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Mp3 Normalizer

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Mp3 Normalizer

Why do we need to normalize an mp3?

Mp3 Normalizer

We need an mp3 normalizer because any user who has a lot of mp4 files or even other audio formats like flac, ogg, m4a etc. (Because Mp4Gain can normalize the main audio and video formats), but why?

For the simple reason that coming from different websites, from different sources, we not only find significant differences in volume level, but also in bitrate, etc.

This means that when playing them on our computer or on other devices, we find that these significant differences in the volume level force us to correct it manually using the volume knob.
Which is neither comfortable nor ideal.

Mp4Gain, as we mentioned, not only do mp3 normalizer, to modify the loudness. But it does the same with the most used audio and video formats, it can even extract the audio from any video and normalize it at the same time.

Mp3 Normalizer 2022

In general, the need to normalize the volume of audio and video files remains in 2022.

Search for mp4 normalizer, avi normalizer, etc. has been added lately. That is, the video normalizer has become a necessity.

Mp4 Normalizer

As we pointed out, Mp4Gain is perfectly capable of normalizing the volume level of videos in the main formats and is both an audio and video converter.

Download it, try it and discover how it improves your audio and video files.

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Mp4Gain Main Window

Mp4Gain Features

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How to normalize mp3 files to play at the same volume

What could we understand by normalizing the audio volume level of an mp3 file or any other format?

Mp4Gain : Mp3Gain alternative : Audio converter Video Converter ...

A very technical definition would be to say that what is sought is to ensure that the upper volume level or the volume peaks of a song do not exceed certain levels.

That would be the answer that many people would give, but that would take us 20 years ago. When the normalizers were very simple and basic.

Today the technical explanation would be much more advanced and it would be something like this:

What a modern normalizer seeks is to ensure that the points with the highest volume level, also known as “peaks”, do not exceed a certain level, but at the same time they must not be below another level X.
In other words, the volume peaks of an audio file must not be outside a parameter, nor higher than one level X, but also not lower than another level X1. Which means that the peaks of a song will be very close to those of the other songs, with which we achieve that the sonority or loudness is very similar.

And if we are going to think about an even more efficient normalizing program, we could think that the lowest levels will not be outside of the x2 and x3 range either.

So we should never have to turn the volume knob to raise or lower the highest volume peaks of a song … nor should we turn the knob to raise parts of a song that sound very low.

In summary, the song (actually, all songs) will sound within a volume range, without falling below a certain level or rising from another level. What will make the activity of listening to music pleasant.

This is only achieved by Mp4Gain, which is also capable of doing all the important audio and video formats.

How to normalize mp3 files to play at the same volume? We hope that the brief explanation given above has been sufficient to achieve an adequate response.
And that helps to understand why some normalizers are definitely superior to others and more modern and efficient.

Because many people only think of volume peaks when thinking about a normalizer, but “sounding at the same volume” also implies that the lower volume parts are similar among all mp3s.

High and low volume parts will have similar levels.

Does bitrate influence? A 320 kbps Mp3 sounds better than a 128 kbps one?

Much has been speculated about the bitrate. Most people do not understand clearly what it is. A few understand, but almost nobody knows if a file with 320 kbps really sounds different or better than the same file but with 128 kbps.

The easiest way is to test:

The first is at 128 kbps

Now let’s hear the 320 kbps option

Notice the difference? In case the note is because it was encoded using the Mp4Gain.
Normally it is almost impercentible, but using a good encoder you get to notice some subtle difference.

It should be taken into account that at higher kbps, if there is a higher quality – although it is not always noticeable – and will always use more disk space.

Therefore it is not the best option to say “all my mp3s will be 320 kbps”, unless the space does not mean any problem at all.

How MP3 files work

The MP3 movement is one of the most incredible phenomena that the music industry has ever seen. Unlike other similar phenomena, such as the introduction of cassette tape or CD, MP3 technology did not start with the industry, but with a huge audience of music lovers on the Internet. The digital MP3 music format has had, and will continue to have a great impact on how people collect, listen and distribute the music.

