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Loudness Normalization: Why is it necessary to Normalize the loudness of an audio or a video?

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Loudness Normalization: Why is it necessary to Normalize the loudness of an audio or a video?

Loudness

The war of volume or loudness war.

Already in the 1940s and in later decades, in the middle of the vinyl record era, a volume war was experienced.

The goal was to make a song sound louder on the radio, louder than other songs and louder than advertising.

Sure, the limitations of vinyl didn’t allow the ability to indiscriminately increase volume to be possible.

Loudness normalization

But with the advent of CDs and digital music it was possible to push the loudness of a song to the max. The situation is that the digitization of the audio allowed it to be manipulated quite precisely, achieving dynamic normalizations that actually ended the dynamics of the music and then played all the time at maximum volume.

By the 90s, groups like Red Hot Chilli Peppersm and their album Californication took this war of loudness to levels rarely seen.

But why did they do that?

Some research on human hearing showed that people did not find that a song sounded better if it had louder loudness.

Every artist, every producer, and every hardware manufacturer has figured out a way to make their production sound louder, louder.

Digitally many limiters and compressors pointed in that direction and made a lot of music sound almost to the point of distortion.

Each one wanted their music to stand out, among other things for being louder and having a greater sound, a higher volume level.

If to this recipe we add the appearance of the mp3 and a great variety of encoders, and also that ordinary people did not understand the effect that the bit rate could produce, then many mp3s with different qualities were generated.

The possibility of sharing these mp3s filled people with mp3s that each had very different sounds. Both for its production and for its coding.

Then a new need appeared: normalize the music to avoid these disparities in loudness, in the volume of the songs.

The holy grail of normalization had to be found.

Many ideas were found, many experiments. The situation matured and certain products like Mp3Doctor and Mp4Gain matured to the point where they actually managed to find the solution: a dynamic standardization that will work well with today’s advanced player equipment.

Then Mp4Gain made the leap, achieving that even videos could not be normalized.

Audio could already be normalized in its main formats (mp34, aac, ogg, floac, etc) with Mp3Doctor, but Mp4Gain added the possibility of these dynamic normalization to video in its main formats (mp4, 3gp, flv, avi, etc. )

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Audio normalization for beginners

What’s more annoying when listening to music is that you have to manipulate the volume control for every song that plays. If you have a computer, a tool allows you to uniformize the atmosphere from track to track while the songs are playing. This is called normalization. Three main means are used to achieve this result more or less effectively.

Audio normalization

Normalization through detection of maximum volume

The player or audio processing software analyzes the sound of the track and detects the highest amplitude. If it is less than the maximum gain value that is imposed, the signal is automatically boosted by the number of decibels required to reach and reach this value in all samples on the track. If the highest amplitude is equal to or greater than the maximum gain value, nothing is done.

Normalization

This method has only one advantage: the avoidance of saturation. However, the drawbacks are many.

This form of normalization cannot be applied in real time, as it is assumed that the maximum signal value is known in advance, which is hardly the case with live audio sources (playback or recording). Also, this type of normalization turns out to be totally ineffective when the overall sound of the song is low, but interrupted by small ridges that can be parasitic. When these peaks reach or exceed the maximum gain value, nothing happens and the overall sound is always reduced, especially if these peaks last only a few fractions of a second.

Normalization in detecting maximum volume is almost never used by reading software. Many audio processing software or even audio CD burning offers this option, such as Audacity and Nero.

Normalization by medium volume detection

Here, the player or audio processing software analyzes the sound of the track and does not detect the highest amplitude, but the average amplitude of the signal. Thus, the volume of the song will automatically increase or decrease by the number of decibels required to reach the imposed value, as appropriate.

Also known as RMS, this method has the advantage that the sound is fairly accurately balanced from one song to another, even if there are sharp peaks in the volume.

However, normal normalization of volume detection, like the previous method, cannot be applied in real time and is ipso facto unsuitable for live audio sources. In addition, saturation can occur if the imposed value to be achieved is not sufficient. It is recommended to use normalization values small enough to avoid this problem as much as possible.

Many reading software programs use this normalization mode, but they all work better or worse than the others. .

