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 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.

Radio Studio Compressor

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.

Dynamic range compression

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.

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