If you have wondered how MP3 files work, or simply want to know what uses can be given, read on. This article will give some features of this popular sound format.

MP3 format

If you know something about how CD’s work, then you know how they store music. A CD stores a song in the form of digital information. The data on a CD uses a decompressed high resolution format. This is what happens when a CD is created:

The music is sampled (fractionated) 44,100 times per second. Each of these parts has a size of 16 bits.
Pieces of these fractions or “samples” are taken from the left and right channels in a stereo system.
With a simple formula we realize how great a single song can be.

Fractions * bits * channels = X bits per second

In our case it would be 44,100 for 16 bits per 2 channels, which would give us 1,411,200 bits per second. 1.4 million bits per second equals 176,000 bytes per second. If the average of a song is 3 minutes, then the average of a song on a CD is 32 million bytes of space. That is a lot of space for a song, and it is especially great if we consider that we are downloading music with a 56K Modem, which will take us a few hours.

The MP3 format is a compression system for music. This format allows you to reduce the number of bytes in a song without damaging the sound quality. The goal of the MP3 format is to compress a CD quality song without letting you see the difference. With MP3, a 32 MB song from a CD, compresses up to 3 MB. This allows you to download a song in minutes instead of hours, and store hundreds of songs on your computer’s hard drive.

Compression and quality

Is it possible to compress a song without damaging the quality? To perform this compression, the use of algorithms is needed, in the same way that we use them to compress other formats, such as graphics, text files, applications, etc. A very popular algorithm for compressing sound is the “perceptual noise shaping” technique. This algorithm uses characteristics of the human ear such as:

There are certain sounds that the human ear cannot hear.
There are certain sounds that the human ear hears better than others.
Its there are two sounds playing at the same time, we can hear the one that is louder, and not the lowest.
Using factors like these, certain parts of the song can be eliminated without significantly damaging the quality of the song for the listener. When you have created the MP3 file, what you have is music with a quality close to that of a conventional CD. It doesn’t sound exactly the same because some things have been removed, but it’s very close.

Using the MP3 format

The MP3 movement – consisting of the MP3 format itself and the ability of websites to distribute it – have done several things in the music world:

It has made it easy for anyone to distribute music at a low cost, or even for free.
It has made accessing music simple and instant.
He has taught people to manipulate music on a computer.
One of the strengths of this format is the ability to edit, create and modify music files thanks to powerful computer software tools. Thanks to these tools, it is extremely easy for anyone:

Download an MP3 file from a website and play it instantly.
Transform or “rip” a song from a CD, to the MP3 format, and listen to it later.
Record a song yourself, convert it to MP3, and make it available to everyone on the Internet.
Convert MP3 files into CD files and make your own audio CD’s with MP3 files downloaded from the Internet.
Have thousands of hours of music stored on one or more hard drives.
Upload MP3 files to portable players and listen to them wherever you want.
To do all this, all you need is a computer with a sound card, speakers, an Internet connection, a CD / DVD player / recorder, and an MP3 player.

Audio quality: Bitrate in MP3 files

In many cases, the term Bitrate is used, which is the bit rate per second that a multimedia file (Audio or Video) has. Currently the MP3 music format is one of the most widespread (Although there are currently other more current formats such as OGG Vorbis, AAC, Flac, Monkey Audio, …) however the audio quality is variable, this is due to the characteristics with which the MP3 in question has been compressed, including:

Mode: It can be of two types mainly:

Mono: With a single channel (The right and left channel go together, not separated which gives worse audio quality).

Stereo: Two channels (Right and Left, improve audio quality).

