Should You Normalize Audio? (When To & Not To Normalize)

Normalizing audio is a fairly simple concept, but its applications are not always fully understood. Is normalization always necessary, never necessary, or only applicable to certain situations? Let's find out.

In this article, we'll discuss audio normalization and the two types of normalization. We'll consider the pros and cons as well as the typical and effective applications of this process.

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What Is Normalization?

Normalization is the process of adding or subtracting a certain amount of gain or amplification to bring an audio recording to a target level, otherwise known as the “norm”. It's a pretty simple concept to grasp.

Proper normalizing does not affect the dynamic range of the audio; it simply adds or subtracts gain from the audio to make it louder or quieter, respectively.

Normalization, as an idea, can be applied to analog audio. However, as a process, it is distinctly done to digital audio, which has easy-to-read information (by digital systems) and clearly defined limits.

Normalization became common practice when digital audio workstations began dominating the recording industry in the 1990s. Today, normalization is often regarded negatively in the audio world, losing ground against other, less invasive techniques.

When used wisely, it can be a great ally in audio editing, mixing, and making audio more consistent. It has applications in music, television, broadcasting, podcasting and more. Furthermore, doing loudness normalization to dialogues and podcasts can enhance their perceived quality considerably.

That all being said, normalization is done in one of two ways:

• Peak normalization
• Loudness normalization

Peak Normalization

The first method, commonly known as peak normalization, is not a complex process but rather a linear one. It is achieved by raising the overall level of the audio such that the highest peak in the waveform is brought to the maximum peak level (0 dBFS).

By applying the same amount of gain across the board, dynamics are respected, and you get a waveform close to the original, only louder (or quieter).

The peak normalization process effectively finds the highest PCM sample value of an audio file and applies gain to, typically, bring the peak up to 0 dBFS (decibels Full-Scale), which is the upper limit of a digital audio system.

Note that this normalization can also be used to bring the audio down and doesn't necessarily have to adjust the peak level to 0 dBFS (though this is the most common).

Note that peak normalization is only concerned with detecting the peak of the audio signal and in no way accounts for the perceived loudness of the audio. This brings us to the next type of normalization.

Loudness Normalization

The second method is called loudness normalization and involves much more complex processing.

Many people choose this second method because of the human perception of loudness. At equal dBFS values (and ultimately sound pressure levels), sustained sounds are perceived to be louder than transient sounds.

For example, let's consider peak normalizing a 2-second clip of a square wave and a 2-second clip of a snare drum hit to 0 dBFS. The square wave, which is sustained, will be perceived as being much louder than the snare hit, even though they'll both be normalized to a peak value of 0 dBFS.

Loudness normalizing, on the other hand, will adjust the levels of the recording to improve perceived loudness. For this, a different measurement called LUFS (Loudness Unit Full Scale) or LKFS (Loudness, K-Weighted, Relative To Full Scale) is employed.

This more complex, advanced procedure considers our natural hearing response/sensitivities. The results are perceived as louder by the human ear.

Although LKFS and LUFS have different names, they are the same. They are both standard loudness measurement units used for audio normalization in broadcast, television, music, and other recordings.

RMS values could also be used to find the “average” level of the audio, though RMS is not directly related to how we perceive sound.

The audible range for human hearing is 20 Hz to 20,000 Hz, though we are more sensitive to certain frequencies (particularly in the 200 Hz to 6,000 Hz range). LUFS/LKFS takes this into account for “perceived loudness,” while RMS values do not.

This process works with EBU R-128 volume detection to find the “average” perceived loudness of an audio file and to adjust the overall perceived loudness accordingly. This normalization process could be used to bring the overall level up or down, depending on the circumstance.

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Normalization Vs. Dynamic Compression

As mentioned previously, dynamic range compression and normalization are similar but different. It's a common mistake to confuse normalization with dynamic compression. Yet, there is a big difference between these processes.

Dynamic range compression is the process of reducing the dynamic range of an audio signal (the difference in amplitude between the highest and lowest points).

Compression does so by attenuating the signal amplitude above a set threshold point and providing makeup gain to compensate for lost levels.

With normalization, as we've discussed, the amount of gain applied to the entire recording is consistent; hence, the dynamic range is preserved. This means the resulting audio is the same as the original, just louder (or quieter).

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The Pros & Cons Of Audio Normalization

Now that we understand what normalization is, let's discuss the benefits and drawbacks of audio normalization.

