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Audio compression, how it works

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Audio compression, how it works

audio compression

It must have a corresponding inverse transform, called decompression or decoding. The audio signal can introduce a lot of noise and some distortion after passing through a codec system

Audio compression technology refers to the application of suitable digital signal processing technology to the original digital audio signal stream (PCM encoding), without losing the amount of useful information, or under the condition that the loss introduced insignificant, reducing (compressing) its code rate, and also called compression encoding. It must have a corresponding inverse transform, called decompression or decoding. Audio signals can introduce a great deal of noise and some distortion after passing through a codec system. The advantages of digital signal are obvious, but it also has its own corresponding disadvantages, ie increased storage capacity requirements and increased channel capacity requirements during transmission. Taking a CD as an example, the sampling frequency is 44.1KHz and the quantization precision is 16 bits, so a stereo audio signal for 1 minute needs to occupy about 10M bytes of storage capacity, that is, the capacity of a CD turntable is only about 1 hour. Of course, the problem is even more pronounced in the world of much higher bandwidth digital video. Are all these bits necessary? The study found that there is a large redundancy in the direct use of the PCM code stream for storage and transmission. In fact, sound can be compressed at least 4:1 under lossless conditions, that is, only 25% of the digital amount is used to retain all the information, and the compression ratio in the video field can even reach to several hundred times. Therefore, in order to use limited resources, compression technology has received much attention since its inception. The research and application of audio compression technology has a long history, like A-law coding, u-law is a simple almost instant compression technology, and has been applied in ISDN voice transmission. Research on speech signals has been developed before and has matured, and has been widely used, such as adaptive differential PCM (ADPCM), linear predictive coding (LPC), and other technologies.

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Compression audio encoding Part 3

Compression audio encoding Part 3

I often hear what is called Hi-Res Audio. The sampling frequency is said to be 96 kHz or 192 kHz, which is over 48 kHz, the number of quantization bits is 24 bits, and the limit (high range) of human hearing is about 20 kHz, but it expresses frequencies higher than that. It will be. It is the same bit rate as the image from a long time ago. .. ..
By the way, it seems that dogs can hear up to 60 kHz and cats up to about 64 kHz.

Hi-res audio example
Sampling frequency Number of quantization bits Number of channels bit rate Frequency that can be expressed
192 kHz twenty-four 2 9.216 kbps 96 kHz
192 kHz 16 2 6,144 kbps 96 kHz
96 kHz twenty-four 2 4.608 kbps 48 kHz
96 kHz 16 2 3,072 kbps 48 kHz
48 kHz twenty-four 2 2,304 kbps 24 kHz
Considering the limit of human hearing (about 20 kHz), according to the sampling theorem, 48 kHz or 44.1 kHz is a sufficient frequency, but what about all of them? .. ..
In my case, I cannot distinguish the high resolution range, but it should be able to reproduce the discarded frequency at 48 kHz to 96 kHz, and when the number of quantization bits is in the 24-bit range, the sound pressure (dB) is a bit. Feels like I’m going up (?) (It’s just a story from my ears).
I’d like to make a comparison if I get the chance, but I don’t think I can tell by ear without a proper regenerator (like an expensive analog amp).

Is it time for cats and dogs to get verified in the acoustic industry? .. ..

Compression audio encoding Part 2

16-bit monaural PCM bit rate (for audio) (example)

Sampling frequency Number of quantization bits Number of channels bit rate Comments
32 kHz 16 1 512 kbps Super Wide Band
24 kHz 16 1 384 kbps
16 kHz 16 1 256 kbps Broadband
8 kHz 16 1 128 kbps Narrowband

Sampling rate
If you check the web, there are explanations like the sampling required to convert analog waveforms to digital conversion. For example, it shows how many samples of an audio signal input from a microphone are taken per second and digitized. The larger the sample, the greater the range that can be recorded. When an analog waveform is digitized, the frequency that can be expressed is half the sampling frequency (sampling theorem). For example, with a sampling frequency of 48 kHz, it can be expressed up to 24 kHz. At 8 kHz (narrow band) and 16 kHz (wide band), which are often used for audio, you can only hear up to 4 kHz and 8 kHz, respectively. The higher the sample rate, the higher the bit rate.