Sampling frequency:

Audio CDs use 44,100 Hz (22,050 Hz per channel), although there are higher frequencies such as 48,000 Hz used in DVDs and lower, the higher the frequency, the higher the quality.
Bits: Audio CDs have 16 Bits (Although MP3 can be compressed at a lower quality such as 8 Bits).
Bitrate (Bit Rate per second): Audio CDs have about 1,400 Kbps (44100 Hz * 16 Bits * 2 channels), meaning that an Audio CD would have a bitrate of 1,400 Kbps (In MP3 format the maximum Bitrate is 320 Kbps, however, it is assumed that an MP3 with a 128 Kbps Bitrate has a quality similar to CD, although in many cases to achieve a quality similar to CD it is necessary to use a Bitrate of 192 Kbps, and to obtain CD quality it is necessary use 256 Kbps or 320 Kbps).

Some of the most common Bitrates are:

8 Kbps Mono: Telephone Sound.
16 Kbps Mono: Better quality than shortwave.
32 Kbps Mono: Better quality than AM.
64 Kbps Stereo: Better quality than FM.
112 – 128 Kbps: Quality close to CD.
160 Kbps: Quality closer to CD.
192 Kbps: Virtually CD quality.
256 Kbps: Quality CD practically undisputed from an original CD.
320 Kbps: CD quality.

Coding method: It can be of two types:

VBR (Variable Bit Rate, Bit Rate Variable): Encodes the file in MP3 with a variable Bitrate.
CBR (Constant Bit Rate, Constant Bit Rate): Encodes the MP3 file with a fixed Bitrate.

In addition, another factor that influences the encoding of the MP3 file is the CODEC (Encoder-Decoder) used, one of the most common and the best result is LAME (Lame Ain’t an MP3 Encoder) which is also free.

One point to keep in mind is that if we recompress an MP3 file that originally has a 128 Kbps bitrate and convert them to 192 Kbps for example, audio quality is not really gained because the MP3 format has some quality loss (MP3 is a loss algorithm, also called lossy). which has occurred when converting the original file (Ex: CD Audio or a 320 Kbps MP3 to a 128 Kbps MP3) so this recompression does not make much sense since we will not gain in audio quality (As they say where there is no one can not get) and the only thing we will achieve in any case is to increase the initial size of the file.

The opposite case (Recompress a 320 Kbps MP3 file for example at 192 Kbps) if it makes some sense because in this case although we lose some audio quality we reduce the weight (Kilobytes or Megabytes) of each MP3 file somewhat.

In conclusion, it can be said that if we need to encode / compress an MP3 file with good quality, the “ideal” would be to do so:

To be able to start from an Audio CD, although an MP3 at 320 or 256 Kbps could also be valid for a recompression of the file.
In stereo mode (With two channels, right and left).
With at least 44100 Khz sampling rate and 16 Bits.
With a minimum bitrate of 192 Kbps or at most 256 Kbps (Using 320 Kbps would give higher quality but also increase the file size considerably).
Use the LAME Codec (Lame Ain’t an MP3 Encoder).

What it is and how to perform a volume normalization on your MP3

Have you ever heard the term audio normalization, without being sure of what it meant? As a lover of music and technology, I also encountered such a doubt many years ago. Basically, giving a short definition, it is about the standardization of the volume, or rather, of the audio spectrum with respect to other subjects, usually of the same disc.

And that, to put it more simply, is the equalization of the volume of the different tracks on a disc. The reasons are many, and usually if the tracks are extracted from the same job they already have the same volume and gain, but what happens if we want to make a mixtape? For example, we decided to make a compilation called The Best 100 Rock Songs in History. Surely have songs from The Beatles or The Rolling Stones, and therefore from different albums. Depending on the year, type of mastering, etc. etc., we can end up with a CD that contains many different volumes, something that can be annoying when listening. That is just one of the reasons to normalize our MP3 collection.

There are add-ons for players that allow us to normalize on the fly. In fact we can say that programs like Spotify already do this by means of the option to equalize volume of all the songs, however the application that I present below allows us to permanently normalize modifying MP3 files and many other formats, both audio and Of video..

