Category: Mp3 Normalizer

Everything you wanted to ask about the mp3


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WHAT IS A CODEC AND WHAT IS IT FOR?

CODEC is an abbreviation of CODifier-DECodifier, which already indicates its functions. In the case of MP3, it deals with processing the sound, applying various compression methods to reduce the size necessary for its storage, and conversely, to reconstruct the sound from the encoded information. The codecs can be found in the form of an executable file, a DLL, or less frequently, incorporated into the code of another program. The DLL format allows the same codec to be easily used by different programs, while the executable format is usually more flexible in terms of configuration, although it is more complex to use.

audio codecs

WHAT DOES “PSYCHOCAUSTIC COMPRESSION” MEAN?

This very complicated word refers to the main compression technique that MP3 uses (although it is not the only one), which simply consists of eliminating the information (sound, in this case) that, in theory, is not able to perceive the ear human. I say “in theory” because, depending on the compression ratio that is applied, that loss may be more or less noticeable, hence some files do not sound as good as would be desired. The different codec also do not use psychoacoustic principles in the same way, so the quality they achieve will not be the same.

DOES THE SOURCE ORIGIN INFLUENCE THE MP3 QUALITY?

Evidently. The ideal is to start from an original CD. Analog sources, such as vinyl records or tapes, have lower quality in themselves, no matter how much we “digitize” or “improve” them with some programs. Recorded CDs may contain errors (which will end up being noises when played), and we usually do not know from which source they have been recorded (perhaps from previously encoded MP3 files). Re-encoding an already compressed music file, in the same or another format, is also a mistake, since quality is always lost.

DO THE PROGRAMS WE USE TO EXTRACT AND CODIFY THE QUALITY OF MP3?

It also seems obvious, so much so that this point should not be here. But many people do not take it into account, without knowing the reasons very well.

DOES THE QUALITY OF PLAYERS INCLUDE IN THE QUALITY OF MP3?

To hear it, you already know the answer, don’t you? J. To have it stored on the hard drive or on a CD, nope. Jokes aside, and in theory, all decoders should get the same sound from an MP3 file. But unfortunately this is not the case, usually because they do not strictly conform to the MP3 standard. You can find the best explanations on this topic here.

DOES THE QUALITY OF THE SOUND CARD AND SPEAKERS INCLUDE IN THE QUALITY OF THE MP3?

Well, I think you already know where the shots are going, we will not continue along this line. But nobody says “is that MP3 does not sound good” when it has a Sound Blaster 16 (what times those!) And some 800 W TurboPowerSound speakers that cost € 18 uros.

WHAT IS BITRATE?

It is the number of bits that are used to encode a second of sound, if we talk about MP3. It is measured in Kbps (kilobits per second).

As there is often confusion, we will give an example. If we have a 4-minute MP3 encoded at 128 Kbps:

-One second occupies: 128 Kbits x (1 Byte / 8 bits) = 16 KBytes.

-The complete song: 16 KBytes / second x (4 minutes x 60 seconds / 1 minute) = 3840 KBytes = 3.75 MBytes.

The formulas could be more correct, but this is not a mathematical thesis and I think they are enough to understand us.

WHAT DOES CBR, VBR and ABR MEANS, AND WHAT IS THE DIFFERENCE?

They are different coding methods. CBR means “Constant BitRate”, VBR “Variable BitRate” and ABR “Average BitRate”. It is easy to deduce that if CBR is used, the bitrate remains fixed, and if VBR is used, the bitrate changes. ABR can be considered a variant of VBR in which the objective is to achieve a certain final size, since we can set the average bitrate to be used (although it will never be as accurate as CBR).

The great advantage of VBR is that it uses more bits when the fragment to be encoded is more complex, and less when there are silences or it is simpler, so that we will achieve the highest quality using the minimum possible space. If the final size is not the deciding factor, it is the most recommended option.

 

WHAT IS THE BEST MP3 CODEC TO COMPRESS MUSIC? WHAT IS YOUR BEST CONFIGURATION?

The first question is easy to answer: LAME. In many different tests, done independently and with means that probably none of us will ever have, LAME is considered again and again as the best codec. And since there is no commercial motivation, since it is free distribution software, we can give full validity to these tests. The Fraunhofer codec and some of its “versions” could be valid in certain circumstances, but … do we need to use a commercial or pirate codec with a free and legal substitute? I think not (and from here, what each one does, is his own thing). For me, codec MP3 = LAME.