Sampling theorem
It is a very simple explanation, but it can express up to half the sample rate. When sampling a signal, if the interval is small, it can be restored close to the original signal, but if it is too thick, it cannot be restored (I would like to write a little more detail when I talk about signal processing or other time ).

44.1 kHz
Why is there a poorly separated rate of 44.1? .. ..
Isn’t the technician deliberately wearing an annoying watch to prevent music CDs from being easily copied? I heard something like that. When I searched, it seems this happened (?) Due to the convenience of an old PCM recorder. In this age, it is difficult to know what 44.1 kHz is in development. The 44.1 kHz ↔ 48 kHz sampling conversion is a headache. For example, USB audio (USB audio device class) exchanges data at 1 ms intervals. In the case of 48kHz, the data is 48 samples, but when considering 44.1kHz, it will be 44 samples (x9) and 45 samples (x1) in 10ms. If a sample of 45 samples is misled (tentatively), it will be 44.0kHz. I think it’s more like that with voice and music, and the human ear is mostly misleading (just my personal opinion).
However, the objective evaluation method will soon come to an end. For example, you can clearly see that you were fooled by a sine wave (sine wave) (maybe you are unexpectedly on the market).

Number of quantization bits
Sampling had to take a value in the direction of time (discretization), but quantization had to take a value in the direction of amplitude. The range that is possible to display the volume of the sound, which is heard often, “dynamic range 96 dB” means that the number of quantization bits is 16 bits and the music signal is played in the range of 0 to 65535 I can do it. The number of quantization bits is also called the bit depth or bit depth.

Bitrate
In communication, it indicates how many bits of data are transferred per hour and is generally expressed in bps (bit / s) of how many bits are transferred (processed) per second. If it is small, the size when saving as a file is small and there is space on the transmission line for communication. For example, when an audio (1 channel) is compressed to 1/3, the 3 channel audio can be sent at the same bit rate. Excuse the old story, but considering from the age of analog communication (analog mobile phone), digitization + compression will be able to support multiple calls with the same radio wave.

compressing using audio encoding

When compressing using audio encoding (AAC, MP3, etc.), the compression rate is determined by the bit rate at the time of encoding.

Specifically, if you set a low bitrate, the compression rate will be high and the file size when saved will be small, but what is the bitrate for the original sound source (PCM) without compression in the first place?

If you save it as PCM, the sound quality of the original sound will be obtained, but it can be a little inconvenient to save it without worrying about the file size. Also, depending on the application, I think the original sound size has enough memory capacity and the communication speed is correct. Therefore, I would like to write about the sample rate and bit rate that are often heard in digital audio.

The bit rate of digital audio is determined by the sample rate, the number of bits assigned to a sample (number of quantization bits), and the number of channels (stereo, monaural, etc.).

PCM bit rate (uncompressed) = sample rate x number of quantization bits x number of channels
As I wrote a bit last time, file containers like wav and mp4 format have this information as the header, so the application can see the header and play it. The compression rate of the encoding is determined by the bit rate specified at the time of encoding for this PCM (uncompressed) bit rate.
For example, as many of you know about music CDs, with 44.1 kHz stereo, this is the next bit rate.

Music CD bit rate: 44100Hz x 16bit x 2ch (stereo) = 1411.2kbps
When encoding this with MP3, AAC, etc., you will naturally specify a bitrate less than 1,411.2 kbps. For example, when encoding at 256 kbps, the compression rate is approximately 18% and the file size is 1/5 or less, assuming the original sound is 100%.

Encode 256 kbps music CDs: 256 kbps / 1,411.2 kbps = approximately 18%
Generally, the sample rates of audio devices actually connected to a PC are 48 kHz and 44.1 kHz for music, 16 kHz and 8 kHz for voice, such as microphones and headphones, and 32 kHz, 24 kHz, 22.05 kHz, etc.