This is Mp4Gain, which stands out for its simplicity of use and is presented under an interface that is ideal to understand exactly what a normalization is and see the before and after. When we open the application we find a window in which we have a grid, which will be populated when we add files or folders, and a keypad with various options.

How do we normalize? Simply change the gain through the specific menu for this.

By pressing OK the application will start working and save our files with the same gain, so it is ideal that before doing the first tests we make a backup. It must also be taken into account that it is an operation that can take time, something that depends on the speed of our processor, the number of issues to normalize and also the size and quality of them.

Audio normalization

The normalization of the audio level is something that is achieved by applying a constant and maintained amount of gain, in volume, to an audio recording to bring the average peak amplitude to a desired level that has been previously defined. To which the same amount of gain is applied to the entire range, the signal-to-noise ratio generally does not change. Normalization differs from dynamic range compression, which applies different levels of gain to a recording so that the amplitude is within a minimum and maximum range. Standardization is one of the most common functions provided by a digital audio workstation.

Peak normalization

One type of normalization is peak normalization, in which the gain is changed to bring the highest PCM value or the highest peak of an analog signal to a given level.1

Since it only searches for the highest level, it does not take into account the apparent volume of the content. As such, peak normalization is generally used to change the volume in such a way as to ensure optimum use of the distribution medium in the mastering stage of a recording. loudness normalization.

Normalization of loudness

Another type of normalization is based on a loudness measure, in which the gain is changed to bring the average amplitude to an objective level. This average may be a simple measurement of average power, such as the RMS value, or it may be a measure of the loudness perceived by humans, such as that offered by ReplayGain.

Depending on the dynamic range of the content and the target level, the normalization of the loudness can lead to peaks that exceed the limits of the recording medium. Some software has the option of using dynamic range compression to avoid saturation when this happens. In this situation, the signal-to-noise ratio is altered.

Modern Audio Normalization

Currently Mp4Gain uses an audio normalizationn that is more similar to that used in modern recording studios or live music group recitals.

It is a normalization of volume focused from a new perspective.

Under this new paradigm, not only does it achieve that all songs have the gain of loudness at the best possible level, but it also achieves that each instrument and / or voice obtains a level of gain that makes it audible. Achieve an optimized level of volume gain normalization.

There is no other normalizer in the market that obtains this level of result. People with training in hearing listening can easily notice the difference., very similar to that obtained with expensive hardware in radio stations or in recording studios or in recital consoles, combining limiters, modern compressors and other processors.
All these results that offer expensive hardware equipment, Mp4Gain does for a few dollars.

In fact, the opposite result is achieved than that achieved with masking, because with masking, which is a method used to compress music, you can no longer perceive some sounds that are behind a more audible sound, that is what is called masking, which leads to the loss of audio quality.

Mp4Gain manages to highlight hidden instruments and sounds, performing an audio normalization by frequency bands to achieve this.

That is why we say that Mp4Gain achieves the same results as those obtained through a series of hardware equipment (limiters, compressors, normalizers, etc.) that are very expensive, while Mp4Gain costs only a few dollars.

Digital Audio – Beginners guide

The Cost of a High Sampling Rate

Although it is true that high sampling rates produce better sound quality … that comes at a price.

That price translates into:

Higher processing load.
Less number of tracks.
Heavier audio files.
So you always give something in return. Professional studies can support higher sampling rates because they use better equipment.

But for most home studios, people often find that the standard 48 kHz configuration is the best.

Following…

4. Bit Depth

In order to understand what bit depth is, we first have to know what bits are.

A bit (or binary digit) is a single unit of binary code, with a value of 1 or 0.

The more bits, the more possible combinations. For example…

As you can see in the diagram below, 4 bits allow a total of 16 combinations.

4 bits

When used to encode information, each of these numbers is assigned a specific value.

As the number of bits increases, the possible values grow exponentially.