Sound compression / modern normalization

The mp4gain audio processing software performs the audio signal analysis, analysis that will lead to increase or decrease the volume of certain areas of the signal according to a complete set of fairly complex parameters inherent in the signal itself. Ultimately, the loud sounds will be attenuated, the weak sounds will improve when multiple presets are reached.

This is the best normalization method if the sound processing values are well established, in which case the sound volume becomes very constant and without saturation, regardless of the source and signal type, in real time or No

However, this type of normalization requires some processing power from the processor. Although the results achieved are much more professional and the only ones that really achieve what the 2020 ear is looking for. Mp4Gain has the most efficient response to normalize audio, either from audio files of the most popular formats or from video files, including the most commonly used formats.

What exactly is normalizing?

Music is distinguished by what is often called “dynamic” and which refers to the changes (more or less abrupt) of the “effort” with which certain notes or passages are interpreted.
Whether it is an instrument or the voice.

singer

Any vocal performance that has been considered virtuous, in general terms, will have a dynamic that goes from very soft passages, almost whispered, to intense passages, with a high volume, singing at full voice.

At the time when vinyl existed as the option to listen to music, it was not felt (at least it went almost unnoticed) the fact of noticeable differences between the loudness or the volume of a song.

It was with the advent of digitization and the possibility of its variants (opting for different bitrates, sample rates, bitdepths, etc.) that this difference became very evident.

And with the appearance of mp3 and its distribution or exchange, at the same time that winamp and distribution lists arose, when it was inevitable and it was even started to look for solutions.

Napster

These first ones were based on the sound peaks and their results were very inefficient.

Returning to the mention of the mp3, situations such as masking (where information is removed) further marked the problem of differences in volume.

Then began to use the RMS that rather mediates the average power that the song had, more than the peaks.

Initially, it was enough to put a slower reaction level to the volume meter, to have a more general idea and less impacted by the volume peaks.

And so, the way of listening to music and considering what normalization was evolved.

Finally it appears to be somewhat closer to a mixture of a volume limiter and a compressor.

What is a volume limiter? It is a hardware (although lately there are also limiters in software version) that ensures that no peak exceeds a maximum limit.

A compressor, on the other hand, is a device or software that is used to “compact” the volume, preventing the parts with the lowest volume from being too low and at the same time preventing the high parts from exceeding a range that has been assigned. We would say that the compressor dampens the increases and decreases in volume.

To this we can add an equalization that differentiates the bands and treats them differently both in the limitation and in the compression. Each frequency band has a different treatment in the Mp4gain and that produces a very efficient result. It is NOT the only improvement offered by Mp4Gain, but this is described here. In other articles we will deal with other differences.

Mp4Gain is the best normalizer of 2020 and this is clear when using it.

Audio Level normalization

The audio levels of the material produced in a radio station
In general, in radio they do not tend to stay within standardized levels for their audio editions (spots), it is not necessary to know much about levels, since an audio processor compresses and limits everything on air.

The console operator does not understand anything about dynamic range, something that has no practical use in the air. And this is how many radios work with adjustments that “work” in the air by trial and error, and not always with the most demanding criteria. successful.

Level normalization

In radio, an editor does not know or manage any level convention, so it could be said that level normalization is not widely used. However, a good professional practice would be that all the material generated by a station “sounds” at the same level. Not to the air, because to the air if it is transmitted normalized or compressed and limited, but inside the station. And for this, there are two ways:

The material is processed “by ear” by comparison.
An RMS value is defined and all publishers normalize their mixes to that average level.

Regarding the first point, differences of up to +/- 2 dB will be absolutely acceptable. But a very common vice is to overcompress the edits, or sometimes the voices, seeking to hear the compact and aggressive sound of the FM on studio monitoring. That sound should be determined on-air by the streaming processor, not the publisher. Editors generally abuse processes like Normalize RMS (Sound Forge) and “maximizers”; Wave Hammer (Sound Forge / Vegas) Ultramaximizer and L1 (Waves). Ideally, how much to “squeeze” the dynamics of the edited material should be a function of the type of processor the radio has. At this point it is possible to clarify a fairly common confusion: STANDARDIZATION has nothing to do with making an audio sound “strong” or “powerful”. Using normalization for that purpose is a beginner’s mistake.

