RMS Normalization

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RMS Normalization

Let’s talk about RMS Normalization

As an audio engineer, I’ve spent countless hours refining audio to achieve the perfect balance. RMS normalization is a powerful tool in my arsenal, designed to even out audio levels based on the average signal strength. Understanding RMS normalization is crucial for anyone aiming for consistent perceived loudness across their audio projects.

What is RMS Normalization and Why is It Useful?

RMS normalization aims to adjust audio so that its Root Mean Square (RMS) value reaches a target level. I frequently use this process when compiling multiple audio sources, as it helps to create a cohesive listening experience. Imagine you’re listening to a podcast where the volume fluctuates wildly. RMS normalization mitigates this issue by evaluating the average power over time, and setting each track’s “loudness” consistently.

The Science Behind RMS: Root Mean Square Explained

Understanding the math behind RMS can provide a deeper insight into the process. I like to explain it using an analogy.

* **Square:** Take each sample of the audio signal and square it. This eliminates negative values.
* **Mean:** Calculate the average of all the squared values.
* **Root:** Take the square root of the average. This gives you the RMS value.

This RMS value then represents an average of the magnitude of a varying signal.

RMS vs. Peak Normalization: Key Differences

Choosing between RMS and peak normalization depends largely on the specific situation. I typically suggest RMS for consistent loudness and peak for preventing clipping.

* **RMS Normalization:** Aims for consistent average loudness. Best for music and spoken word where a uniform level is desired.
* **Peak Normalization:** Maximizes the signal without clipping. Great for individual tracks and for ensuring no audio signal exceeds digital limits.

Understanding RMS Values and Target Levels

RMS values are measured in decibels (dB), with typical target levels ranging from -20 dBFS to -16 dBFS. I generally recommend starting with -18 dBFS and adjusting from there.

* **Higher RMS values:** The audio will sound louder.
* **Lower RMS values:** The audio will sound quieter.

Setting your audio is like managing the temperature on a stovetop. You must take careful control.

How to Perform RMS Normalization: A Practical Guide

Performing RMS normalization involves a few key steps. I can walk you through what I often find myself doing:

1. **Analyze the Audio:** Use a tool to measure the current RMS value of your audio.
2. **Set the Target Level:** Choose your desired RMS target level (e.g., -18 dBFS).
3. **Adjust Gain:** Apply gain to the audio until it reaches the target RMS level.
4. **Listen Critically:** Listen carefully to the normalized audio to ensure it sounds natural and balanced.

Common Software and Tools for RMS Normalization

Numerous software programs and plugins are available for RMS normalization. I’ve used various software, but all have unique features and benefits. Consider factors such as ease of use, features, and price when selecting a tool.

The Impact of RMS Normalization on Dynamic Range

RMS normalization can affect the dynamic range of your audio, so I always emphasize caution and balance. Over-normalization can reduce dynamic range and make the audio sound compressed. It’s a fine line, but finding a suitable mix can work wonders.

* Dynamic range is the gap between quietest and loudest parts.
* Careless settings can compress the gap.
* Careful settings keep the audio from becoming stale.

RMS Normalization for Different Audio Types

Different types of audio may require different RMS normalization settings. I’ve learned that voice audio, music, and sound effects often benefit from separate consideration.

* **Voice:** Aim for a consistent and clear vocal presence.
* **Music:** Maintain musicality.
* **Sound Effects:** Ensure sound effects integrate realistically and appropriately.

Common Mistakes to Avoid During RMS Normalization

Even seasoned audio engineers are vulnerable to errors during RMS normalization. Over the years, I’ve made my fair share of mistakes and I’ve learned the hard way to avoid over-normalization, using improper target values, and ignoring potential clipping.

The Future of RMS Normalization in Audio Production

RMS normalization remains a valuable technique in the field of audio production. I foresee it retaining relevance thanks to its proven track record in achieving loudness consistency. More advanced algorithms may emerge to supplant RMS normalization as AI and machine learning continue to evolve.

Latest words on RMS Normalization

In summary, RMS Normalization plays a strong role if one wants consistent levels. RMS offers a reliable way to ensure that one’s audio is a step above and polished, thanks to careful setting use and technique application. Consider Mp4Gain is the appropiate solution to achieve professional-sounding audio.