Audio Normalization Pros

Audio normalization has a few pros worth mentioning:

Volume Consistency

The first pro is common practice: level-out audio recorded in different conditions and places. For example, to level the differences in tracks from a recording session.

Of course, there are other (and in my opinion, better) ways of doing this, but normalization can help here as well.

Avoid Peaks Above 0 dBFS

At high enough sample rates and bit depths, normalizing a track to reach maximum loudness will keep the audio at or below the digital maximum, thereby avoiding digital clipping/distortion. Note that limiters also offer the same result, though, like compression, they do so by affecting the dynamic range of the audio.

No Dynamic Alterations

Normalization doesn't alter the dynamic range of the signal, it simply turns it up or down to a specified level. Therefore, it's a good way to help level tracks to more consistent levels without changing their sound.

Audio Normalization Cons

Here are the cons of audio normalization:

Potential For Destructive Processing

If you happen to normalize a signal at 16 bits using truncation, despite most DAWs being able to handle at 32-bit float, there will likely be some destruction on the resulting processed signal, particularly near the peaks.

This can happen even with dithering (the process of adding a small amount of random noise to a digital audio signal in order to reduce the distortion caused by quantization error, particularly when converting from 24 or 32-bit down to 16-bits).

If you happen to discard or lose the original, un-processed signal, you may not be able to undo the normalization process and be left with whatever the result was.

Potential For Inter-Sample Clipping At Peaks

If the sample rate is too low, the 0 dBFS may actually be exceeded when normalizing during the bounce down. As the name suggests, this clipping happens as the digital audio is reconstructed as analog audio and the resulting continuous waveform clips between two or more digital samples that were too close to the 0 dB limit.

Potential For Losing Information When Normalizing To Lower Levels

If we choose to normalize to a lower level than the original audio, we risk losing some amount of digital data as the possible digital dynamic range is reduced.

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When To Normalize Audio

So, normalizing a recording does have pros and cons, but when is it better to use audio normalization? Personally, I never normalize audio, but that's just me. Let's take a look at some scenarios.

Importing Multitracks Into A Mix Template

Normalizing audio tracks to specific levels can be useful when you're using a mix template across multiple mixes, especially if the songs make up an album. It can help get more consistency across your mixes and lead to a more holistic final result across the album.

This could be considered part of your gain staging process.

Samples

Normalizing audio samples grants you that, when mixing, you won't have to touch levels to achieve consistency. So, normalizing each sample before they go into the mix is a great idea.

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When To Avoid Normalizing Audio

While I never use normalization in my workflow, I'm not necessarily against it. However, here are a few cases when I'd definitely recommend not normalizing your audio.

Bouncing Your Mix

A common mistake is to normalize when bouncing. As discussed previously, inter-sample peaking may occur, which may produce distortion and artifacts in the final track that weren't there in the session.

Having said that, I could see a case for normalizing our mixes to a specific level for mastering, so long as the normalization process doesn't alter the audio. However, it's much better, in my opinion, to mix to a certain level, even if it means utilizing gain staging on the mix bus, in order to achieve the desired pre-master mix levels.

Master Album Assembly

When putting together a collection of recordings, loudness normalization can be used to ensure there are no volume jumps from track to track.

However, when it comes to mastering multiple tracks into an album, I wouldn't ever recommend using normalization to accomplish consistent levels across the tracks. Rather, compression, gain staging and a mastering limiter will yield much better results.

Evening Out Differences In A Dynamic Recording

You might be tempted to normalize vocal and other dynamic sources, chopping them down to its bits and using your ears as a guide. Performing normalization only on the low-volume sections will bring up the volume only where you think it's lacking and create a more consistent take.

However, volume automation and dynamics processing like compression and limiting are far more common for a good reason: they work much better. Of course, we could opt for normalization if we wanted, but it would be a waste of time, in my opinion.

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Related Questions

What is the difference between audio compression and limiting? Dynamic range compression and limiting both work on the same principle of reducing an audio signal's dynamic range. Limiting is compression with a very high (often infinite) ratio. Compressors reduce signal levels above a set threshold by a ratio. Limiters are designed to set a maximum output level.

What is upward compression? Upward compression is a type of dynamic range compression that boosts the amplitude of an audio signal below a certain threshold while maintaining the amplitude above the threshold. Upward compression is available in digital plugins and via parallel compression with hardware or software.