The bit rate of PCM (uncompressed sound source) with 16-bit quantization bits is as follows.

Stereo (for music) PCM 16-bit bit rate (example)
Sampling frequency Number of quantization bits Number of channels bit rate Comments
48 kHz 16 2 1,536 kbps
44.1 kHz 16 2 1,411.2 kbps Music CD
32 kHz 16 2 1,024 kbps
24 kHz 16 2 768 kbps
22.05 kHz 16 2 705.6 kbps

Audio compression for music lovers

Audio compression for music lovers

the truth about high bitrate lossy compression

In the opinion of most people, the word music lover is most often associated with a person who not only loves and collects music, but also appreciates high-quality music, and not only in artistic and aesthetic terms, but also the quality of the recording of the phonogram itself. Just think, a few years ago, an audio CD was considered the standard for music quality, whereas a computer, even in dreams, could not compete with the quality of a CD. However, time is a great joker, and he often likes to turn things upside down. It would seem that quite a while, a year or two passed and … that’s it, the CD on the PC went into the background. Don’t ask “why?”, You know the answer to this question yourself. Everything is to blame for the revolution in the world of computer sound: audio compression (hereinafter referred to as audiolo compression which means lossy compression to reduce the size of the audio file), which made it possible to store music on disk hard, lots of music! In addition, it was possible to exchange it over the Internet. New sound cards have been released, capable of almost “squeezing” studio quality out of a piece of hardware that seems useless in terms of music. Today, even having a computer that is not very smart in performance, having bought a Creative SoundBlaster Live! and remembering that since Soviet times there is a good amplifier and good acoustics, you will get nothing but a high-quality music center, the sound of which is inferior only to very expensive audio equipment (average or even the highest Hi-Fi category ). Add to this the general availability of music files and you understand that you have the power in your hands. And then there is a revolution, and you understand that a compact disc is no longer so convenient, you are fascinated by something completely different: the magic “MP3” signs. You cannot eat or sleep; you are faced with the seemingly insoluble “chicken and egg” question: how to “squeeze” and, most importantly, how to “squeeze” …

This is where I will help you. This article is the beginning of my new series of informational materials on music on the computer. For over a year developing OrlSoft MPeg eXtension and maintaining an extensive database of MP3 files, I have accumulated a great deal of research on audio compression. It is these studies that I will try to share with you. Many articles have been written on audio compression by different respected authors, so I will try not to write what I can easily find in other sources of information. I would like to put my position on the subject we are considering simply and clearly. We will not consider audio compression to be as compact a tool as possible put audio information on your hard drive (so that you can record so many hours of music there). Yes, compression allows you to record music more compactly, but my goal is to minimize quality loss by converting “pure” audio to compressed audio. This is why only high bit rates and qualitatively compressing encoders are considered in these modes. So it is much more convenient to work with compressed audio – instant access to any track from any album, convenient software for playback. And, of course, the financial issue has not been forgotten either.

Of the audio compression formats that exist today, in my opinion, three deserve attention: MP3 (or MPEG-1 Audio Layer III), LQT (as representative of the MPEG-2 AAC / MPEG-4 family) and a Completely new OGG format (Ogg Vorbis) developed by a group of enthusiasts:

MP3 is by far the most used of these (mainly because it is free). Let me remind you that it was thanks to the MP3 format that the victorious procession of compressed audio took place. However, as often happens with pioneers, little by little it is losing ground and giving way to new and better formats.
The second format, LQT, is a representative of a new direction of audio coding algorithms, a representative of the AAC family. This is a fairly high quality, but commercial and highly classified format.
OGG became widely known to the public this summer and is currently developing rapidly, soon (with the launch of the Encoder and Decoder) it should beat MP3 with better sound quality with smaller file size.

How to choose the perfect compressor configuration

Compressors and how to use them, explained.

Compression is one of your most powerful mixing tools. It is the essential element behind any good mix.

