DIFFERENCES BETWEEN NORMALIZE AND MASTERIZE

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The process and the differences between normalizing and mastering are often confused. Although it may seem to be the same, it is not.

Mastering can be of crucial importance according to which processes, for example: in musical matters, there are mastering engineers who are dedicated exclusively to that.

That does not mean that we cannot learn or acquire the necessary knowledge to be able to properly use some processing effect or some plugin in an appropriate way to be able to get more out of our audio file.

But you have to keep in mind that this audio processing helps your audio montage, song … sound with more punch, more strength, more energy, have more life.

Is mastering compressed or limited?

Rather those two processes and some more are done.

Its mission is to maintain the same volume amplitude throughout the audio file, that is, it compresses when it has to compress and limits when it has to limit.

I’m going to give a rough example of what manual mastering would be like.

Can you still imagine the sound technician who detects when the signal volume is too high (the singer gets too close to the microphone, shouts …) and lowers the fader. Or the opposite case, when it detects the low volume (the singer moves too far from the microphone, does not speak with enough force …) and raises the fader. Always trying to maintain the same volume amplitude.

I’m going to give you a homemade definition: “lower what is high and raise what is low“.

As before it was an invented example, to do the job of processing the sound we regulate the different parameters available to the “processor” (Mastering is also called “processing” since in the past a device called “processor” was used which comes from “dynamics processor”). These parameters are:

The threshold (threshold): fundamental characteristic of the compressor that represents the point or level from which if the volume of the sound exceeds or lowers it, the dynamics processor is put into operation.

Ratio (Attenuation or Gain Ratio): Defines the amount of attenuation or gain that is applied to the signal. At noise gates the attenuation can be preset so that it really is a mute.

Attack time: This is the time it takes for the signal to attenuate, limit, mute or amplify. In general, slower times work best at low frequencies and fast ones at high frequencies. When processing a signal containing all frequencies, a compromise situation is forced.

To maximize the energy of the signals, particularly in broadcasting applications, there are multiband compressors that divide the spectrum into several bands and apply different times to each.

Release time: It is the opposite of the attack time, that is, the time it takes to go from the state where the processing is running to rest. They are usually longer times than those of attack.

Hold (maintenance time): Specifies the minimum time that processing will take place.

Stereo link (stereo link): With dynamics processors in general when used to process a two-channel (stereo) signal, it is necessary to link the processing action of both channels to happen on both at the same time. Otherwise, the sound image will be confusing and changing from the center to one side or the other.

Automatic: This function allows you to control any of the parameters listed automatically depending on the characteristics of the signal.

By pass (deactivation): Activating it allows you to hear the unprocessed signal, while if it is not activated you hear the processed signal.

Normalization is a process by which the highest peak is sought and reduced or increased (dB) as adjusted. Never pass the 0dB in normalization or mastering, because then it would be itching “clipping”.

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The quality of YouTube videos leaves much to be desired: they need an update

When we watch a video on the platform, we can usually appreciate that, despite finding videos in 1080p resolution, the compression applied by the platform is too aggressive. This causes the final quality of the video we are watching to differ greatly from that of the original file. The codec that YouTube uses is H.264 / MPEG-4 AVC, using various profiles or “levels” that specify the maximum resolution, frames per second and maximum bitrate of each quality.

We have analyzed a few videos, and we have taken a fairly representative one that is available on both Vimeo and YouTube to see how both platforms compress the videos. In addition, we have seen the maximum and minimum bitrate that each video can have according to the YouTube Help page for each resolution. The audio, as we discussed in summer, reaches 128 Kbps, leaving 320 Kbps only for YouTube Red users.

What sound quality (bitrate) do YouTube videos have?

The bitrate for 1080p videos is too low: 4K is the way to go
The bitrates that YouTube says it assigns to each video are the following, with the profile level in parentheses:

4K / 2160p
60 fps: Between 20,000 and 51,000 Kbps (L5.2)
30 fps: Between 13,000 and 34,000 Kbps (L5.1)
1440p
60 fps: Between 9,000 and 18,000 Kbps (L5.1)
30 fps: Between 6,000 and 13,000 Kbps (L5.0)
1080p
60 fps: Between 4,500 and 9,000 Kbps (L4.2)
30 fps: Between 3,000 and 6,000 Kbps (L4.1)
720p
60 fps: Between 2,250 and 6,000 Kbps.
30 fps: Between 1,500 and 4,000 Kbps.
480p: Between 500 and 2,000 Kbps.
360p: Between 400 and 1,000 Kbps.
240p: Between 300 and 700 Kbps.

In our tests, the bitrates we obtained for the previous video were the following:

4K at 30 fps
Vimeo: 19.4 Mbps (file size: 943 MB) (capture)
YouTube: 17 Mbps (file size: 821 MB) (capture)
1080p at 30 fps
Vimeo: 4.31 Mbps (file size: 219 MB) (capture)
YouTube: 3.2 Mbps (file size: 160 MB) (capture)
vimeo vs youtube compression

As we see, Vimeo files occupy more not only because of the lower compression of the videos, whose quality is superior to the naked eye, but that Vimeo’s sound quality doubles that of YouTube, since it reaches 256 Kbps by 128 Kbps from YouTube. So that you can see the difference in image quality, you can open the same New Zealand Ascending video on YouTube and Vimeo, and we have also left four captures at the same moment of each video so you can save them and see comfortably the video difference.

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.

The great experiment on MP3 quality: no, there really isn’t that much difference with CDs

This article was originally published in Cooking Ideas, a Vodafone blog where we collaborate weekly with the goal of creating stories that “feed the mind of ideas.”

A programmer named Jeff Atwood said some time and several entries from his blog, the always recommended Coding Horror, to a healthy entertainment he called The Great Experiment of bitrate in MP3. Its objective: to verify empirically if for ordinary people there are really qualitative differences when listening to music in various MP3 formats compared to traditional ones.