4 Bits = 16 possible values
8 Bits = 256 possible values
16 Bits = 16,536 possible values
24 Bits = 16,777,215 possible values
With the bit depth in the digital audio, each value is assigned a specific amplitude of the waveform.

The greater the bit depth, the greater the volume increase between high and low … and a greater dynamic range in the recording.

A good rule of thumb is: for every extra bit, the dynamic range increases by 6dB.

For example:

4 Bits = 24 dB
8 Bits = 48 dB
16 Bits = 96 dB
24 Bits = 144 dB
In the end, what this means is that… the greater the bit depth, the less noise.

Because by adding more processing margin (or headroom), the useful signal (at the high end of the spectrum) can be recorded higher above the background noise (at the low end of the spectrum).

small vs large bit depths

Following…

5. Quantization Noise

Impressive that a 24-bit recording can result in almost 17 million possible values, right?

However, that remains much less than the infinite number of possible values that exist in an analog signal.

Therefore, in almost all samples, the actual value is somewhere between two possible values. The solution of the converter is simply to round it or “quantify” it to the nearest value.

The resulting distortion, known as quantization noise, takes place in 2 phases of the recording process:

at the beginning, during the A / D conversion, and
at the end, during mastering
With mastering, the sampling frequency / bit depth of the final track is usually reduced by converting to the final digital format (CD, mp3, etc.).

When that happens, some of the information is erased and “re-quantized”, generating more distortion in the sound.

The most frequent solution to deal with this problem is …

6. Dither

When reducing a 24-bit file to 16 bits, the screen is used to mask much of the resulting distortion …

Adding a low level of “random noise” to the audio signal.

As it can be difficult to visualize the concept in audio, to explain it, we usually turn to the popular analogy of the screen plot.

Is that how it works:

When a color photo is converted to black and white, a mathematical estimate is made to determine if each color pixel should be “quantized” in a black pixel, or a white one …

As is the case when digital audio samples are quantized.

As you can see in this picture, the “before” photo is pretty bad, right?

dither

But with the plot …

a small number of white pixels are randomly distributed in black parts, and …
a small number of black pixels are randomly distributed in white parts …
By adding that “random noise” to the image, the “after” photo looks much better. Well, the screen in the audio works very similarly.

Following…

7. Latency

The GREAT PROBLEM of current digital studies is the amount of latency that accumulates in the signal chain, especially with DAWs.

With all the calculations that are processed, the audio signal takes time to leave the system between a few milliseconds and a few DOCENAS of milliseconds.

Between 0-11 ms of latency – it is short enough, so a normal person does not notice it.
Between 11-22 ms – an annoying delay is heard which it is difficult to get used to.
More than 22 ms – there is so much delay that it is impossible to play or sing at tempo with the track.
In a normal digital signal chain there are usually 4 phases that contribute to the total latency:

A / D conversion
DAW Buffer
Delay of the Plugins
A / D conversion
The A / D and D / A conversion are the least harmful, contributing to total latency with less than 5 ms.

But nevertheless…

The DAW buffer and certain plugins (including compressors and virtual instruments) can add up to 20, 30 or 40 ms or

Beginner’s Guide to Digital Audio for Recording Music

62c-digital audio When recording at home began to become popular …

It happened for a simple reason:

The analog equipment of the past decades was being slowly but inexorably replaced …

For a new generation of audio interfaces and other digital equipment that was cheaper and easier to use.

And that trend has continued since then.

Today … digital audio is the standard in almost all studios, both professional and amateur.

However, surprisingly, there are few people who really understand what it is about.

So let’s see what it is about:

1. The Rise of the Digital Age

binary code Although digital audio is the standard in today’s music …

It has not always been that way.

Originally, music information only existed as sound waves in the air.

Then, as technology progressed, people discovered new ways to convert that information to other formats, including:

notes on a page
electrical signals inside a cable
radio waves in the atmosphere
relief on vinyl records
But in the end, with the rise of computers, digital audio ended up being the dominant format in the music production industry, since it allowed copying and transporting songs in a simple and free way.

