that opens.
The [Music file name] property window is then displayed. The Details tab shows the exact bit rate of the MP3.

When you close the Properties window and double-click the selected MP3 file, the corresponding MP3 file will be loaded into your PC’s MP3 player and played.
That’s basically all you can do. A bad MP3 is impossible to improve on. Converting music to an MP3 file not only compresses but also removes data from the music file that you have been able to read.

And the lower the bit rate, the more data is generally lost and impossible to recover.

This means that when you have downloaded a low quality MP3 file, you have no choice but to search for a better quality MP3. The same goes for an MP3 whose recording quality is not very good.

Collecting the best possible MP3 files takes some effort. But this effort will be amply rewarded once you start listening to your favorite music, and the sound quality will certainly contribute to the actual enjoyment of the music.