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What is bitrate? Bitrate video, audio, internet and more

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What is bitrate? Bitrate video, audio, internet and more …

Surely we have heard the word bitrate countless times when an expert user refers to a video or audio in digital format, and we have come to know that it is the element that defines the flow of data. But what exactly is bitrate? The question arises because not only in these fields is this parameter used.

Like the resolution and the final format of the digital video or audio, another determining factor to obtain excellent quality in an image or sound is, without a doubt, bitrate, a parameter that perhaps is not always taken into account and that not only applies to the field of audio or video. That is why in this article we will find a lot of information to perfectly understand what bitrate is.

Bitrate: Why it is so important in our digital life

Electronic devices have reached unthinkable operating speeds just a few years ago, and that is why today we hope that our device, be it a smartphone or a tablet, a computer or a hard disk, will respond to us at the moment and without hesitation. In this they have to see many and varied factors, but one of the most important is the bit rate at which it can exchange or process information.

The term bit rate, used in computing and telecommunications systems, basically refers to the amount of bits that can be transmitted in a given unit of time through a transmission system or between two digital devices. Depending on the context in which the term is used, the bit rate, or bitrate in English, is measured in Kbit / s or Mbps, kilobits per second or megabits per second, respectively.

Regardless of the unit of measurement for defining bitrate, higher numbers always mean better and higher quality values, although we must not forget that low bit rate values can also mean less signal processing by the hardware, very convenient in equipment such as smartphones, tablets or netbooks.

Bit rate on the Internet

In the case of the bit rate applicable to the Internet, the higher bit rate is better, since the content we receive from the network arrives faster. In other words, the higher the bitrate we get from our ISP, the better the connection and we can work much more comfortably.

A higher bitrate in an Internet connection means streaming movies and video in high definition, playing online with no delay and downloading really large files without problems and in a few seconds.

In the event that we want to know exactly what the bitrate of our connection is, we can do so easily and comfortably by accessing with our browser a site that is responsible for performing this test. One of the best in the market is speedtest.net.

Bit rate in audio and video

If we talk about audio and video, the meaning of the term bit rate differs a bit from what we use for the Internet. In this context, the bit rate refers to the amount of data stored for every second of data that they reproduce. To take an example, an MP3 file of a 320 kbps song offers a much higher quality than the same 128 kbps encoded file, obviously as long as both files have been created from the same source.

But we must always remember that if the source from which we obtained the files was of poor quality, then the copy will also be of poor quality, it has been encoded at 128 kbps or 320 kbps.

This also happens with videos, a much higher bit rate will offer a much better viewing quality than a video with the same resolution but at a lower bit rate.

The bit rate could be expected to increase each time the resolution grows as a larger amount of data is being processed. This means that while high bitrate rates can offer excellent display quality, they also require much more effort to process part of the hardware, forcing it, especially in modest and older hardware, to produce pauses and cuts.

Another aspect that we must also take into account since it is very important, is that video file formats use different sets of compression algorithms, which could also offer high quality with a more discrete bit rate. However, the extra process load for these types of videos can also complicate the processor and the systems involved in decoding.

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Digital audio

Digital audio is the representation of sound signals through a set of binary data. A complete digital audio system usually begins with a transceiver (microphone) that converts the pressure wave that represents the sound to an analog electrical signal.

This analog signal goes through an analog signal processing system, in which limitations in frequency, equalization, amplification and other processes such as compaction can be performed. The equalization aims to counteract the particular frequency response of the transceiver used so that the analog signal closely resembles the original audio signal.

After analog processing the signal is sampled, quantified and encoded. Sampling takes a discrete number of analog signal values per second (sampling rate) and quantification assigns discrete analog values to those samples, which means a loss of information (the signal is no longer the same as the original). The coding assigns a sequence of bits to each discrete analog value. The length of the bit sequence is a function of the number of analog levels used in the quantization. The sampling rate and the number of bits per sample are two of the fundamental parameters to choose when you want to digitally process a certain audio signal.

The digital audio formats try to represent that set of digital samples (or a modification) of them efficiently, so that it is optimized depending on the application, either the volume of the data to be stored or the processing capacity necessary to obtain the starting samples. In this sense there is a very widespread audio format that is not considered digital audio: the MIDI format. MIDI does not start from digital samples of sound, but stores the musical description of the sound, being a representation of the score of the same.