And the device that made all that possible was … the digital converter.

Let’s see how they work …

2. Digital Converters

In recording studios there are 2 types of digital converters:

Those that are an independent device, which are normally seen in more advanced studies, or …
Those that are integrated into the audio interfaces, which are usually seen in home studios.
To convert the audio to binary code, they take tens of thousands of samples (samples) per second to make an “approximate” image of the analog waveform.

The image is not accurate because in the intervals between samples, the converter basically has to guess what is happening.

Digital waveform

As you can see in the diagram, in which:

the red line is the analog signal, and …
the black line is the conversion …
The results are not perfect, but they are good enough to generate excellent sound quality.

How excellent? That depends largely on …

3. Sample Rate

Check out this image:

sample rate diagram

As you can see…

When taking more samples per second, the highest sampling rate:

Collect more real information,
Go less to the estimate, and
It generates a much more accurate image of the analog signal.
Logically, the end result is … better sound quality.

Let’s talk about specific data:

Normal sampling frequencies in professional audio range around:

44.1 kHz (audio CD)
48 kHz
88.2 kHz
96 kHz
192 kHz
The minimum of 44.1kHz is due to a mathematical principle known as …

The Nyquist-Shannon Sampling Theorem

To record digital audio accurately, converters have to capture the entire human listening spectrum, which is between 20Hz – 20kHz.

According to the Nyquist-Shannon Sampling Theorem …

To capture a specific frequency, at least 2 samples are needed for each cycle … to measure both the upper and lower points of the sound wave.

That means that recording frequencies of up to 20kHz require a sampling rate of 40kHz or more, which explains why the audio CDs are just above that minimum, at 44.1kHz.

What is an audio compressor.

In the field of professional sound, a compressor is an electronic sound processor designed to reduce the dynamic range of the signal without noticing its presence too much. This task is done by reducing the system gain, when the signal exceeds a certain threshold.

Traditionally, compressors have been electronic equipment with one or two rack units, but software versions of them have appeared for some years.

A compressor acts in such a way that it attenuates the electrical signal by a certain amount (normally measured in decibels) and from a certain input level. The objective is to ensure that the resulting dynamic excursion is lower than the original, to protect certain equipment against possible signal peaks or, if it is a saturated sound, to try to hide the error.

Reasons to compress a signal

-Control the energy of the signal: The human ear is very sensitive, so the compression must be smooth and subtle so as not to capture it. This type of compression is used when there is a signal in which the intensity varies, so it is compressed to achieve a more constant signal within the values assigned to it.

-Control the peak level of the signal: Often the equipment is limited, so the amplifiers can saturate and therefore be damaged. In this case the compression is used to control the signal and thus protect the equipment.

-Reduce the dynamic range of the signal: By attenuating the peaks of a signal, we reduce its dynamic range. Many devices are limited by the peaks, and this allows the RMS level of the signal to be raised.

Compressor Uses

In the field of music, its use ranges from applications for musical recordings to live sound. For example, it is often used to add more glued to the sound, an effect that is achieved by compressing the signal to subsequently apply a gain to the output of the device, which usually conceals possible interpretation failures by the artist, at least as Dynamic control refers. A compressor is highly recommended (and with certain musical styles, indispensable) for when using an electric bass. The slapping effect (hitting the strings with the finger) produces extremely high output peaks (20 dB or 10 times more than normal), which at low output levels generate distortion, and at high volumes (as in recitals) they can cause serious damage to the amplifier, and even the speaker (an excess of “excursion” can cause the speaker to tear from its suspension). Even in the (theoretical) case of a musical system with an infinite dynamic range, the difference, auditory speaking, using or not the compressor is imperceptible. Its use is also very frequent in voices, since not all singers use the appropriate technique so the signal level varies constantly.

-It is widely used in broadcasting, to improve the speaker’s diction.
-Compress during mastering improves the sound definition of the final mix.
-To protect the equipment (speakers).