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Audio Normalization, understand what it is about

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Audio Normalization, understand what it is about

Difference between Peak level and RMS in Audio

Something that is mentioned a lot, for example when audio recordings are produced, is about the so-called Peak Level and RMS, Peak and RMS (Root Mean Square), which are detected by meters (software, or hardware) But… What are they exactly these values?

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It is important that someone who does not record audio but simply listens to understands these differences.
This will make you a true expert, even if you are just someone who has a good collection of music, but knows how to distinguish who is normalizing and understands the subject.

DIFFERENCES

The Peak value will inform us of all those maximum values that occur in our music in real time. To understand us … If we have, for example, a recorded song where a drummer emphasizes playing the tarola or a cymbal, we will see that our peak meter will show a higher value for a moment, because it is the one that is sounding louder in that instant. This meter will work with fast attack times, to be able to immediately measure these peaks and maybe use a limiter to avoid them.

What is RMS?

The RMS value, however, will mark the average value of the loudness or volume of our music … how does that do it? , for this it will use attack times, much longer longer. To be clearer … This value will give a reference of the energy level or volume (how high or low is the volume that is playing) but will not be affected by the peaks.

When we say that it has a slower attack value, this means that it does not measure variations so quickly, but rather that it is “slow” to react and therefore shows us something that could be an “average” volume level.

In any case, the suitable normalizer must be a mixture of limiter (that device that prevents the music from distorting because it has exceeded the maximum possible level) and a compressor, which is the one that prevents the peaks from exceeding a level and also prevents them from Volume drops drop more than a preset value.

In this way the music always remains within a medium range, without exceeding a limit neither up nor down.

Professionally recorded or broadcast music is always limited and compressed to keep it playing its best within a suitable range.

The only software that does exactly this is the Mp4Gain. That is why it has been accepted not only by amateurs, but by professionals.

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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.

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.

How does a modern mp3 normalization really work and why is it superior?

Actually we will refer to an audio normalization in general, regardless of the format.

-First attempts to normalize the audio

As soon as the digital format emerged and people were able to store audio files on their computers, it was necessary to encode them to save space, without losing quality.

This is how formats such as mp3 emerged, because they managed to make the same song occupy 10 times less space than a wav format, and if it was well coded and / or with a quality encoder, it sounded practically the same. It was gained in space and the loss of quality was practically zero.

For this it was necessary to maintain a samplerate of at least 44,100 kHz and a bitrate of at least 128.

Initially the most accessible format was the mp3 and although later other formats have emerged (even looseless like FLAC), the mp3 continues to take the place of favorite.

But when a person listens one song after another, he will soon perceive differences in the sonority of these songs. It seems that the volume level is not even, and that is when it becomes necessary to normalize, to avoid having to manually correct the audio volume level of each song by turning the volume knob.

Even Spotify has had to incorporate some normalize option to avoid these annoying volume gain level changes between songs.

The difference is that Spotify only normalizes the volume while playing the songs, so if we stored them on our hard drive, the volumes would remain uneven.

-The solution of volume peaks

The first attempts to normalize the audio of an mp3 were inefficient, they only looked for the peak of a song and amplified it all to the point where those peaks did not distort.

But this did not eliminate the perception that one song would sound louder than others.

Over time the volume level normalization algorithms have been refined and have become more subtle and complex. They have been achieving similar results to those obtained by the great sound consoles in musical concerts, where the music is highly processed to – among other things – make the volume levels are even and unchanged, and also achieve a sussuro of The singer’s voice is audivble even if the battery gives tremendous blows in a roll.

Mp4Gain has come to obtain these results, without the need for you to invest thousands of dollars in hardware like these musical groups or large radio stations do.

Today, audio normalization really is a combination of compression, limiters and normalization, which achieves an amazing result. Far from those first attempts to level the volume gain of each song to avoid big differences.

DIFFERENCES BETWEEN NORMALIZE AND MASTERIZE

The process and the differences between normalizing and mastering are often confused. Although it may seem to be the same, it is not.

Mastering can be of crucial importance according to which processes, for example: in musical matters, there are mastering engineers who are dedicated exclusively to that.

That does not mean that we cannot learn or acquire the necessary knowledge to be able to properly use some processing effect or some plugin in an appropriate way to be able to get more out of our audio file.

But you have to keep in mind that this audio processing helps your audio montage, song … sound with more punch, more strength, more energy, have more life.

Is mastering compressed or limited?

Rather those two processes and some more are done.

Its mission is to maintain the same volume amplitude throughout the audio file, that is, it compresses when it has to compress and limits when it has to limit.

I’m going to give a rough example of what manual mastering would be like.

Can you still imagine the sound technician who detects when the signal volume is too high (the singer gets too close to the microphone, shouts …) and lowers the fader. Or the opposite case, when it detects the low volume (the singer moves too far from the microphone, does not speak with enough force …) and raises the fader. Always trying to maintain the same volume amplitude.

I’m going to give you a homemade definition: “lower what is high and raise what is low“.

As before it was an invented example, to do the job of processing the sound we regulate the different parameters available to the “processor” (Mastering is also called “processing” since in the past a device called “processor” was used which comes from “dynamics processor”). These parameters are:

The threshold (threshold): fundamental characteristic of the compressor that represents the point or level from which if the volume of the sound exceeds or lowers it, the dynamics processor is put into operation.

Ratio (Attenuation or Gain Ratio): Defines the amount of attenuation or gain that is applied to the signal. At noise gates the attenuation can be preset so that it really is a mute.

Attack time: This is the time it takes for the signal to attenuate, limit, mute or amplify. In general, slower times work best at low frequencies and fast ones at high frequencies. When processing a signal containing all frequencies, a compromise situation is forced.

To maximize the energy of the signals, particularly in broadcasting applications, there are multiband compressors that divide the spectrum into several bands and apply different times to each.

Release time: It is the opposite of the attack time, that is, the time it takes to go from the state where the processing is running to rest. They are usually longer times than those of attack.

Hold (maintenance time): Specifies the minimum time that processing will take place.

Stereo link (stereo link): With dynamics processors in general when used to process a two-channel (stereo) signal, it is necessary to link the processing action of both channels to happen on both at the same time. Otherwise, the sound image will be confusing and changing from the center to one side or the other.

Automatic: This function allows you to control any of the parameters listed automatically depending on the characteristics of the signal.

By pass (deactivation): Activating it allows you to hear the unprocessed signal, while if it is not activated you hear the processed signal.

Normalization is a process by which the highest peak is sought and reduced or increased (dB) as adjusted. Never pass the 0dB in normalization or mastering, because then it would be itching “clipping”.