The second option is the most accurate way of working -although this precision is not necessary- normalizing all the editions to a given RMS value. This does not impact the sound in the air but it does the internal prolixity of the station. RMS is not an accurate measurement of loudness or “volume”, but for what you need in radio it is enough.

The streaming audio processor knows nothing about the level of the audio file. The processor receives an audio level from the console and works accordingly. What affects the behavior of the processor is the dynamics of the material, if it has dynamics or is super-compressed / limited.

Normal working values

The level at which operator-editors generate material has two well-defined extremes to avoid: very high levels of compression / cliping and excessively low material (less than 24 dB RMS). When we talk about level, we must be clear about the differences between peak level and average level.

PEAK level

Regarding the peak level, the logical maximum limit is digital cliping. Needless to say, a cliping mix is unacceptable.
It is advisable that the maximum peak level is not 0 dBfs, as this will generate overshoot cliping in the D / A converters and especially if the compressed material (MP3) is exported.
An appropriate value for the material on a radio is maximum peak – 1dBfs (the recommendation if using mp3 compression is -3 dBfs). But this does not mean that it should be -1 dB. If no peak reaches the established maximum it is not a problem as long as the material complies with the appropriate working level. The peak level does not matter, but in general the signal will always reach the maximum peak level.

Listening level (RMS)

The “listening level” or mix level is determined by the RMS or “average” value of the material. This is true even if the publisher has never measured the RMS value of their audios. In general the radio editor “compresses”, “maximizes” or -conception error by- “normalizes” your edits “so that they sound”. And in that “so that they sound”, it is taking the cuts to a certain value.

The question that arises is what should that value be? How much should the final mix “squeeze”? The final value should not be a value that generates excessive compression, as this is the task of the transmission processor. How to compress is a topic of discussion for another article, since it is fine spinning and the radios in general do not take into account these aspects. In general lines we will say:

If the radio has a simple analog processor, type M31 or Solidyne 362, they will perform better with material that has a more compact sound (more compression).
If the station has a high-end digital processor, and especially if it works with a highly processed sound in the air, it is not recommended or necessary to excessively maximize the material generated by the station, because these audio equipment respond better when the material is origin is not over compressed.

But what if the file level is very low? It depends. Depending on the PC-Console connection, the operator typically has at least 15 dB of gain range for level correction from the PC. In turn, if the level is low with the fader on, the AGC of the processor has between 10 and 20 dB more correction to compensate the level in the air. But if the file were generated too low, it could fall outside the operator / processor correction range and go low on air.

GENERAL AND ELEMENTARY CONCLUSIONS:

Different materials generated in the radio must sound at the same level, either by ear or measured RMS.
It should not be overcompressed, much less cliping.
The peak level should not exceed -1 dB.
It should not be too low as it may fall outside the processor’s AGC / operator correction ranges.

Put in values:

RMS values between -16 to -13 dB RMS are acceptable.
Values between -13 and -10 dB RMS generally indicate strong compression.
Values less than -10 dB RMS indicate excessive compression, not recommended as it generates a very loud but “muffled” sound that cannot be “improved” by the air processor.

Normalization of an audio file.

Normalization is used to increase or decrease the level of the song as a whole, so that its maximum volume peaks assume the indicated level.

Loudness Normalization

For example, if the maximum intensity points of the song are -3 dB (therefore well below 0, which should represent the maximum before distortion), normalizing to 0 dB means increasing the level of the entire song so that these peaks reach 0 dB.

This is the typical normalization of the peaks.

There is also RMS normalization (which takes into account not the peaks but the actual average level of the song).

Audio Normalization

AUDIO CDs, which have good dynamic possibilities (various intensity tones, from pianissimo to fortissimo), are generally recorded so that the maximum volume points are at 0 dB.

Normalizing your WAV recordings can be helpful in adjusting them to the average level of a CD in case they are too low (because you had been careful in level during recording) but one important thing to note:

Normalization of this type alters the original dynamics, that is, the reciprocal relationships between weak and strong sounds.

Although all levels are raised by the same amount, the relationship between 2 levels changes (small mathematical example:
2/5 = 0.4 ma (2 + 1) / (5 + 1) = 0.5 …

The result is that the weaker sounds, after abrupt normalization, sound much louder and those that were already playing only sound a little louder … altering the dynamic relationships that had been envisioned by those who originally recorded the music and making the sound output to lose depth.

