As an audio optimization expert, I’m often asked about the best techniques for normalizing audio levels. In this article, I will explore two popular approaches: peak normalization and loudness normalization. These techniques, peak vs. loudness normalization, have their own unique advantages and considerations. Let’s dive in and uncover the secrets of achieving balanced and consistent audio!
Peak Normalization: Unleashing the Power of Dynamics
When it comes to peak normalization, it’s all about preserving the dynamics of your audio. Imagine a breathtaking symphony where the crescendos and diminuendos transport you to a different realm. With peak normalization, you ensure that the highest peaks of your audio reach their full potential without clipping or distortion. It’s like giving your audio the freedom to express itself with intensity and impact.
Loudness Normalization: The Harmony of Consistency
Now, let’s turn our attention to the world of loudness normalization. Have you ever experienced the frustration of constantly adjusting the volume while switching between songs or TV shows? Loudness normalization comes to the rescue! By analyzing the perceived loudness of your audio, it ensures a consistent listening experience across different tracks. Say goodbye to sudden volume jumps and immerse yourself in a harmonious soundscape.
Dynamic Range: The Dance of Soft and Loud
In the realm of audio normalization, we encounter the concept of dynamic range. Dynamic range represents the difference between the softest and loudest parts of an audio signal. Peak normalization respects the natural dynamic range, allowing the delicate whispers and thunderous roars to coexist in perfect balance. On the other hand, loudness normalization aims to reduce the dynamic range, providing a more even playing field for all elements of your audio.
Audio Clipping: Taming the Wild Peaks
Audio clipping is a notorious villain that can ruin your audio experience. Picture this: a sudden burst of sound that distorts and crackles, disrupting your enjoyment. Peak normalization acts as the hero in this story, taming those wild peaks and ensuring that your audio stays within safe limits. With peak normalization, your audio remains clean and free from the dreaded clipping monster.
LUFS: The Measure of Perceived Loudness
In the realm of loudness normalization, we encounter the term LUFS, which stands for Loudness Units Full Scale. LUFS provides a standardized measure of the perceived loudness of your audio. Loudness normalization algorithms analyze the integrated LUFS value and adjust the overall volume to match a specific target level. It’s like having a universal translator that ensures consistent loudness across different tracks and platforms.
Listening Environment: From Living Rooms to Concert Halls
Let’s talk about the listening environment and its impact on audio normalization. Every space has its unique characteristics, from the cozy intimacy of a living room to the grandeur of a concert hall. Loudness normalization takes into account these variations, delivering a consistent listening experience regardless of the environment. So whether you’re enjoying your favorite tunes at home or attending a live performance, the magic of normalization will make every moment memorable.
Personal Preference: Customizing Your Audio Journey
We all have our individual tastes and preferences when it comes to audio. Some crave the raw power of peak normalization, while others seek the comfort of consistent loudness through loudness normalization. The beauty of audio normalization techniques is that they allow you to customize your audio journey according to your personal taste. It’s like having a tailor-made suit that perfectly fits your unique style.
Metadata and Replay Gain: Enhancing the User Experience
Metadata and Replay Gain are powerful allies in the realm of audio normalization. Metadata provides valuable information about your audio, guiding normalization algorithms to make the right adjustments. Replay Gain takes it a step further by applying metadata tags to your audio files, ensuring consistent playback volume across different tracks. Together, they create a seamless and enhanced user experience, elevating your audio enjoyment to new heights.
Compression: Controlling the Sonic Landscape
Dynamic audio content, such as movies or live performances, often presents challenges for normalization. This is where compression enters the scene. Compression techniques allow you to shape the sonic landscape, reducing the dynamic range while maintaining audio quality. It’s like having a skilled conductor who ensures that every instrument is heard clearly, regardless of its volume.
Audio Editing and Mastering: Polishing the Gems
Lastly, let’s not forget the crucial role of audio editing and mastering in the pursuit of sonic perfection. Audio professionals meticulously fine-tune various parameters during the editing and mastering process. Audio normalization techniques become valuable tools in their arsenal, ensuring that the final product shines with balanced and consistent audio. It’s like adding the final touch of brilliance to your audio gems.
In conclusion, the choice between peak normalization and loudness normalization depends on your desired audio outcome. Whether you embrace the dynamic range or seek consistent loudness, these techniques empower you to create an audio experience that resonates with your vision. So go forth, unleash the power of normalization, and let your audio journey be a harmonious symphony of sound!
RMS Normalization: Understanding and Applying the Technique
RMS NormalizationRMS Normalization
What is RMS Normalization?