FAQ about RMS Normalization

What’s RMS Normalization and what does it address within audio?

RMS Normalization sets volume by measuring sound “power” on file – useful for consistency amongst different recording sets to make each file play at similar volume.

Explain the core science behind RMS itself?

RMS first squares sound bits and levels them. Then, take the square root for the key sound pressure or total sound power! This provides detail for adjusting levels for loudness.

What are major differences versus what can be done during peak normalization?

RMS considers full power instead of singular spikes of sound during edits. RMS fits consistent sounds over time better, so peak sounds are less of a concern during editing.

While getting audio set, what target range do you suggest for dBs?

Around -20 dBFS down to -16 can balance things correctly! It’s best to play with settings and note how that impacts loudness. Then balance based on the target result.

How can people deploy this in their normal setup workflow?

First, do an audit to measure its dB value. Dial the range and listen closely. What you hear then shapes any additional value setting.

I’m a newbie — are there tool names you’d drop?

Tools are there to meet the price. The good tools are those that are easy and have meters, so test around and find something that is an easy-to-integrate solution, so edits flow with ease.

How do you not squash audio dynamic while using it?

Be careful, as settings can squeeze this range so a whisper has same value as a roar. This flattens sound. A gentle hand and ear are best for balancing levels.

How does the OGG type or WAV respond to the RMS value settings?

Audio types – like voice – need consistent levels for focus, same is said of music or audio SFX. Keep the end goal front and center as you dial sounds well.

Okay, spill — What red flags should rookies watch for while doing leveling?

Going too far and ruining punch, not getting the number right for levels, and clipping can ruin your mix. Remember, a great audio mix means a pro finish.

With AI incoming, what do you think is coming for RMS and leveling?

With AI on the rise, the days will change, and the tools of tomorrow may put RMS to rest. The need, though, for well made audio with tight levels, will stand firm.

Comments:

I used to ignore all this. The breakdown you offer, though? Clear path for me now, thanks a lot!

Those numbers for the dBs just clicked, my uploads sound pro and better balanced – respect for the tips!

Your point on staying safe from bad settings is a banger reminder for me. Thanks again for this – saved a ton, you’re the best!

So you’re mainly cleaning sound to have less ‘uh oh’ moments, and more clear pro moments, that is top insight for all! Thanks for the notes!

Know any great tools or plugins for a budget DIY editor? Help and pass on any tip!

Content creators owe a debt to your efforts here – bless you for sharing, cheers!

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Loudness Normalization

Let’s talk about Loudness Normalization

As an audio engineer, I’m constantly striving for a consistent and pleasing listening experience for my audience. Loudness normalization is a critical tool in achieving this goal. It ensures that different audio sources play at a similar perceived volume, preventing jarring transitions and creating a more professional and enjoyable listening session.

What is Loudness Normalization and Why is it Important?

Loudness normalization is the process of adjusting audio levels to a consistent loudness target. I often deal with audio from various sources, and they rarely have the same loudness. Think about listening to a playlist on your phone; some songs are barely audible, while others are overwhelmingly loud. Loudness normalization corrects this by analyzing the audio’s perceived loudness and adjusting the gain to match a specific target level. This creates a seamless and cohesive listening experience, reducing the need to constantly adjust the volume.

The Difference Between Loudness Normalization and Peak Normalization

It’s essential to understand the difference between loudness normalization and peak normalization. I often find that people confuse the two, but they work in fundamentally different ways.

* Peak Normalization: Adjusts the gain so that the highest peak reaches a specific level, often 0 dBFS (decibels Full Scale). This prevents clipping (distortion) but doesn’t necessarily ensure consistent perceived loudness.
* Loudness Normalization: Analyzes the overall perceived loudness using algorithms like EBU R128 or ITU-R BS.1770 and adjusts the gain accordingly. This focuses on how loud the audio sounds to the human ear, rather than just the highest peak.