The contestants were the traditional formats called “no loss of quality”, basically CD (Compact Disc) and FLAC versus compression formats with loss of quality: MP3 with different bitrates. The bit rate, better known by its name in English, is a key feature because it basically determines how much information is transmitted per unit of time: in this case it is the waves that define the music and become human voices and instrument notes . In the world of MP3 encodings of 64, 128, 192 or 320 Kbps (kilobits per second) are usually used.

Like everything in life, music coding is a compromise between quality and quantity: a song stored in the best possible format – for almost all experts, that is the CD – can occupy about 50 MB (megabytes), maybe 40 or 35 only using some of the lossless compressors that save some space without loss of quality (FLAC, Apple Lossless, etc.). That same song in MP3 can vary between 4, 8 and 12 MB depending on the bitrate (64, 128 and 192 Kbps). To further complicate the matter, you can also choose between a constant (CBR) or variable (VBR) bitrate that is usually optimal when compressing different moments of the songs with various bitrates.

For many users, being able to store between 5 and 10 times more music in the same space is an important saving, easy to translate if one takes into account the price of hard drives, flash memories or storage on iPods, tablets and the like. But there have always been two schools confronted: that of audiophiles who believe that nothing can equal the maximum quality of the CD and that of those who, with a more practical sense, consider the differences between an MP3 and CD ridiculous, if at all there are.

Atwood’s experimental study sought precisely to shed some light on these theories based on the basics: listening to music, quantifying its “quality” and deciding which is the best format based on the various variables. For this, he prepared five different audio files: one of them uncompressed and another four tablets at different bitrates between 128 and 320 Kbps. He put them on his server so that people could listen to them and vote (with a quality “note” of 1 to 5) without knowing which was which. And in total he got more than 3,500 people to contribute to the results – hundreds more than for many of the “quality studies” mentioned in the TV commercials.

The results were analyzed with a spreadsheet and various statistical tools, which showed trends and conclusions quite clearly:

The only sample that could really be considered very different from the rest was the MP3 at 128 Kbps CBR, the worst quality. That quality is not enough to compare with the rest. The best simply ignore it.

The MP3 at 160 Kbps VBR is the highest quality sample, even better than the MP3 at 320 Kbps CBR. This indicates that the coding with a variable bit rate is higher than the fixed one even at those values, and that 160 Kbps VBR up is impossible to improve qualitatively.
Ironically, this would indicate that there are MP3s that are heard “better” than audio CDs. Several things can happen here: that the “artifacts” created by compression seem to improve the audio or that when testing people “imagine things,” which could also happen. The truth is that the data serves to feed the theory that from 160 Kbps people no longer distinguish one quality from another, as it is deduced from the data.

The conclusion of the study confirms the hypothesis that an MP3 at 192 Kbps VBR has such quality that not even the ultrasensitive and powerful ear of a dog would notice the difference with an audio CD. Wow!
In conclusion, we already know at what rate to code and compress if we want a good saving in storage without losing quality: a MP3 of 192 Kbps VBR, the winning format of the test.

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.

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.

Mp4Gain is the Mp3Gain of this century ?

Mp4Gain is really something different , it works in a lot of audio and video formats . His internal algorithm has no relation with ReaplayGain , but has the option to apply ReplayGain if the person so desires .

But in reality its operation is very different, it is based on a post-2010 paradigm.
With hardware devices and software players we have today , with a previously unimaginable power , it was necessary to approach from another angle the process of normalizing audio , whether a song or a video.

In fact , adding the posibilidd to normalize audio from videos is a leap into the future . You stay ahead on how much to possibilities and options .

Today many people no longer collects mp3s, but uses the mp4 format for collecting favorite songs . Because it is able to play on a cell phone , a portable device in a tablet , a laptop, a desktop PC … then people have hechado hand mp4 to collect music or other kinds of videos.

The possibility of having storage media with capacities of terrabite very cheap indeed allow people to store thousands of music videos, with the classic problems of volume : each video sounds with different volume and one would like all sounded a loudness similar or homogeneous .

Mp4Gain can be used either to normalize mp3s, up to normalize all kinds of videos (the most popular formats like AVI, WMV, FLV, MP4, 3GP, etc.).
This goes hand in hand with other functions, like being able to apply EQ or modificr the tempo without affecting the pitch and vice versa.

That is, one can change the pitch of the video and the speed remains the same as the original, or you can change the BPM (Beats per minute) without changing pitch.

We could say that the Mp4Gain, is a real luxury, brought from the future, to be used today.

These are the formats that the software supports:

Video Formats:
mp4, flv, avi
mpeg, mpg
3gp, wmv.

Audio Formats:
mp3, mp2, flac
ogg, m4a, aac
wav, ac3.
That is talking about a:

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Mp4Gain 2017 is the new version, very powerfull at the same price !

We have now released Mp4Gain 2017 at the same price as the previous version. In addition, if you buy Mp4Gain 2017 you receive FOR A SINGLE PAYMENT OF $ 40 USD BOTH PROGRAMS, the “standard” version and the 2017 version.
Mp4Gain is definitely the alternative for the mp3gain for this 2017.

Mp3Gain clipping

Mp4Gain does not produce clipping, like mp3gain clipping, because the new algorithm is much more powerful, modern and effective.

Mp4Gain offers, in addition to the new ultramodern mode of volume normalization, also the ReplayGain (enhanced for 2017) and all other gamma of functions like changing the pitch without modifying the tempo and vice versa.

Change BPM (Change tempo without affecting pitch)

This function is so powerful that it allows to achieve amazing changes. Also the possibility to equalize the songs, as well as convert between audio and video formats.