RMS normalization is a technique used in audio engineering to adjust the volume of a sound signal. It is a type of gain staging that measures the Root Mean Square (RMS) value of the audio signal and adjusts it to a desired level. The purpose of RMS normalization is to make sure that the volume of a sound signal is consistent and compatible with other audio files in a mix.
How Does RMS Normalization Work?
RMS normalization involves calculating the RMS value of an audio signal and then adjusting its gain to match a target RMS level. The RMS value is the average energy of the signal over a certain period of time. To calculate the RMS value, the audio signal is squared, then averaged over time, and then the square root of that average is taken.
Once the RMS value of the audio signal has been determined, the gain is adjusted so that the signal’s RMS value matches the desired target level. This can be done using software tools that are available in most Digital Audio Workstations (DAWs).
Why is RMS Normalization Important?
RMS normalization is important in audio engineering because it ensures that the volume of a sound signal is consistent and compatible with other audio files in a mix. In a mix, audio signals that are too loud can cause distortion, while audio signals that are too quiet can get lost in the mix. By using RMS normalization, audio engineers can make sure that each audio file is at a consistent level, which makes it easier to create a balanced and well-mixed track.
How is RMS Normalization Different from Peak Normalization?
Peak normalization is another type of gain staging that is commonly used in audio engineering. Peak normalization adjusts the gain of an audio signal so that its highest point (or peak) matches a certain level. While peak normalization can be useful for preventing clipping and ensuring that the loudest parts of a signal don’t exceed a certain level, it does not take into account the overall energy level of the signal.
RMS normalization, on the other hand, measures the average energy of the signal over time, which is a more accurate representation of the signal’s perceived loudness. RMS normalization is a more useful tool for ensuring that the overall volume of a signal is consistent and compatible with other audio files in a mix.
What are the Pros and Cons of RMS Normalization?
Pros:
Ensures consistent volume levels across a mix
Provides a more accurate representation of perceived loudness
Can improve the clarity and balance of a mix
Cons:
Can reduce the dynamic range of a signal if overused
May not be suitable for all types of audio signals
Can introduce unwanted artifacts or noise if not used correctly
How to Apply RMS Normalization in Practice?
To apply RMS normalization in practice, follow these steps:
Import the audio file into your DAW
Identify the target RMS level for your mix
Measure the RMS value of the audio file using a metering tool in your DAW
Adjust the gain of the audio file so that its RMS value matches the target level
Repeat steps 3 and 4 for all audio files in your mix
Adjust the overall volume of the mix to achieve a balanced and cohesive sound
When to Use RMS Normalization
RMS normalization should be used in situations where consistent volume levels are important, such as in music production, film and video post-production, and podcast editing. It can be especially useful when working with multiple audio files that were recorded at different levels.
RMS normalization can also be useful when mastering a track for release. In mastering, the goal is to create a final version of the mix that sounds great on a variety of playback systems, such as speakers and headphones. By using RMS normalization to ensure consistent volume levels across the mix, the mastering engineer can create a cohesive and balanced final product.
However, it is important to note that RMS normalization should not be used as a fix for poorly recorded audio. If an audio file has significant issues with noise, distortion, or clipping, it should be addressed at the source rather than relying on normalization to fix the problem.
Can RMS Normalization Improve the Sound Quality of a Mix?
While RMS normalization can help create a more balanced and consistent mix, it is not a magic bullet for improving sound quality. The quality of a mix depends on many factors, including the quality of the source material, the arrangement and processing of the tracks, and the skill of the mixer.
That being said, using RMS normalization as part of the mixing and mastering process can certainly improve the sound quality of a mix. By ensuring consistent volume levels and reducing the risk of distortion, RMS normalization can help create a clearer, more balanced mix that is easier to listen to.
Are There Any Alternatives to RMS Normalization?
Yes, there are several alternatives to RMS normalization that can be used in audio engineering. One alternative is manual gain staging, where the gain of each track is adjusted by ear to achieve a balanced mix. This method requires more time and attention to detail, but it can result in a more natural and dynamic sound.
Another alternative is using a compressor or limiter to control the dynamic range of a signal. A compressor reduces the volume of loud parts of a signal, while a limiter prevents the signal from exceeding a certain level. While these tools can be useful for controlling the dynamic range of a signal, they do not provide the same level of consistency and accuracy as RMS normalization.
How Does RMS Normalization Affect the Dynamic Range of a Signal?