The digital audio system usually ends the reverse process to that described. The set of samples they represent are obtained from the stored digital representation. These samples go through a digital-analog conversion process providing an analog signal that after a processing (filtering, amplification, equalization, etc.) affects the output transceiver (speaker) that converts the electrical signal to a pressure wave that represents Sound.

Digital audio quality

The quality of the digital audio depends strongly on the parameters with which that sound signal has been acquired, but they are not the only important parameters for determining the quality.

One way to estimate the quality of digital sound is to analyze the signal difference between the original sound and the sound reproduced from its digital representation. According to this strategy we can talk about a specific signal to noise ratio. For audio systems that perform lossless digital compressions, this measure will be determined by the number of bits per sample and the sampling rate.

The number of bits per sample determines a number of quantification levels and these a signal-to-noise ratio of carrier peak that depends quadratically on the number of bits per sample in the case of uniform quantification. The sampling rate establishes a higher level for the spectral components that can be represented, and linear distortion may appear in the output signal and aliasing (or spectral overlap) if the signal filtering is not adequate.

For digital systems with another type of compression, the signal to noise ratio can indicate very small values even if the signals are identical to the human ear.

The reason is that the signal to noise ratio is not a good parameter of sound quality measurement because the quality perceived by the listener is determined by the response of the human ear to the sound waves, which does not perceive many of the possible differences Logically, if the signals are very similar, the ear cannot differentiate them, but they can also be very different and can be perceived as the original signal. Therefore, the evaluation of the quality of a digital system through sensitivity parameters of the human ear and specific tests with specialized listeners seems more appropriate.

It is in this sense that the quality of digital audio systems is evaluated today. Both MPEG and Dolby Digital (AC-3), which establish perceptual compressions, perform test benches to estimate the quality of the encodings.

The truth about audio formats and their quality

As you well know, there are three digital ways to play a song. From the original recording, through a copy with lossless compression (what we usually find when we buy a CD) or through a copy with lossy compression (which we usually download legally from the G.G internet).

The three files differ basically in the same property, which is none other than bitrate, that is, the information it contains per second. As more bitrate, more weight, so an audio file of a song without compression can quietly occupy 200mbs. In the second case (lossless understanding), the weight is reduced a lot (over 40, 50mbs), and is obtained by reducing the bitrate in those parts with silence, or with a wave oscillation in a single spectrum. To understand each other, the healthy young ear recognizes between 20Hz and 20KHz. A file with lossless compression (usually .flac files or those found on a CD) maintains this spectrum, reducing it when it is not necessary (silences). And finally there are the files with compression and loss (.mp3, .mp4, .flv, …) files that reduce this spectrum to the one that most ears recognize, leaving it around between 15Hz and 15KHz, obtaining a weight per file of song around 4mb, 5mb.
The latter has always been questioned, especially in musical circles, which assured that compression with loss greatly diminished the quality of what was reproduced, not allowing to admire the most serious or the most acute, thus losing all the completeness of the work.
As if that were not enough, mp3 files have different encodings (64, 128, 192 or 320 Kbps), with a greater or worse loss, and even constant (CBR) or variable bitrate (VBR) that is usually optimal when compressing with various bitrates Different moments of the songs.

Well, it has been more than 50 years before a good music lover programmer named Jeff Atwood decided to see if there really is a substantial change for the human ear between the different formats. In his blog, after several entries and several weeks of study in what would be called The Great Experiment of bitrate in MP3, we have finally obtained an empirical version of this eternal question.

But let’s make a brief summary of what we have in hand.
To test his hypothesis, Atwood decided to hang five audio files from his website, one of them being the original (without any digital treatment that modifies the bitrate), and another four tablets at various bitrates between 128 and 320 Kbps. The objective was that the user entered, listened to the five, and chose which one seemed to him to have higher or lower quality. Best of all, he obtained a not insignificant opinion of 3,500 visitors, hanging the results weeks later.

And from his observations you can get some gold reefs:
No doubt people knew how to differentiate the worst of all, such as the mp3 encoded with the worst bitrate 128 Kbps CBR.

The variable bit rate coding proved to be higher than the constant.
The most positive audio obtained was that of 160Kbps VBR, even higher than 320 Kbps CBR, and paradoxically also superior to the original audio of the CD.

