What is a loudness normalizer or an audio volume normalizer?


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What is a loudness normalizer or an audio volume normalizer?

audio volume normalizer
audio volume normalizer

A volume normalizer is used to make sure that audio files play at the best possible volume without clipping and also that all audio files play at a similar volume.

audio volume normalizer
audio volume normalizer

A volume normalizer analyzes an audio file and then adjusts it to sound at a specific volume level. This is often done with audio files that are uploaded to file sharing sites, so that all users can listen to the audio at a similar volume.

It is quite common to find volume differences in the files that are downloaded from the internet, since these have been created from a wav but different sampling and bit frequencies have been used to create them.

A volume normalizer can analyze an audio file and then apply gain or attenuation to adjust the volume of the file to a specified level. This is useful if you want to ensure that all audio files played on your website or in your application sound at a similar volume.

Because it’s frustrating to have a collection of audio or even video files and find that when you play them they play at different volumes.

For this reason, it is necessary to use a volume normalizer, with which you can make sure that all audio and video files are heard at a similar volume.

It is important to ask yourself if the bit rate is important for the quality of an audio or video file, the same for the sample rate.

The answer is no, not always.

Bit rate and sample rate refer to the amount of information that can be stored in an audio or video file.

 

The higher the bit rate, the higher the quality of the audio or video file.

 

However, sometimes a low-quality video or audio file can sound better than a high-quality file.

 

This is because the bit rate and sample rate are not always indicative of the quality of the audio or video file.

 

There are many factors that can affect the quality of an audio or video file, such as the encoder used to encode the file, the quality of the microphone used to record the file, the quality of the equipment used to play the file, etc.

 

In summary, the bit rate and the sample rate are not always indicative of the quality of the audio or video file.


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What is an audio normalizer?

What is an audio normalizer?

audio normalizer
audio normalizer

How can an audio normalizer help?

audio normalizer
audio normalizer

Mp4Gain is a tool that will adjust the volume of an mp3 file so that the loudest and softest parts of the sound are more balanced.

The main advantage of an audio normalizer is that it can be used to make a song louder without clipping or distorting it. It achieves this by increasing the volume of softer sounds, which in turn makes louder sounds quieter.

An audio enhancer is a similar tool, but instead of balancing out the volume, it increases certain frequencies to make a song clearer and more pleasant to listen to.

Normalizing the volume of audio files is crucial for many reasons: it makes listening to music more pleasant, it increases the clarity of speech, and it can even help you sleep better.

Mp4Gain is an audio normalizer and volume booster. It can be used to automatically adjust the volume of all your music files so that they are at the same level.

Mp4Gain is an easy-to-use tool for adjusting the volume of all your music files to a uniform level. It does not need any technical knowledge, just drag and drop your music files into the program window and click “Normalize”.

Mp4Gain is an audio normalizer, it can help you to increase the volume of mp3 files. It can also be used as a volume booster or audio enhancer.

Loudness normalization

Loudness normalization

Loudness Normalization

When you have a lot of mp3 files, you often look for loudness normalization.

Loudness Normalization

What usually happens is that we have mp3s (although Mp4Gain can do Loudness normalization of many other audio and video formats!!) that have been created with different settings, for example different bit rates… which causes them to have a loudness different and that is annoying to the ear.

Many times we have been collecting mp3s from different sources, finding one here and another there and over time we have managed to have a good collection that is worth thinking about, but we have a problem: the loudness differs between different music or video files.

And this has generated that we desperately need to find a solution.

Mp4Gain is the result of many years of experience and is definitely the best normalizer out there, I have no doubt.

Even for very advanced users, it offers different settings to adjust exactly what you are looking for. Pewreo if you are a common user, you will not need anything, just load the song or video (you can normalize one or hundreds at the same time) and click a button, it’s that simple.

Mp3 normalize volume level software

FAQ

Normalize Audio

Mp3 normalize volume level software

Normalizing the volume level of an mp3 is quite simple using Mp4Gaion, which also allows you to normalize the volume level of other audio and even video formats.

Convert audio and video files and normalize them?

It’s perfectly possible to do it with Mp4Gain, you can normalize audio or video files in all major formats simultaneously and get any format you need.

Mp3 normalize volume level software

Audio Normalization

The normalization of volume levels is something that has existed for many years.
This arose with the need to be able to get the different songs or files to have a similar volume level.
It really wasn’t necessary in the vinyl era, for a lot of reasons.

First of all, changing from one disc to another took time, enough so that I didn’t notice if there was any difference in volume level. Unlike any playlist of mp3s or any other format, which play one song after another and if there is a noticeable difference in volume level, we perceive it immediately.

We also have the fact, which is not minor, that the quality of audio playback today is much higher.

Today any device used to play an audio file has enormous capacity in terms of sound quality. Today we handle as a common thing to talk about sample rates of 44100 or 48000 frames per second or 192 and up to 320 kilobits, etc. In other words, we are already very familiar and we have at our fingertips the possibility of choosing options that directly affect not only the volume level but also the quality.

Mp4Gain is the most powerful and modern normalizer that can not only normalize audio in many formats, but can also normalize videos or extract audio from video and convert it to mp3 or any other format you want.