But for your compressors to work, you must first understand what compression is.

It can seem intimidating to start learning such a broad subject, especially when the controls and how they affect the signal are difficult to understand in relation to the sound.

This article will help you understand what compression does, how to choose the perfect compressor setting, and some common mistakes to avoid.

But before…

What is compression in music?

Compression in music is the process of reducing the dynamic range of a signal. Dynamic range is the difference between the loudest and quietest parts of an audio signal.

audio compression

You must reduce the dynamic range of most audio signals to sound natural to a recording.

For example: imagine a whisper and a scream on the same audio track. If they had the same volume difference as they do in real life, it would be very annoying!

Compressors fix all of this by attenuating the loudest parts of the signal and boosting what is output so that the quieter parts are more noticeable.

Imagine a whisper and a scream on the same audio track. If they had the same volume difference as they do in real life, it would be very annoying!
Using compression
Experienced engineers often talk about how one compressor is more “musical” than another.

It is an important concept. Its dynamics is one of the fundamental aspects for its sound to be unique.

When you use a compressor to change the dynamics, the sound engineer becomes part of the musical performance.

If your compressors work properly, they will positively contribute to performance and improve recordings.

Transients: understanding high energy moments.

To understand compression, you need to know what transients are.

Transients are the first high-energy moments of a certain sound in its waveform. These explosions give our brain a lot of information about the quality of a sound.

Since transients are usually louder than the rest of the waveform, they are greatly influenced by compressors.

For example: think of a nice roaring trap. As soon as the trap enters, there is an initial peak in the waveform that narrows slowly. That initial energy spike is your transient.

transient compresor

Compression helps you find the perfect balance for a track that has good dynamic range with a beautiful, full body.
A waveform with good dynamics will have a lot of transients when some sounds hit and then decay in the composition. Transients and their final decay are what make a waveform similar to a fish bone.

There is even an overly dynamic trail. If your song is transient without a body, its sound will not be of interest to your ear.

The reverse is also true, no dynamics can lead to lifeless, exhausting sound for the human ear and a waveform that looks like a big brick.

Compression helps you find the perfect balance for a track that has good dynamic range with a beautiful, full body.

Limiter

The threshold determines the signal level at which the compressor will start operating. The threshold is measured in dB, therefore any signal above the set threshold will be compressed.

When setting the threshold, decide what part of the signal you want to reduce.
With the threshold low, the compressor gain reduction is applied to a larger portion of the signal. Setting it higher affects only the most aggressive peaks and leaves the rest intact.

To determine what the perfect threshold is, think about what you’re trying to accomplish by compressing the audio and which parts of the signal are the most troublesome.

Are strong signal transients distracting you from the rest of your mix? Or maybe your final decadence is imperceptible in the mix?

A good rule of thumb for compression is “do no harm.”
Set the threshold to hear compressor operation on the part of the signal that needs to be addressed and not lowered.

Setting the perfect threshold will depend on your needs. Play the track and tweak it on the go to find the perfect amount.

Relationship

The ratio determines the amount of gain reduction applied by the compressor when the signal exceeds the threshold. It is called a relationship because it is expressed in comparison with the unaffected signal.

The higher the first number in the report, the greater the gain reduction factor.

For example, we can say that an uncompressed signal would have a 1: 1 ratio

What is the compressor and how does it work?

The compressor, together with the equalizer, is one of the most important and most used processors in professional audio, but its operation is not always so intuitive and knowing how to master the compression technique sometimes requires years of experience. In this new article we begin to explore this fundamental processor.

What is the compressor for?

First of all, let’s start to see what the compressor’s function is: to reduce the dynamic range of an audio track, that is, to decrease the distance in volume between the weakest signal and the strongest signal. Initially created to optimize recording on magnetic tape and to avoid saturation of the input stages, the compressor is still used today during recording and mixing. Reducing dynamic range also allows us to keep multiple tracks in the mix, such as a voice, for example, always at the same volume throughout the song so that they are not dominated by the other instruments in the most crowded sections, as well as to avoid Output saturation.