From all this follows a corollary:
People are unable to ensure that it has more quality above 160Kbps, so it sends lossless formats to the horn that occupy one more scumbag, and that in practice, our ear cannot discern.
So you know. That’s over 15, 20 songs for a CD. There is no excuse.

MP3 vs FLAC vs AAC vs OGG: what differences does each audio format have?

Although streaming platforms such as Spotify are more fashionable than ever thanks to their great musical variety, reduced price and convenience of not having to manually download the files, many users still prefer to have music stored locally, for which there are formats like MP3, FLAC, AAC and OGG.

These formats are currently the most widespread for music on our devices, being able to pass files between PC and mobile without relying on the Internet and without being afraid of depleting our data rate. Most of the formats that we are going to deal with are formats that compress information, and therefore have quality losses. About the compression of images and files we talked a while ago.

Why does a compressed file occupy less?

Audio formats with losses: MP3, AAC and OGG

The first of the formats that we are going to try is MP3. This format, whose acronym stands for MPEG Audio Layer III, is the most commonly used format with loss of quality. It is not the one that offers the best quality or best compression, but its great compatibility has made the standard format for music for decades.

Another widely used format for sound in recent years is AAC. It is very similar in MP3 performance, but has the advantage that it is able to offer the same quality in a smaller size. This is the reason why platforms like Apple’s iTunes use it, and the fact that Apple uses it has made its compatibility as great as MP3’s today. AAC is also used to compress stereo sound in movies of 1 or 2 GB in size that we find in various torrent portals, direct download or streaming.

The next most used format is OGG, or OGG Vorbis, it is a free alternative to AAC and MP3 (although the MP3 patent ended last May). Its size is similar to that of MP3, but its compression is smaller, keeping a higher audio quality than MP3, especially at high frequencies, which destroys the MP3 the lower the bitrate. In addition, while MP3 reaches 320 Kbps, OGG reaches up to 500 Kbps.

Lossless audio formats: FLAC, ALAC and WAV

On the other hand, we have FLAC. This lossless format is free, as indicated by its name (Free Lossless Audio Codec). The size of your files is between 5 and 10 times larger than MP3, but it has no losses, although the audio is “compressed.” Thus, it occupies much less than uncompressed formats such as WAV or AIFF, and maintaining the same sound quality.

The equivalent of FLAC in Apple is ALAC. Although it is not as efficient as FLAC (its files occupy more), ALAC owns Apple, and is the only alternative that can be used in iTunes, since the platform does not read FLAC.

In short, the best format to use is always FLAC if you can afford its large size, followed by AAC and OGG. If you have no choice, MP3, although it is the least desirable option, is the most widespread today, and what you will be forced to use for a lot of music on the network.

What audio formats exist? All you need to know

FLAC, WAV, AIFF, DSD … these are just some of the acronyms you can find when looking for a digital format. They are also accompanied by technical data such as sample rates and bit depth. So many terms can leave you more misplaced than a chicken in a dance. And unless you are an expert in digital sound, the process to choose the audio format that best suits your needs can be a mess. But if they explain it to you, the subject is relatively simple. That is why in Culturasonora we have prepared a complete guide on the different audio formats used. This will prevent any acronym from taking you on the dark side, dear Padawan.

Sample Rate and Bit Depth.
MP3s vs WAVs vs AIFF.
OGG vs FLAC vs ALAC.
What is the DSD format?
How to listen to the DSD?
MQA audio Hi-Res.
What is Bit Depth and Sample Rate?

These two concepts are basic. To understand how audio formats work, you need to know what Bit Depth and Sample Rate are. They are two measures that indicate the quality of a digital audio file. We will try to summarize it so that you stay with the general idea

When you read the specifications of the audio formats you find a couple of figures. For example: 32-bit / 192kHz or 24-bit / 96kHz. These numbers indicate the bit depth and the sample rate. These references tell us how much information the different formats transmit and the sound quality. For example, the audio we hear on a normal CD, or on a Spotify stream, is 16bit / 44.1kHz. Samples are always measured in Hertz (or hertz) and bit depth in Bits.
Softwares or hardwares do not usually work with a continuous flow of information but often use pieces, samples or samples to effectively manage the data that is transmitted. The sample rate is the number of samples per second that are obtained from a recording. The higher the number of times a device plays the samples, the higher the sound quality. Each of these extracts or samples has a certain amount of information, which is the bit depth, or bit depth.
To understand it better, we are going to make a slightly beast analogy, which is not entirely true, but which will help you to make sense of all this. What interests us. If you control a bit of photography and image you will get it right away: the sample rate would be something similar to the frames or frames per second of a video, and the bit rate would be similar to the pixels of a photograph. The higher the bit depth number, the more information each sample will have. The more pixels an image has, the more resolution each frame of a video will have. The more frames per second a movie has, the greater the definition. In short: the higher the number of the Bit Depth and the Sample Rate, the higher the quality of the audio file.