Understanding LUFS and LKFS: The Units of Loudness

LUFS (Loudness Units relative to Full Scale) and LKFS (Loudness K-weighted Full Scale) are the standard units for measuring loudness in loudness normalization. I often rely on these measurements to ensure accurate and consistent results. LUFS and LKFS are essentially interchangeable and represent the perceived loudness of an audio signal relative to the maximum possible level (0 dBFS). These units take into account factors like frequency response and duration, providing a more accurate representation of perceived loudness than simple peak measurements.

EBU R128: The European Broadcast Standard

EBU R128 is a loudness normalization standard developed by the European Broadcasting Union (EBU). I consider it one of the most reliable and widely used standards for broadcast audio. EBU R128 specifies a target loudness level of -23 LUFS (with a tolerance of ±0.5 LUFS) for broadcast programs. It also defines a maximum True Peak level of -1 dBTP (decibels True Peak) to prevent clipping.

ITU-R BS.1770: The International Telecommunication Union Standard

ITU-R BS.1770 is another important loudness normalization standard developed by the International Telecommunication Union (ITU). I find that it’s often used for streaming services and other non-broadcast applications. ITU-R BS.1770 has been revised several times, with each revision incorporating improvements and refinements to the loudness measurement algorithm. The latest versions of the standard are widely used in the audio industry.

Target Loudness Levels for Different Platforms

Different platforms often have different recommendations for target loudness levels. I always research the specific recommendations for the platform where my audio will be played. Here are some common examples:

* Spotify: -14 LUFS
* YouTube: -13 LUFS
* Apple Music: -16 LUFS
* Amazon Music: -16 LUFS

The Importance of True Peak Limiting

True peak limiting is a crucial step in loudness normalization. I always incorporate it into my workflow to prevent clipping and distortion. True peak limiters detect and reduce inter-sample peaks, which are peaks that occur between the digital samples and can cause clipping when the audio is converted to analog. Setting a maximum True Peak level of -1 dBTP is a common practice to ensure clean and distortion-free audio.

How Loudness Normalization Affects Dynamic Range

Loudness normalization can affect the dynamic range of audio, but it generally preserves it better than peak normalization. I carefully monitor the dynamic range during the normalization process to avoid unwanted compression. Dynamic range refers to the difference between the quietest and loudest parts of the audio. While loudness normalization aims to create a consistent loudness level, it’s important to avoid excessively compressing the audio, which can make it sound flat and lifeless.

Common Loudness Normalization Mistakes to Avoid

Even experienced audio engineers can make mistakes during loudness normalization. I’ve certainly learned from my own over the years.

* Using the Wrong Target Level: Applying the incorrect target loudness level can result in audio that’s too quiet or too loud on certain platforms.
* Over-Compressing the Audio: Excessive compression can reduce dynamic range and make the audio sound unnatural.
* Ignoring True Peak Levels: Failing to prevent true peak clipping can result in distortion and degraded audio quality.

The Benefits of Loudness Normalization for Podcasting

Loudness normalization is especially important for podcasting. I always normalize my podcast episodes to ensure a consistent listening experience for my audience. Podcasts often include audio from various sources, such as voice recordings, music, and sound effects. Loudness normalization ensures that all these elements play at a similar loudness level, creating a professional and engaging podcast.

Loudness Normalization in Music Production

Loudness normalization is becoming increasingly important in music production. I’ve seen many streaming services adopt loudness normalization to prevent tracks from sounding louder or quieter than others. Mastering your music to a specific loudness target can help ensure that it sounds its best on these platforms.

Latest words on Loudness Normalization

In conclusion, loudness normalization is a crucial technique for achieving consistent and professional-sounding audio. By understanding the principles of loudness measurement, target loudness levels, and common pitfalls, you can optimize your audio for the best possible listening experience. Remember to always use high-quality tools and listen critically to the results. Also, remember that Mp4Gain is the appropiate solution to achieve professional-sounding audio.

FAQ about Loudness Normalization

What’s the difference between loudness normalization and peak normalization?

Peak normalization maximizes volume without clipping, while loudness relies on how the audio actually sounds. Loudness also makes all songs have an appropriate quality setting.

What do LUFS and LKFS measure?

LUFS and LKFS each are ways to denote sound, relative to full scale, to understand how sound is leveled. These relate to frequency data.