Some types of music, generally already deficient dynamics (rock, metal, etc.) since the excursions between the minimum and maximum volume are almost never very consistent, are more “normalizable” without problems, while the genres in which there may be Large Dynamic excursions (classical music or music with passages from pianissimi to fortissimi) are more problematic.

In addition, it is necessary to take into account that if you normalize a large wav file that contains many songs (not yet divided) there can still be, even in genres with little dynamics, substantial differences, in this case between one song and another and not between different points of the same song.

So a light normalization can do and is actually used (to raise the level of the part), but it would be better to make sure you don’t need it (recording from the beginning with a good level) or at least not have too much. remember, however, that the dynamics are somewhat flattened.

Normalize with Mp4Gain

This software is capable (it is the only one that can do this) of normalizing the main audio and video formats and its standardization algorithm is by far the most efficient and the one that produces the best results.
For this reason it is used by musicians, radio broadcasters, universities, television stations, producers, etc.

Audio normalization explained

Audio normalization – Audio normalization

Audio normalization is the application of a constant amount of amplification of a sound recording to bring the amplitude of a target level (standard). Because the same amount of gain over the entire recording, the signal-to-noise ratio and relative dynamics are unchanged.

Two basic types of audio normalization exist. Peak normalization adjusts the recording based on the highest signal level present in the recording. Loudness normalization adjusts the recording based on perceived loudness.

Normalization differs from dynamics compression, which applies varying levels of gain across a recording to fit the level within a minimum and maximum range. Normalization adjusts the gain with a constant value over the entire recording.

Normalization is one of the functions usually provided by a digital audio workstation.

Peak normalization

One type of normalization is peak normalization, where the gain is changed to bring the highest PCM sample value or analog signal peak to a certain level – usually 0 dBFS the loudest level allowed in a digital system.

Peak normalization

Since it only goes to the highest level, only peak normalization does not take into account the apparent loudness of the content. As such, peak normalization is commonly used to change the volume so as to ensure optimal use of the available dynamic range during the mastering phase of a digital recording. In combination with compression / restriction, however, peak normalization becomes a feature that can provide a volume advantage over off-peak normalized material. This feature of digital recording systems, compression and limiting followed by peak normalization, sets contemporary trends in program loudness.

Loudness normalization

Another type of normalization is based on a measurement of loudness, where the gain is changed to bring the average amplitude to a target level. This average can be a simple measurement of average power, such as the RMS value, or it can be a measure of human perceived loudness, such as that offered by ReplayGain, Soundcheck and EBU R128.

Loudness Normalization

For example, YouTube reference level -14 LUFS, so if a program analyzed at -10 LUFS, YouTube will decrease the level 4 dB to the reference of -14 LUFS.

Loudness normalization was made in different volume combat when listening to different music in a series. Before loudness normalization, one song in a playlist would be quieter than the rest, so the end listener would have to put a volume knob to adjust the playback volume.

Depending on the dynamic range of the content and the target level, loudness normalization may result in peaks that exceed the storage medium. Software offering such normalization usually offers the option of using dynamic range compression to avoid clipping when this happens. In this situation, signal-to-noise ratio and relative dynamics changed.

Volume normalization, an explanation

Audio Normalization: Make Your Audio & Video Consistently Loud

Audio normalization is a process in which the amplitude (volume) of an audio recording is increased or decreased in a constant relationship over time, so that the maximum amplitude or the maximum effective value or the perceived volume (volume) reaches a predetermined level, the standard. If the signal has multiple tracks, they all undergo the same correction.

Normalize Audio

Example: normalization of peaks to -3 dB:
A collection of digital recordings is made with a peak modulation standard of -3dB FS.
A new stereo recording is measured. The highest maximum level is -5.5 dB FS on the left track, -5.7 dB FS on the right track.
Normalization consists of applying a constant gain of 5.5 – 3 = 2.5 dB.
Standardization requires two passes. The first determines the maximum level, the second applies the correction to the entire recording.