RMS normalization can affect the dynamic range of a signal if it is overused. Dynamic range refers to the difference between the loudest and quietest parts of a signal. If a signal is heavily compressed or normalized, the dynamic range can be reduced, which can make the signal sound less natural and more compressed.
To avoid reducing the dynamic range of a signal, it is important to use RMS normalization judiciously and in conjunction with other gain staging techniques, such as manual gain staging or the use of compressors and limiters.
Can RMS Normalization Cause Clipping?
RMS normalization itself does not cause clipping, but it can exacerbate clipping that already exists in an audio file. Clipping occurs when the volume of a signal exceeds the maximum level that can be recorded or processed without distortion. If an audio file has clipping, normalizing it using RMS normalization can cause the clipped parts of the signal to become even louder, which can make the distortion more noticeable.
To avoid clipping, it is important to monitor the levels of each track during recording and to use proper gain staging techniques during mixing and mastering. If clipping is present in an audio file, it should be addressed at the source rather than relying on normalization to fix the problem.
What is the Difference Between RMS and Peak Level?
RMS level measures the average energy of a signal over time, while peak level measures the highest instantaneous level of a signal. In audio engineering, RMS level is often used to measure the perceived loudness of a signal, while peak level is used to measure the potential for clipping and distortion.
RMS normalization adjusts the volume of a signal based on its RMS level, while peak normalization adjusts the volume based on its peak level. RMS normalization is generally considered to be a more accurate method of normalization, as it takes into account the overall energy of the signal rather than just its peak levels.
How Do You Perform RMS Normalization?
To perform RMS normalization, you will need audio editing software that supports the feature. Here are the basic steps:
Open the audio file you want to normalize in your audio editing software.
Select the portion of the audio file you want to normalize, or the entire file if you want to normalize the entire track.
Find the RMS normalization function in your software. It may be called something like “normalize to RMS” or “normalize by loudness.”
Select the desired target RMS level. This will vary depending on your specific needs, but a common target level for music production is -16 dBFS.
Apply the normalization and listen to the results. If the results are satisfactory, save the file. If not, adjust the target RMS level and try again.
What Are Some Best Practices for Using RMS Normalization?
Here are some best practices to keep in mind when using RMS normalization:
Use RMS normalization as part of a comprehensive gain staging strategy that includes manual gain staging and the use of compressors and limiters.
Avoid using RMS normalization as a fix for poorly recorded audio. Address any issues with noise, distortion, or clipping at the source.
Be careful not to overuse RMS normalization, as it can reduce the dynamic range and naturalness of a signal.
When applying RMS normalization to a mix, make sure to normalize all tracks to the same RMS level to ensure consistency.
Listen to the results of RMS normalization carefully and make adjustments as necessary to achieve the desired sound.
What Are Some Common Misconceptions About RMS Normalization?
Here are some common misconceptions about RMS normalization:
That it can fix poorly recorded audio. RMS normalization can help balance and normalize the volume levels of a signal, but it cannot fix issues with noise, distortion, or clipping that are present at the source.
That it can automatically improve the sound quality of a mix. While RMS normalization can help create a more balanced and consistent mix, it is not a magic bullet for improving sound quality. The quality of a mix depends on many factors beyond volume levels.
That it can replace proper gain staging techniques. While RMS normalization can be a useful tool in the gain staging process, it should be used in conjunction with other techniques, such as manual gain staging and the use of compressors and limiters.
How Does RMS Normalization Differ from LUFS Normalization?
RMS normalization and LUFS (Loudness Units Full Scale) normalization are both methods of adjusting the volume levels of audio signals, but they differ in their approach and application.
RMS normalization adjusts the volume of a signal based on its RMS level, while LUFS normalization adjusts the volume based on its perceived loudness. LUFS normalization takes into account the characteristics of the human ear and how it perceives loudness, while RMS normalization is based purely on the energy of the signal.
FS normalization is often used in broadcast and streaming environments, where it is important to maintain a consistent perceived loudness across multiple programs or pieces of content.
While RMS normalization is useful for balancing the volume levels of individual tracks or sections of a mix, LUFS normalization is better suited for ensuring consistency and compliance with loudness standards.
Frequently Asked Questions About RMS Normalization
1. Can RMS normalization be used on any type of audio file?
Yes, RMS normalization can be used on any type of audio file that can be opened in your audio editing software.
Audio normalization is the process of adjusting the loudness of an audio recording to a standard level. The goal is to ensure that all audio files have a consistent volume, making them easier to listen to and preventing ear fatigue. In this article, we will explore the different types of audio normalization and how they work.