Compressor

Back to basics: what is the compressor and how does it work

The controls

Now let’s see in detail what the various compressor controls are and what they are for:
— Threshold: or threshold, expressed in dB, indicates the point beyond which the compressor begins to operate.
— Ratio: is the compression ratio and indicates how much the signal will compress when it exceeds the Threshold. For example, with a 2: 1 ratio, each signal that exceeds the threshold will be halved at the output, that is, every 2 dB at input 1 will be returned at the output.
— Make Up Gain: This is the output of the compressor and is used to recover the volume lost due to compression.
— Attack: expressed in milliseconds is the time it takes for the compressor to start once the signal has passed the threshold.
— Release: always expressed in milliseconds, it indicates the time it takes for the compressor to stop compression once the signal has returned below the threshold.
— Gain reduction meter: it is not a control but a visual indicator, led or pointer, which informs how much the signal is compressed, through a scale in dB.
— Bypass: shuts down the processor, making the signal pass through the machine without alteration.

With the advent of digital and accessories, we can find controls that not all hardware compressors have:
— Knee: indicates the type of curve at the point where the compressor begins to operate, which can be abrupt (Hard Knee), soft (Soft Knee) or various intermediate values.
— Automatic: sets the time control to which it refers (attack, release or both) automatically, depending on the input signal (program dependent).
— Sidechain eq or External Sidechain: Sidechain is the signal that drives the compression circuit, where in most cases it is the signal itself to compress, but sometimes it can be a version of the input signal with different equalization, for example without low frequencies, so that they don’t start the compressor too soon. Or it can be an external signal, such as the one used on the radio where the speaker’s voice signal drives a compressor on the background music signal, so it automatically turns off when it starts to speak (Ducking), or Classic Speaker Use to activate the compressor on various instruments in the mix or the Master Buss.
— Mix: used to mix the compressed signal with the original signal. This way, you can use Parallel Compression directly on the compressor, without having to use two mixer tracks (one for the dry signal and one for the compressed signal).
Back to basics: what is the compressor and how does it work

Compressor

Compressor or limiter?

What is the difference between a compressor and a limiter?

Essentially, the compression ratio: over 10 dB ratio, the processor is considered a limiter. A separate case is the Brickwall Limiter, a compressor with immediate attack and a compression ratio of infinity to 1, so that no signal can exceed the Threshold. It is mainly used on the master buses so as not to exceed 0dBFS on the output and then send the converters to clips.

Usage examples

As we already said, the compressor is used to keep the volume excursion under control. One track in the mix: in this case, using a fairly fast attack, slow release and not too aggressive ratio, allows us to compress the signal constantly and transparently, that is, without making your intervention feel excessively.
The compressor can also serve to emphasize the attack of a percussion instrument: in one case, for example, by setting a medium slow attack.

Audio normalization or compression

The function of a compressor is to reduce the dynamic range of the signal, that is, the level difference between the strongest and weakest signal parts.

Why compression or normalize?

At the time of analog, the limited dynamics of the main musical supports (vinyl, audio and video cassettes) did not allow to reproduce the dynamics of a classical, jazz or even rock orchestra in the case of the audio cassette. Therefore, the signal was compressed to avoid distortion in the transmission medium.

audio compression or normalization

Now that the music is converted to 16-bit or more, recorded in digital format, and then streamed to CD / DVD or downloaded, the dynamics of the media is enough to faithfully reproduce the dynamics of almost any orchestra. The old technical limitations have disappeared, therefore compression is no longer essential.

However, whatever the musical genre, some sources (voices) are compressed almost systematically. The goal of modern compression is therefore to optimize sound recording, either to get closer to reality or, conversely, to create a less faithful but denser, more controlled, more powerful sound, etc., or even a sound. totaly new.

And to do all this, the compressor is satisfied with a simple principle: it reduces dynamics by attenuating the signal level when the latter exceeds a given threshold level.