Audio Normalization

Maximum normalization changes the level, but not the dynamics of the sound.
Volume normalization or perception of loudness often includes compression that changes the dynamics of sound.

Peak normalization

Peak normalization applies a constant gain to a recording to bring the highest peak to a target level, 89% professional audio (-1 dBFS true peak (True Peak)).

The sound dynamics of the recording are more or less preserved, except that maintaining a low distortion level after multiplication of all samples may involve the application of a known quantization error decorrelation noise. under the name redithering (tingling of the least significant bit) 2, which slightly increases the background noise level.

Volume normalization

The purpose of volume normalization is to bring all sound elements in a collection to the same sound volume level, so you can hear them without having to adjust the volume. In fact, the normalization of the maximum level in no way guarantees a homogeneity of the perceived sound volume (Loudness).

A simple approach to volume normalization, which is provided by various software programs, is to normalize the RMS value of the integrated signal within a few tenths of a second. The most advanced machines use extensive algorithms for more accurate evaluation of the perceived noise level. The European Broadcasting Union published a recommendation 1 in 2011, which provides a relatively simple method for this evaluation.

If the standard is not low enough, volume normalization involves compression for recordings whose sound dynamics would be higher than implied when setting the standard from the maximum level. If not, the signal peaks would exceed the quantization limits.

In the simplest implementation, volume normalization collects volume data during the first pass, determines the gain or attenuation necessary for the maximum volume to reach the norm, and applies this correction to the second pass. If the elements of the collection have the same characteristics, from form factor to top factor and dynamics, as is the case with popular music collections or recorded speech, this approach produces satisfactory results.

Extensive implementations use a standard that includes not only the volume of the sound, but also the maximum maximum values and dynamics of the sound. They collect loudness levels and maximum values

VOLUME NORMALIZATION: WHAT IS THE VOLUME NORMALIZATION FUNCTION?

Audio Normalization

HOW IS THE VOLUME BETWEEN TITLES NORMALIZES?

WHAT ARE the benefits of activating the normalization feature?

The “NORMALIZE VOLUME” volume normalization feature allows you to achieve a volume of the same level, music title after music title, regardless of which one succeeds during playback.

How Audio Normalization Works

This provides undeniable listening comfort rather than having to, as before, sometimes turn the volume up or down depending on certain pieces of music.

Note that this difference between a high volume and a low volume sound is called dynamic. If this sound is short or long (1 second or 3 minutes …), be it music, voice or noise.

WHY IS THE SOUND MUSIC OR LETTERS STRONG, SOME TIME?

We must not forget that music, recorded or not, as well as everyday acoustic sounds (those that surround us) are something “alive” which, like during a human discussion, necessarily contains volume passages. weaker sound and others that are louder.

The human ear is by definition used to these differences in sound levels. If these sound differences between the low and high levels did not exist, it could end up giving us a headache because the sound heard would not be natural. The ear needs moments of rest, even if only for a moment, and stronger moments for words to remain audible (and to work the ear again!). The human ear needs this natural “breath”.

Today’s music is very “compressed” (constant sound level, few low levels) when recording (mixing), that is, there are few passages with a big difference between the lowest and highest passage of the song. The STANDARDIZE feature can even be activated and not work much if all the titles of an album are very “compressed”.

Finally, the sound world is like the aquatic world: there are high and low waves. Some tracks are not recorded (mixed) like others. They leave a big difference between low and high noise levels. The NORMALIZE VOLUME feature allows you to level up and try to get everything back to the same level.

WHEN SHOULD I USE THE STANDARDIZATION FUNCTION?

Eg. On the street or in the subway, standardization plays an additional role in making your music more audible. And of course, first and foremost to get a comfortable listening when listening to different music titles with as much sound as possible.

WHY disable the standardization feature?

When the need arises, you can turn this feature off at any time when you want to find this “breathing sound”, titles read (play), with multiple moments that contain smoother (weaker sounds) and higher variations (it’s loud).

Especially if all the titles in your album or playlist are compressed at source, disabling the NORMALIZER feature will help your ear rest, at the end of the day you will be less tired.

Deezer’s NORMALIZE feature does not compress sound and fatigue, it only reduces the major differences between high-level and low-level titles.