Audio Normalization
Peak Normalization
Peak normalization is the process of adjusting the peak amplitude of an audio recording to a certain level. The peak amplitude is the highest point in the audio signal, and it is measured in decibels (dB). The goal of peak normalization is to ensure that all audio files have the same peak amplitude, making them easier to listen to and preventing ear fatigue.
Peak normalization is typically used for digital audio files, such as MP3 and WAV files. These files are usually stored in a digital format that allows for easy manipulation of the audio data. However, peak normalization can also be applied to analog audio recordings, such as cassette tapes or vinyl records.
RMS Normalization
RMS normalization is the process of adjusting the root mean square (RMS) level of an audio recording to a certain level. The RMS level is a measure of the average power of an audio signal, and it is measured in decibels (dB). The goal of RMS normalization is to ensure that all audio files have the same RMS level, making them easier to listen to and preventing ear fatigue.
RMS normalization is typically used for digital audio files, such as MP3 and WAV files. However, it can also be applied to analog audio recordings, such as cassette tapes or vinyl records.
RMS normalization is often considered to be a more accurate method of normalizing audio than peak normalization because it takes into account the average power of the audio signal, rather than just the peak amplitude.
Loudness Normalization
Loudness normalization is the process of adjusting the loudness of an audio recording to a certain level. The loudness of an audio recording is measured in loudness units (LU). The goal of loudness normalization is to ensure that all audio files have the same loudness, making them easier to listen to and preventing ear fatigue.
Loudness normalization is typically used for broadcast audio, such as television and radio. Loudness normalization is required by many countries to ensure that the audio levels of all broadcast programs are consistent, making them easier to listen to and preventing ear fatigue.
Loudness normalization is often considered to be a more accurate method of normalizing audio than peak or RMS normalization because it takes into account the perceived loudness of the audio signal, rather than just the peak amplitude or RMS level.
Conclusion
Normalizing audio is an important process for ensuring that all audio files have a consistent volume, making them easier to listen to and preventing ear fatigue. There are several different types of audio normalization, including peak normalization, RMS normalization, and loudness normalization. Each method has its own advantages and disadvantages and is best suited for different types of audio.
When it comes to audio normalization, one solution that stands out is Mp4Gain. It is a software that allows you to normalize your audio files in a quick and efficient way. It can be used to normalize a single audio file or multiple files at once. It also supports a wide range of audio file formats, including MP3, WAV, and more. Furthermore, Mp4Gain is user-friendly and easy to navigate, making it a great option for both professional and casual users.
In conclusion, audio normalization is a crucial process for ensuring that all audio files have a consistent volume, making them easier to listen to and preventing ear fatigue. There are several different types of audio normalization, including peak normalization, RMS normalization, and loudness normalization. Each method has its own advantages and disadvantages and is best suited for different types of audio. Mp4Gain is a powerful and easy-to-use software that can help you normalize your audio files quickly and efficiently.
Loudness Normalization: Why is it necessary to Normalize the loudness of an audio or a video?
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.
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. )
When music began to digitize, going from analog to digital, problems were introduced that had existed since the 70s.
Using a 16-bit depth bit introduces noise into the recording. There are people who say that it does not make a difference to digitize to 16 bits or 24 bits and do the wrong calculation. Actually, there is a difference and it is fundamentally the noise it generates.
On the other hand, it was discovered a long time ago that listening to the loudest music in terms of volume gives the impression that there is an improvement in quality.
Hence, in the 70s, what is today known as “the volume war” was born.
Although it is completely true that there is a big difference in that some music sounds more or less loud in volume and that it is necessary to normalize it.
Also what is if the normalization is high gamma (high quality) as it can only be done by the Mp4Gain, which normalizes using the RMS per band (each sound band is normalized separately using the RMS -Root Mean Squared-), this achieves an improvement in sound of more than 35% percent.
Normalizing by peaks is a thing of the past, even RMS normalization is outdated unless a banding is introduced and maskings are removed and disguised, in addition to a harmonic reconstruct.
When encoding music, what is sought is to eliminate information and a good part of this information is that produced in masking, where a louder sound covers the weaker sounds and these are eliminated when encoding, but this supposes a loss of information that will result in a loss of quality.
Harmonics are also eliminated (sounds that represent mathematical multiples of the main sound that sound very low, but help to give a special timbre to each sound).
Then the modern normalize like the one that gets MP4 Gain which is done by RMS and by band and with reconstruction of harmonics and trying to reverse the masking, produces a noticeable improvement in quality.
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.
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.
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.
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 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.
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.
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.
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.
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.
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
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.
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.