Audio formats: MP3 vs WAV vs AIFF

What is the MP3 format?
If you are interested in getting some audio fidelity and decent sound from your files, you will want to avoid this format. Why? Because basically an MP3 is a file that sacrifices audio quality to minimize size. They weigh very little for any device to read. The negative? The compression of these files provides a poor, almost lifeless sound. Nowadays almost nobody uses that format seriously. Even its creators recently finished the license declaring her dead. But surely every now and then you find a zombie file with this format.
What is the WAV format?
WAV (Waveform Audio File Format) are equally common but better for anyone who wants a decent audio format. They are higher resolution files than MP3s. A WAV is an audio piece that is encoded with something known as Pulse Code Modulation (PCM), a medium that encodes analog audio parts and converts them into digital so that they can have the Sample rates and the Bit Depth of the that we have talked about before.
What is the AIFF format?
The audio format AIFF (Audio Interchange File Format) is very similar to WAV, since it also uses the PCM to encode analog audio pieces and present them in digital format. This format was born as an answer from Apple to the Microsoft WAV, and at the beginning it could only work on MAC computers. Currently, the AIFF and WAV are more or less interchangeable.
In summary…
To close this topic we will tell you that if you have a file in WAV or AIFF audio formats you will hear a piece of good quality sound. Normally these formats are used in files that we play through our services, such as the iTunes music library. We will not see them in online streaming services, which tend to use special types of files. Now we will review that point

Do you differentiate between an mp3 encoded at 128 and one at 320 kbp?

Surely more than once you starred in or attended a dispute between people who say that you notice a lot of difference between an MP3 encoded with one or another level of compression, or between a CD and an MP3. However, there are very few people able to distinguish these nuances. That’s why at mp3ornot.com we propose this challenge:

Are you able to differentiate between an mp3 encoded at 128 kbps from another at 320 kbps? If you think you have your ear developed enough to capture that difference, I challenge you to take the test … and then tell me.

Data:

The Mp3 (MPEG-1/2 Audio Layer 3) was one of the first types of audio compression with almost imperceptible losses to the human ear. Its compression rate is measured in kbps (kilobits per second), with 128 kbps being the standard quality, in which the file size reduction is about 90%, that is, a ratio of 10: 1. That compression rate can currently reach up to 320 kbps, the maximum quality, in which the file size reduction is about 25%, that is, a ratio of 4: 1, going before 192 kbps, 256 kbps, that is, the maximum quality that can be removed in Mp3.

The lossy compression method used in the compression of the Mp3 consists in removing from the audio everything that the human ear would normally not be able to perceive, due to phenomena of masking sounds and limitations of human hearing (although people with absolute hearing can perceive such losses).

How to compress an MP3 file

Knowing that the MP3 audio format has become the most standardized and used worldwide in recent years, we have thought it pertinent to talk about the different parameters that make an MP3 file respond to one quality or another.

The first thing we have to know is the meaning of MP3, and it is nothing more than a compressed digital audio format that although by nature suffers a loss of information in the conversion process, it is not audible by the human ear, which It implies an assumable loss since we will not be able to perceive it in broad strokes.

Generally, an MP3 file is capable of reducing the size of an original audio file without altering quality. What this means is that in the conversion process for example of an audio file with CD quality, the result of the MP3 file would be practically identical to the original, leaving as standard ratio 1 minute = 1 MB.

That said, we can begin to clarify some parameters that will determine the quality of an MP3 file, which in its vast majority, depends on the bitrate or Bitrate.

Impact of Bitrate in MP3 quality
The MP3 file format allows you to select the compression ratio of the source file. The margins at the domestic level are between 8 Kbps and 340 Kbps, with 128 Kbps being the transfer rate equivalent to CD quality.

Bitrate is the unit of measure for the rate of data transfer read from an MP3 file. The higher bitrate an MP3 file has, the greater the amount of data that a player can obtain in the unit of time (Second).