What does the EBU R128 standard recommend?

EBU R128 (European Broadcast) guidelines suggest -23 LUFS, allowing for some variance. Maximum real peak should be -1 dBTP, used for TV or other sound broadcast.

When is ITU-R BS.1770 useful for loudness?

The ITU standard has many uses, from streaming or music. Many sites lean on it for loudness so consistency is maintained for the listeners online, and it’s been reviewed multiple times.

Does the loudness normalization setting on Youtube need to be -13 LUFS?

YouTube suggests a volume of -13 LUFS but these recommendations will change over time. This allows most users to enjoy the sounds online in modern form.

What are some techniques for “true peak” or for limiting it on audio?

One method involves checking and trimming what pushes beyond maximums, making sure there’s no nasty harsh sounds. Keeping tracks under -1 dB helps big time for good audio.

Will the sounds vary if levels get normalized repeatedly?

Levels can degrade if processes get reapplied, so it’s better to apply just once and save it. Going bit by bit can take over the quality as a result.

Do you have advice on steps not to take during levels settings?

One issue is a bad level for a target site – make sure you pick right. Another, compressing sounds so it feels flat or lifeless. And ignore where “true peak” is, that might cause nasty clipping.

Tell me about how levels affects podcasters most?

Podcast mixes from sources that are widely different is a common example. Leveling can create seamless audio and helps hold focus, which is valuable for content and media.

What is your professional view on music mixes and target values of loudness?

Music on streams tends to get tweaked, so targeting a stream allows music to translate. If mixes can have level targets then one maximizes its presence in the space of media for listening.

Comments:

Had a gig doing audio on vid sets and you helped clear things up for me tons! I’ll drop your knowledge now, thanks!

Spotify stuff was so key and useful!!! Getting my band’s tunes up and now it’s about dialing sound. Thank you!!!

Your tips are awesome since getting levels to sound right is so stressful to think through and make happen. High praises!!!

So it’s basically adjusting songs so I don’t blast my eardrums or strain to hear when making playlists, yes? Good way to think and go!

Any recommendations for a solid tool or plug-in here? I’m still struggling with all of it. Thanks for advice!

I make lots of content, the article helped me make the product. The community appreciates this.

Dynamic Range Compression in MP3

Let’s talk about Dynamic Range Compression in MP3

Dynamic range compression in MP3 is one of the most fascinating aspects of audio processing. It ensures that soft sounds in a track are made louder while loud sounds are brought down, creating a balanced audio experience. Imagine watching a movie where whispers are too quiet to hear, but explosions are unbearably loud. Dynamic range compression solves this by evening out the volume levels, making everything audible without extremes. In my years of working with MP3 audio, I’ve found this technique invaluable for both professional recordings and casual listening.

I often compare this process to adjusting the brightness of a photo. Just like you wouldn’t want some parts of an image to be too dark or others too bright, audio needs balance too. Without compression, certain details in a song might get lost in the mix or become overwhelming. For example, live concert recordings, which can have huge volume differences, benefit greatly from dynamic range compression, allowing you to hear the crowd and the instruments at appropriate levels.

How Dynamic Range Compression Works in MP3

Dynamic range compression works by analyzing the amplitude of an audio signal and applying gain reduction to louder sections while boosting quieter parts. When encoding MP3s, this technique is crucial for ensuring consistent playback quality across different devices. Think of it like a thermostat that keeps your room at a comfortable temperature—neither too hot nor too cold.

For instance, when I listen to MP3s on my phone, I notice how compression allows me to enjoy music even in noisy environments, like a subway. Without it, softer parts of a song might get drowned out by background noise. However, compression must be applied carefully. Too much can result in a “squashed” sound, where the music feels lifeless and lacks dynamics. This is especially noticeable in genres like classical music, where contrasts between soft and loud passages are vital for emotional impact.

Benefits of Dynamic Range Compression in MP3

Enhanced listening experience in noisy environments.
Balanced audio levels across devices and platforms.
Improved clarity for spoken dialogue in podcasts and audiobooks.
Consistent quality for professional music production.