Digital audio normalization

In the last decade the term digital audio normalization has become popular. You could say that most people have a vague idea of what they mean. However, it is important to understand some concepts that relate very closely to the issue of the volume gain of an audio file.

One of these issues is audio quality, so we think it is very important to start by explaining what kilobytes per second means.

It is not difficult to understand this concept, however very few people understand it and much less people manage to understand

So let’s try to understand what the subject of kilobytes per second means and how it impacts the quality of an audio file of any format.

This will allow us to have a greater vision to understand the issue of volume, digital audio normalization and loudness given that all this is closely related to audio quality.

So let’s begin to understand why at higher kilobytes per second we will usually have better audio quality.

For this it is necessary to use some examples. But first we need to understand that the greater amount of kilo bytes per second means a greater amount of information per second.

Many will ask And why more information per second synonymous with better audio or video quality?

For that it is important to keep in mind that audio or video files are capturing information and this information is usually very rich in data. For example, the amount of data per second in the performance of a musical group with five or six instruments is quite a lot. Or say the information per second in an image What is very many. So if we lower the number of bytes per second we are reducing the amount of information which impoverish our audio or video file.

The war of volume

For some years now, music recording companies have detected that people listen as a synonym for quality if there is a greater volume And then they have opted for the strategy of increasing the volume of the music they record a little more and produce.

If we had a graphic that will show us the volume and loudness that music used to have in the 70s and we were comparing by decade we could see that the loudness and volume level and volume gain have been increasing decade after decade.

This as I mentioned produces a deceptive effect of perception in the human being that confuses an increase in volume with an increase in audio quality.

And this has been called the war of volume because as we mentioned they have gradually increased the volume level of musical productions to make it appear that they have a higher sound quality.

And how does this compare to bytes per second? As it happens that the amount of information per second does really determine a higher quality and does not need an artificial increase in volume to appear to have a higher quality of digital audio.

So a modern digital audio normalization like the one offered by mp4gain is not misleading, but tries to ensure that each musical passage and each instrument have their optimum volume so that the loudness is constant and so that the quality is the best possible.

Audio normalization

The concern to achieve an adequate sound in the music and a correct balance of the volume levels is something that actually takes centuries.

As soon as the first string quartets or the first symphony orchestras were formed, it was detected that there was a need to maintain an adequate balance of the volumes of each instrument section.
In fact, it is not accidental with a symphony orchestra to have a certain amount of each of the instruments, nor is it accidental that they are accommodated in the way they are accommodated when they are going to play a symphonic concert.

The reason for this is to achieve an adequate balance to the sound of the complete musical piece and it is something that has been thoroughly studied for many centuries.
That is, the human being many years ago of text, what was important to take care of the loudness of each part of a musical piece And of course the global musical work.

Hence we have become accustomed to musical dynamics, What is it that music has more intense moments and softer moments but must be audible all the time, it must be something that is not taken lightly.

Great masters of classical music paid a lot of attention and that is how it was achieved that symphonic orchestras were formed from the amount of instruments that are formed and that were accommodated in a specific way in order to achieve adequate volume management.

This has generated that our ear is used to hearing an adequate musical balance with the correct audio levels and when that is not so we detect that there is a problem and we seek to solve it.

With modern technology and the emergence of digitized music it has been very noticeable for all people that it has become an almost indispensable necessity to normalize the audio.

Normalizing the audio means achieving a standardization where the highest and lowest levels are maintained within a certain standard or certain limits so that it is audible and pleasant for us to listen to all the musical parts that make up a song.

The normalization of the audio has evolved since the music that at first was simply looking for the highest volume peaks was digitized and now in programs such as mp4gain it has achieved very important sophistication that resembles a compression mix, with the use of limiters, and an adequate management of the volume gain of each of the instruments and sounds that form a song.

In other words, we are facing much more modern and advanced algorithms that manage to normalize the audio of music files and videos, even being able to convert from one format to another without problem and normalizing it at the same time. Even The Advance are modern What can perfectly modify the speed of the audio without affecting the Pitch and vice versa.

If you need to achieve the normalization of the audio of a video in the most popular formats of an audio file in the most popular formats the solution is mp4gain which will allow you to easily convert to the format you want and normalize in the most efficient way that exists in the moment your audio or your favorite video.