The more instrumental content or quality an MP3 audio file contains (sound effects, recorded audio tracks, high frequencies, low frequencies, etc.), the higher the transfer rate it will require to fully reproduce the information, and at this point, it is where it is defined The quality of the MP3 file, since if we compress that file, we reduce that bandwidth, we will be sacrificing some of that data, resulting in loss of information that will influence the final result of the MP3 conversion.

In summary:

If the file lasts 5 minutes and weighs 3 MB, we would be talking about a low quality MP3 file.

If the file lasts 5 minutes and weighs 9 MB, we would be talking about a high quality MP3 file.

What are the differences between mp3 and mp4?

When we want to listen to our favorite songs, we don’t skimp on tastes or ways of listening to those pieces of the moment. Although there is no doubt that when we have two good devices through which we can listen to them we look for the advantages of some compared to the others to select the best one.

The differences between an mp3 and an mp4 are varied, everything will be judged by the qualities that make them more attractive to us according to a value judgment based on their characteristics.

When we make a comparison of differences between an mp3 and an mp4 we observe that the first only supports audio formats while the second allows all types of audio and even video and images with the possibility of having greater storage capacity since it allows compressing much more the formats in order to have many more songs, videos and images in it.

Both are equally functional as they offer a possibility to listen to our favorite songs from the device. Another notable difference between an mp3 and an mp4 is the possibility of storage so it is very regular to check how many files we can save in one or the other.

Likewise, the mp3 due to its MPEG-1 Audio Layer 3 condition decreases the size of the original file, which generates losses when listening to the songs that we store in it, since they are compressed and undermine their quality. The mp4 player for its part, was designed years later so it is seen that it supports the video and image format but in turn does not cause the sound quality to be lost both in the songs and in the videos although it also Compress without altering your audio levels.

In their design there is no notable difference because they have similar models only changes is the name of the device. The two big differences are that the mp4 is multimedia while the mp3 only allows you to play sounds. And its compression causes the audio quality to be lost in the mp3 while the mp4 is an improved device that compresses the format without losing its quality and efficiency.

High resolution audio: myths and realities – 2

If we stick to the characteristics of the CD we can see that our music is obtained by taking 44,100 samples per second (correspond to 44.1 kHz) from the original analog signal, and each of them is encoded in a data package that It uses 16 bits. And at this point, finally, it is where high-resolution audio comes into play.

Coding

The starting point of this technology is easy to understand: it presupposes that if we increase the resolution, the sampling frequency, or even both parameters at the same time when passing an analog signal to the digital domain, we can “reconstruct” the original analog signal With more precision. And it really is. For this reason, the specifications commonly used in high resolution audio formats are 24 bits and 96 kHz, or 24 bits and 192 kHz. Both options, on paper, should allow us to recreate the original continuous signal more accurately than the 16 bits and 44.1 kHz of the CD, or, what is the same, will discard less information from the original sound.

But this is not all. In addition, increasing the resolution to 24 bits increases the dynamic range and improves the signal-to-noise ratio (our Xataka Smart Home partners explain what these parameters mean in this post). A resolution of 16 bits allows us to encode a total of 65,536 possible levels for each of our samples, while a 24-bit one reaches 16,777,216 levels.

The resolution commonly used in high definition audio formats is 24 bits, and the sampling frequency 96 kHz or 192 kHz

The difference between the two extremes, which is where the lowest and highest levels are located, indicates the dynamic range difference between one resolution and another. With all this data on the table we can think that high resolution sound should offer us more quality than the audio of a standard CD. And it is so, but, as we will see later, there are factors that limit the experience and that users must take into account, beyond what the industry “sells” us.

Internet: key to the success of HD audio

At this point we can understand without difficulty that the size of a sound file depends on the resolution and sampling frequency used to encode the music it contains. The same issue occupies much more if we digitize it at 24 bits and 96 kHz than if we do it at 16 bits and 44.1 kHz. However, we have a very interesting resource that helps us save space: compression. Currently, high resolution audio is usually distributed in six different formats (some of them offer compression without loss of quality): FLAC (compress without loss), ALAC (the lossless compression technology proposed by Apple), AIFF (it is the format of Mac sound file), WAV (this is the sound file format created by Microsoft and IBM for PCs), DSD DFF (SACD format encoding technology) and DSD DSF (DSD variant for Sony VAIO computers).