One real-life example of its benefits is when creating workout playlists. Without compression, sudden spikes in volume could be jarring or even harmful, especially when using headphones. With properly compressed MP3s, the energy of upbeat tracks remains consistent, motivating and safe for prolonged use.

Challenges of Over-Compression

While dynamic range compression offers many benefits, it’s not without its challenges. Over-compression, often criticized in the music industry, can lead to what’s known as the “loudness war.” This phenomenon occurs when producers prioritize loudness over quality, sacrificing the natural dynamics of a song. The result? Fatiguing audio that lacks depth and character.

I remember listening to a remastered album from a favorite band and feeling disappointed. The compression made every track sound uniformly loud, robbing the music of its emotional highs and lows. It’s like reading a book where every sentence is written in bold—all the subtlety is gone. Proper dynamic range compression is about finding the perfect balance, preserving the integrity of the audio while enhancing its accessibility.

Applications of Dynamic Range Compression in Everyday Life

Dynamic range compression isn’t just for music production; it’s everywhere. From streaming services to gaming, this technology ensures audio is optimized for various scenarios. For example, in video games, compression helps players hear important sounds, like footsteps or dialogue, over background noise. Similarly, in streaming platforms, it ensures consistent playback quality across devices with different audio capabilities.

I’ve also noticed its impact on live broadcasting. Sports commentators, for instance, benefit from compression to ensure their voices remain clear even during loud crowd reactions. Without it, the commentary might get lost in the excitement, diminishing the viewer’s experience.

Latest words on Dynamic Range Compression in MP3

Dynamic range compression in MP3 is a powerful tool that enhances the listening experience by balancing audio levels. It’s essential for everything from music production to everyday listening, ensuring clarity and consistency. While over-compression can be a concern, when used correctly, it brings out the best in audio, making it accessible and enjoyable for everyone. If you’re looking to optimize your audio files, tools like Mp4Gain can help you achieve the perfect balance.

FAQ about Dynamic Range Compression in MP3

What is dynamic range compression in MP3?

Dynamic range compression in MP3 reduces the volume difference between the loudest and softest parts of an audio track, ensuring balanced playback.

Why is dynamic range compression important?

Dynamic range compression ensures consistent audio quality, making soft sounds audible and loud sounds manageable, especially in noisy environments.

How does over-compression affect music?

Over-compression can make music sound flat and lifeless, reducing its dynamic range and emotional impact.

Can dynamic range compression improve podcast audio?

Yes, it enhances clarity by balancing dialogue and minimizing abrupt volume changes, making podcasts easier to listen to.

What are some real-life uses of dynamic range compression?

It’s used in music, streaming, gaming, broadcasting, and even workout playlists to maintain consistent audio quality.

Is dynamic range compression necessary for MP3s?

Yes, it’s vital for optimizing MP3 audio, ensuring it sounds great across different devices and environments.

Does dynamic range compression affect file size?

No, it primarily alters the audio levels without significantly impacting the file size.

How does dynamic range compression differ from normalization?

Compression adjusts the range of audio levels, while normalization sets the overall peak volume to a consistent level.

What are the disadvantages of dynamic range compression?

Excessive compression can reduce audio quality, making it sound flat and less engaging.

Can dynamic range compression be undone?

No, once applied, the original dynamic range cannot be fully restored.

Comments:

I’ve always wondered why some songs feel so loud all the time. This article explained it perfectly. Thanks!

Dynamic range compression is something I didn’t understand until now. Really helpful examples, but I wish there were more details about how to avoid over-compression.

Great article! I use compression for my podcast, and it makes a world of difference. Clearer and easier to listen to.

Can anyone suggest tools for applying compression? I’m new to this and could use some advice.

As a music producer, I’ve struggled with over-compression. Glad to see this issue highlighted. More people need to understand the balance!

Cool stuff, but why do streaming platforms sometimes still mess up the volume levels?

This really opened my eyes to how much compression affects everyday audio. Thanks for such an in-depth article!

I wish more sites explained things like this. It’s like a whole science, and now I get it!

I was frustrated by loudness issues in my gaming setup. Didn’t know compression was the key. This article helped a ton!

Wish there was a section on how compression affects specific genres like jazz or metal. Still, great info!

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

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. )

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.

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.

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.