Level settings

– Threshold (threshold level, in dB)

This parameter determines the threshold level from which the compressor is triggered. As long as the input signal level remains below the threshold, the compressor does not start and no treatment is applied. As soon as the source signal exceeds the threshold level, compression is applied.

– Ratio (compression ratio)

The ratio determines the amount of level reduction applied to the part of the signal that exceeds the threshold level, the rest of the signal is not processed. Depending on the compressor, the ratio can vary from 1: 1 to Inf: 1. Quésaco?

Set up a compressor

With a 1: 1 ratio, no compression is applied, the level of the input signal is equal to that of the output signal. With a ratio of 2: 1, the level of the signal portion that exceeds the threshold is divided by 2 in the output signal. With a 3: 1 ratio, it is divided by 3, etc. When the compression ratio is infinite (Inf: 1 ratio), the compressor behaves like a limiter: the output signal never exceeds the threshold level, regardless of the input level.

Therefore, the compression intensity applied to the signal is a compromise between the threshold and the compression rate setting:

The lower the threshold, the larger the compressed signal portion.
The higher the ratio, the greater the level reduction applied to the signal portion above the threshold.
Depending on the compressors, you may find other parameters, for example, an input level setting instead of the threshold, or a gain setting (also called the offset or output level) that amplifies the signal to compensate for the drop in level resulting from compression.

Time settings

– Attack (attack, in ms)

Attack corresponds to the time the compressor needs to reach the given ratio when the signal level exceeds the threshold level. A quick attack of a few milliseconds triggers strong compression as soon as the signal level exceeds the threshold; With a slower attack, the compressor passes the first transients of the signal peaks, keeping one side alive and well cut.

Set up a compressor

– Launch (launch, in ms and s)

Release corresponds to the time the compressor needs to return to the 1: 1 unit ratio when the source signal falls below the threshold level. A quick launch of a few tens of ms allows the original character to stay alive. Slower relaxation improves instrument resonance and reverberation, but can cause compression of the first peak transients when the latter are close together.

– Knee (literally knee!)

The Knee parameter determines the increase in compression, that is, the transition between the compression ratio of the unit (1: 1, no compression) and the compression ratio set to ratio.

Applications

At the output, the compressor can be used as a limiter to control signal peaks and prevent distortion from occurring in the analog / digital conversion stage.
When taking and mixing, light compression can bring out weak parts of the signal and thus reveal certain details.
In the mix, the compressor allows you to increase the average level of the audio volume output.

Destructive compression vs non-destructive

Destructive compression is compression obtained by losing information. This means that if you extract the compressed signal with this technique, you will not find the start signal.

Destructive Vs Non-Destructive Audio compression

In destructive compression techniques, there are basically methods that take advantage of the properties of the human ear. The latter listens to frequencies between 20 Hz and 20 kHz. If a song contains frequencies outside this range, we can easily delete them without losing the audio quality because the ear does not hear them. In fact, frequencies between 2 kHz and 5 kHz are generally heard correctly. In fact, less than 5 dB is required to hear frequencies in this band, while more than 20 dB is required to hear frequencies below 100 Hz or above 10 kHz. These results can be used to reduce the size of the files. For example, we can conclude that all frequencies above 15 kHz are suppressed.

audio compression

MP3 also uses the principle of masked frequencies. If in a frequency group some have a much higher noise level than others, it is not necessary to keep the frequencies low – we will not hear them. Imagine yourself in your garden listening to the birds sing. The chord goes over your head (even very high). We no longer listen to birds because the sound they make is much quieter than that of the plane. It is as if the birds no longer exist or have stopped singing. Obviously, it is not necessary to code all the frequencies present in a song so that the human ear can always perceive it well. Finally like the two ways

What do we find among non-destructive techniques?

Mainly coding techniques.