Of all the formats that I have just mentioned, the most used to distribute high resolution music on the Internet are FLAC and ALAC because both offer a very interesting compression rate, and without loss of quality. And we all know that size matters on the Internet. And a lot. In fact, the network is playing an essential role in popularizing high resolution sound.

High resolution audio: myths and realities

High resolution sound is in vogue. A good part of the manufacturers of music equipment, and more and more record labels, especially those that sell over the Internet, seem determined to convince us that high-resolution audio is what all of us who love music should aspire to if We want to enjoy it with the highest possible quality.

On paper there are technical foundations that justify the existence of high resolution audio and indicate that its quality should be higher than that offered by the CD. But there are also solid reasons that invite us not to take their superiority for granted, at least not in such a clear way, and to question some of the virtues that the industry sells us. Let’s see what high resolution sound is, what we need to enjoy it, and, above all, if it really offers us a better experience than music with standard quality (that of the CD).

What is high resolution sound?

To understand in a simple way what high resolution sound offers us, it is good to review how music is stored on CDs that we all know. These discs, unlike vinyl, allow us to store information in the digital domain, while vinyl discs are analog. This means that the music on a CD is encoded in the form of ones and zeros, in exactly the same way as the information we have on the hard drive of our computer, which is also digital.

The sound of your smartphone may resemble that of cinema thanks to neuroscience: this is how Dolby has integrated Atmos audio into it
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The sound of your smartphone may resemble that of cinema thanks to neuroscience: this is how Dolby has integrated Atmos audio into it
But a CD does not have an infinite capacity; in fact, the size of the tiny notches that we see on its surface if we look at it with a microscope and the distance that separates each one from the adjacent ones reveals how much information it is capable of holding. Precisely, in a simplified way, this is what differentiates the CD, the DVD and the Blu-ray Disc: the size of the small holes that encode the information and the distance that separates them. If we compare two discs with the same diameter, the one with these smaller notches and more together will have more capacity. This parameter is precisely what determines the wavelength of the laser that we must use to extract the information.

Formats

The technology of the CD format was developed at the end of the 70s by Philips and Sony, and it was the engineers of the latter company who proposed to encode the information using a resolution of 16 bits and a sampling frequency of 44.1 kHz. But these figures were not chosen at random; These specifications allow this format to reproduce the sounds that are in the frequency range that goes from 20 Hz to 20 kHz, which coincides quite accurately with the frequency limit that the human auditory system is able to perceive, even bearing in mind that Not all people have the same hearing ability.

The CD uses a sampling of 44.1 kHz to, according to the Nyquist-Shannon theorem, be able to reproduce frequencies up to 20 kHz
To understand what is the resolution and the sampling frequency without going into too cumbersome details we can think that to be able to store an analog signal, and, therefore, continuous, in a digital medium, which has a limited capacity, it is essential « Chop up that continuous signal into small fragments, or samples, and introduce as many as fit into the digital medium. The resolution indicates the number of bits that we can use to describe each of these samples, which, in turn, reveals the number of variations or possibilities that each of them can adopt. And the sampling frequency tells us how many we are going to be able to take.

(Part 1)

Audio Normalizer in 2019

We all enjoy music, and we have all modified our behavior by adapting to the new devices.
Today nobody sits to look at the cover of an LP while it turns on the turntable.
Those times are behind.
Today, however, it is quite easy to get an audio file (mp3, flac, ogg, m4a, etc.) of some song that we listen to and like.

But that “get” implies downloading a file from an unknown source. There is no risk in it, at least not a risk of getting infected. But if we run the risk of downloading music files that have been encoded by someone who has no idea what they are doing.

I wonder, why do we have some that have 128 bitrate and another 92 and another 320k? And the same goes for all the settings with which it was encoded. It is as if they were chosen randomly, without really knowing what each thing is.

And then comes the consequence that they all sound “different”, with different volume levels, which we perceive with different volumes and sounds. Our collection, which sometimes becomes very large, is really uneven.

Then appears the desperation to find a software like Mp4Gain that allows us to be a volume leveler, volume enhancer, volume booster. In short, so that everyone can understand: an audio normalizer. And this audio normalizer should be able to normalize the most popular audio and video formats, the mp3 is no longer enough, as was the mp3gain.

Unfortunately some of our audio files will be co-opted by encoders that are not really standard and maybe the file cannot be repaired, but that is the minority, the other 99% will be able to normalize perfectly.