Audio compression for music lovers

Audio compression for music lovers

Lossy compression

 

the truth about high bitrate lossy compression

lossy compression

In the opinion of most people, the word music lover is most often associated with a person who not only loves and collects music, but also appreciates high-quality music, and not only in artistic and aesthetic terms, but also the quality of the recording of the phonogram itself. Just think, a few years ago, an audio CD was considered the standard for music quality, whereas a computer, even in dreams, could not compete with the quality of a CD. However, time is a great joker, and he often likes to turn things upside down. It would seem that quite a while, a year or two passed and … that’s it, the CD on the PC went into the background. Don’t ask “why?”, You know the answer to this question yourself. Everything is to blame for the revolution in the world of computer sound: audio compression (hereinafter referred to as audiolo compression which means lossy compression to reduce the size of the audio file), which made it possible to store music on disk hard, lots of music! In addition, it was possible to exchange it over the Internet. New sound cards have been released, capable of almost “squeezing” studio quality out of a piece of hardware that seems useless in terms of music. Today, even having a computer that is not very smart in performance, having bought a Creative SoundBlaster Live! and remembering that since Soviet times there is a good amplifier and good acoustics, you will get nothing but a high-quality music center, the sound of which is inferior only to very expensive audio equipment (average or even the highest Hi-Fi category ). Add to this the general availability of music files and you understand that you have the power in your hands. And then there is a revolution, and you understand that a compact disc is no longer so convenient, you are fascinated by something completely different: the magic “MP3” signs. You cannot eat or sleep; you are faced with the seemingly insoluble “chicken and egg” question: how to “squeeze” and, most importantly, how to “squeeze” …

This is where I will help you. This article is the beginning of my new series of informational materials on music on the computer. For over a year developing OrlSoft MPeg eXtension and maintaining an extensive database of MP3 files, I have accumulated a great deal of research on audio compression. It is these studies that I will try to share with you. Many articles have been written on audio compression by different respected authors, so I will try not to write what I can easily find in other sources of information. I would like to put my position on the subject we are considering simply and clearly. We will not consider audio compression to be as compact a tool as possible put audio information on your hard drive (so that you can record so many hours of music there). Yes, compression allows you to record music more compactly, but my goal is to minimize quality loss by converting “pure” audio to compressed audio. This is why only high bit rates and qualitatively compressing encoders are considered in these modes. So it is much more convenient to work with compressed audio – instant access to any track from any album, convenient software for playback. And, of course, the financial issue has not been forgotten either.

Of the audio compression formats that exist today, in my opinion, three deserve attention: MP3 (or MPEG-1 Audio Layer III), LQT (as representative of the MPEG-2 AAC / MPEG-4 family) and a Completely new OGG format (Ogg Vorbis) developed by a group of enthusiasts:

MP3 is by far the most used of these (mainly because it is free). Let me remind you that it was thanks to the MP3 format that the victorious procession of compressed audio took place. However, as often happens with pioneers, little by little it is losing ground and giving way to new and better formats.
The second format, LQT, is a representative of a new direction of audio coding algorithms, a representative of the AAC family. This is a fairly high quality, but commercial and highly classified format.
OGG became widely known to the public this summer and is currently developing rapidly, soon (with the launch of the Encoder and Decoder) it should beat MP3 with better sound quality with smaller file size.

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.

What exactly is normalizing?

Music is distinguished by what is often called “dynamic” and which refers to the changes (more or less abrupt) of the “effort” with which certain notes or passages are interpreted.
Whether it is an instrument or the voice.

singer

Any vocal performance that has been considered virtuous, in general terms, will have a dynamic that goes from very soft passages, almost whispered, to intense passages, with a high volume, singing at full voice.

At the time when vinyl existed as the option to listen to music, it was not felt (at least it went almost unnoticed) the fact of noticeable differences between the loudness or the volume of a song.

It was with the advent of digitization and the possibility of its variants (opting for different bitrates, sample rates, bitdepths, etc.) that this difference became very evident.

And with the appearance of mp3 and its distribution or exchange, at the same time that winamp and distribution lists arose, when it was inevitable and it was even started to look for solutions.

Napster

These first ones were based on the sound peaks and their results were very inefficient.

Returning to the mention of the mp3, situations such as masking (where information is removed) further marked the problem of differences in volume.

Then began to use the RMS that rather mediates the average power that the song had, more than the peaks.

Initially, it was enough to put a slower reaction level to the volume meter, to have a more general idea and less impacted by the volume peaks.

And so, the way of listening to music and considering what normalization was evolved.

Finally it appears to be somewhat closer to a mixture of a volume limiter and a compressor.

What is a volume limiter? It is a hardware (although lately there are also limiters in software version) that ensures that no peak exceeds a maximum limit.

A compressor, on the other hand, is a device or software that is used to “compact” the volume, preventing the parts with the lowest volume from being too low and at the same time preventing the high parts from exceeding a range that has been assigned. We would say that the compressor dampens the increases and decreases in volume.

To this we can add an equalization that differentiates the bands and treats them differently both in the limitation and in the compression. Each frequency band has a different treatment in the Mp4gain and that produces a very efficient result. It is NOT the only improvement offered by Mp4Gain, but this is described here. In other articles we will deal with other differences.

Mp4Gain is the best normalizer of 2020 and this is clear when using it.

Audio Normalization, understand what it is about

Audio Normalization, understand what it is about

Difference between Peak level and RMS in Audio

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

Tube Compressor-Limiter

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

DIFFERENCES

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

What is RMS?

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

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

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

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

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

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

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.

Radio Studio Compressor

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.

Dynamic range compression

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.