Let’s explain. A sound is a frequency. A second of music is therefore a sequence of frequencies. Imagine that in the series of samples that make up a second of music (remember that there are 44,100), we have the same frequency several times in succession, for example 10 times. If instead of storing these 10 points, we only store 1 and the number of times it is repeated, we must encode 2 digits and not 10. If we also apply this method to frequencies which are no longer identical, but very dense together (so close that the average human ear cannot distinguish them), we can still save space. This time, the compression is destructive because we are replacing one frequency with another frequency (almost identical).

MP3 also uses the algorithm of Huffman (1952) as a method of encoding information. This method is used in all compression algorithms (compression of text files, compression of images, compression of sound). It is based on the use of a variable length code and the probability that an event (in this case a frequency) will occur. The more a frequency appears, the shorter the code (low number of bits to display it). The file is read for the first time and a table appears with the frequencies that appear and the number of times they appear. We derive the right code. This encryption was last used. It is the final phase of compression. This is non-destructive coding.

MP3 works on the properties of the ear, first to reduce the size of a part, then processes the stereo sound and possibly applies encodings which end with Huffman encoding.

The use of all the reduction options mentioned depends on the location you want to give within 1 minute of your tablet and therefore on the compression speed to apply.
To encode MP3 audio files, we are talking more in terms of bit rate than compression rate.
Bit rate is the number of bits allowed in 1 second.
Therefore, we have the following relationship: the more we want to compress a song (so that it takes up the least space possible), the lower the bit rate.

Choice of compression ratio (bit rate)

Obviously, the more you compress, the worse the sound quality.
You have to compromise the file size and audio quality.
This commitment can be dictated by your needs, but also by the use you want to make of your MP3 files. It may not even be demanding if your MP3s are intended for your portable music player and are too demanding to be listened to on a stereo system.

Does MP3 affect the sound quality?

The compression of songs affects the quality, but the losses are not necessarily audible.

Is compression of MP3 songs harmful to the sound quality? Whether it is HD music or “normal” definition, the question of compression remains. The advantage is that the weight of the songs is reduced, so they take up less space in the memory of a phone or a portable music player. With standard MP3 compression, a music album ranges from 500 MB to 45 MB.

But by the way, the music is damaged. The sound seems a little less natural, less precise, less dynamic. Some of the audio information is literally destroyed. It doesn’t always sound good, but for some songs the difference is clear until everyone will notice.

Fortunately, you can improve the quality of an MP3 song by compressing it with less force. The loss of sound quality becomes less clear, but in return the song weighs more. MP3 isn’t the only compressed music format that corrupts music. The most famous competitors are AAC, Ogg Vorbis and WMA. MP3 is not the most efficient compression format, this title applies to the Ogg Vorbis, but it is still a good option. All music players can play MP3 and online record stores prefer this format.

Lossless compression

However, some music lovers are reluctant to MP3. They swear by “nondestructive” compression, which does not remove sound information. The music has been completely preserved: we hear absolutely no difference. The best known non-destructive formats are Flac, APE and Alac. Unfortunately, not all electronic devices can play music recorded in these formats. Few artists offer their music in “non-destructive” compression. And the weight of the parts thus compressed is still very heavy. An album quickly reaches several hundred megabytes. However, the Flac stands out as the reference format for the most demanding music lovers.

Is it reasonable to keep using MP3? This remains a smart choice for most music lovers, as long as they choose an appropriate compression ratio. Which one to choose: 192 kbit / s, 256 kbit / s or 320 kbit / s? The stronger the compression, the lighter the number, but the lower the quality. With 128 kbit / s, the sound has clearly deteriorated, most of us can hear it. At 192 kbit / s, degradation becomes difficult for most of us to observe except for some rare numbers.

With 256 kbit / s, you have to have a musical ear and good sound equipment to make the difference. With 320 kbit / s, you need a well-trained ear and highly accurate audio equipment to make a difference. We only see a difference in quality in certain titles and only in certain passages. Therefore, most of us can settle for 192 kbit / s recording. Music lovers should expect a minimum of 256 kbit / s. And professionals will choose formats of 320 kbit / s or ‘lossless’.