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Understanding Audio Normalization

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.

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.

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Sound level, volume, normalization

Normalize Audio

This article provides a brief explanation of the terms Sound Volume, Sound Level, Normalize, Gain, and some others, and their relationship and use in relation to the Digispot broadcast automation system.

Volume normalization

Sound level
The term sound level refers to the amplitude level of the sound signal. With regard to a programming item, MDB item, or another piece of sound, we are talking about the peak (maximum) signal level in the entire piece. This level is measured in units of dBFS and is almost always negative. This level is important because it depends on how much the level can be increased, and therefore the volume of the sound, without exceeding the theoretical threshold of 0 dBFS.

The signal level indicators are intended for visual observation of the current signal level in real level.

A diagram of the signal level change over time is called a signalgram and is used to display phonograms and other sound elements in various windows of the Digispot system, for example, the splice editing window, when editing audio, etc.

In the Digispot system, the maximum level of the programming item and CDM is calculated once and stored for later use, eg for normalization.
The determination of the peak signal is combined with the simultaneous determination of its loudness, these values are always calculated together.

True sound level
The term True Sound Level refers to the hypothetical amplitude level of an analog sound signal, which is an interpolation of an existing digitized soundtrack. The difference with just “Level” is that when sampling, the sample points on the time axis may not reach the maximum points of the analog signal. For example, if we have a sinusoidal signal with a frequency of 11025 Hz and we digitize it with a frequency of 44100, then the peak value of the digitized phonogram level can have a value from -3dBFS to 0dBFS, depending on the phase shift of the point of sampling on the time axis. enter the sign. At higher signal frequencies, the peaks can be further underestimated.

ITU-R BS.1770-3 (Annex 2) defines the algorithm to calculate the “True Peak Level”. The proposed procedure is reduced to increasing the sampling frequency 4 times and filtering, then the maximum amplitude is found from the interpolation of the signal obtained.

In the Digispot system, the peak indicators in the editor, property windows, and splices have the ability to display the actual sound level.

Sound volume
Loudness is an estimate of the intensity with which the listener perceives the material. This value is calculated using a special algorithm that takes into account the perception of human sound, developed by ITU \ ITU-BS.1770.

Loudness is measured in LUFS units, which are physically identical to decibels. The volume is directly related to the level of the signal: the higher the level of the signal, the higher its volume.
Numerically, this relationship is linear: if the signal level increases by 6 dB, the volume will also increase by 6 LU. (To be mathematically precise, the relationship is not linear, but for most practical applications, the deviation from the linear relationship can be neglected.)

The loudness control in real time is carried out by volume indicators, there are two of them: M – Momentary and S – Short term, they differ in the measurement intervals: 0.4 sec and 3 sec, respectively.

To evaluate the loudness of a range of sound, a special technique has been developed that calculates the value of the loudness of the range, denoted by the value I and called integrated loudness. This is the value you refer to when talking about the loudness of a programming item or MDL.

In the Digispot system, the integral loudness of the programming item and MDB is calculated once and stored for later use, eg for normalization.

In Russia, the methodology for measuring the volume of programs is determined by the order of the Federal Antimonopoly Service of May 22, 2015 No. 374/15. The loudness of the programs is regulated by Federal Law 338.

Relationships between digital audio peak level, actual peak level, volume, and notation
When talking about the signal level (more precisely, the peak level), the notation dBFS – dB Full Scale is used. This scale has a 0dB point tied to the full range of the signal represented in the bit width used. For example, with 16-bit audio samples, the representable values are -32768 to +32767, so the signal level value in dBFS is calculated as 20 lg (s / 32768), where s is the value of the sample in this representation or the maximum absolute value of the samples in the interval of interest.