The second question is much harder to answer. There is no configuration that is perfect for everything (because if not, the modifiable options would be left over and everything would be much easier) J. But there are some recommended options in general, which we can establish with the assurance that the result will be of good quality. I have to admit that I barely change them, but as in this case (and in many others) there are no absolute truths, everyone has to try until they establish the one that best suits their preferences (or their ears).

The first thing that we must be clear about is that, in an MP3 file, size and quality are conflicting terms. We have to look for a practical middle ground. I put that term between 128 and 192 Kbps (some will say it is too low … I will try to justify my opinion). In principle, the purists established the CD quality at 256 kbps CBR. That is, using that bitrate, the sound of the CD and that of the MP3 file were absolutely indistinguishable. But, right now, LAME is already very developed. It uses the most advanced techniques, and manages to do perfectly things that other codecs are not able to solve properly (VBR, Joint-Stereo), which allows you to improve the quality and reduce the size.

In r3mix.net, after the corresponding tests, they have established an “ideal” configuration based on quality criteria. This configuration is so accepted, that in the latest versions of CDex, and other programs, it is already included as one of the preset configurations. But be careful: this does not mean that this is the only option, nor that it will always achieve the highest possible quality. Another thing is that using this option our ear can differentiate the results of the original audio CD, and this is where the “trick” of MP3 compression is.

After having used this configuration, I have observed that, although there are important variations, the average of the files is about 170-175 kbps, and that they rarely go beyond those 192 Kbps that I mentioned before, so I consider that overcoming that bitrate does not it will hardly improve the quality, but it will significantly increase the size. Although I repeat, there will always be someone to whom this bitrate seems insufficient, and will use even 320 Kbps, which is the maximum supported by LAME.

The second part is to take into account the equipment we have. If the MP3s are not going to sound through a card, an amplifier and very good quality speakers, the bitrate can be adjusted a little more without practically noticing variation in quality. The average 160 Kbps seems like a very reasonable amount to me, and that’s how I usually code my CDs. And I can say that they don’t sound bad at all. If someone is able to do a “blind” test and differentiate between several files encoded (correctly) at 160 and 192 Kbps, you really have to congratulate him on his ear.

Finally, if we consider size as a decisive factor, we could go down to 128 Kbps, but considering that the quality may already be too compromised. Summarizing:

Configuration that seems more appropriate for practical purposes, and looking for a high quality (about 192 Kbps):

-Joint-Stereo -> Get reduce the final size without losing quality. Sometimes, where there is a lot of variation between channels and with high bitrates, it could be configured as Stereo.

-VBR MTRH Quality 1 or R3Mix Preset -> Change the bitrrate according to the needs of each moment, improving the quality without increasing the size with respect to CBR (normally).

-Bitrate minimum 64 Kbps -> To compress to the maximum silences and very simple fragments.

-Bitrate maximum 256 Kbps -> So you have room to improve the quality in complicated fragments.

 


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Does the birate of an mp3 affect the quality or is it just an impression?

Since the mp3 appeared, I always understood that it is a lower format than the CD quality, no matter how much bit rate it contains. When it comes to mixing and producing my personal recordings, I have used wave support and when converting it to mp3 it is impossible not to distinguish the differences: at least, the reduction of bass and treble is very noticeable. The dishes of the drums or the bass are very opaque and hidden, sometimes they even disappear. But in these assessments I think that perhaps my subconscious has betrayed me in some cases.

bitrate

First of all I will explain some basic things about mp3 and its bitrate (or bit rate). Compressing sound means loss of quality, so you have to pay attention to how much information is transmitted per unit of time: the most used encodings for mp3 are 64, 128, 192 and 320 kbps (kilobits per second). Thus, a song of 64 will occupy less space than one of 128 and so on. Nor should we forget the CBR, constant bitrate, and the VBR, variable bitrate. The latter is more advisable when compressing different parts of a recording with various bitrates.

Well, there has always been controversy and confrontation between those who prefer to save space and opt for the musical amount, claiming that the quality differences of the mp3 are almost imperceptible to our ears, and those who bet on the qualitative details of the music, preferring heavier files and with higher bitrates, emphasizing the multiple peculiarities that our auditory system loses to lower quality media. I have been a supporter of the second group, although current opinions about it make me doubt my choice.

volume booster

I recently informed myself about a study conducted by programmer Jeff Atwood through his blog, in which he tried to discover if normal people, fond of music (but without becoming music experts) notice the differences between mp3 formats . More than 3,500 people participated in the study, who had to listen to 5 different audio files (with bitrates between 128 and 320 Kbps and one without compression) and vote from 1 to 5 depending on the quality they had received. Naturally, users were unaware of the characteristics of the media so as not to influence their objectivity. The results were as follows: the 128 Kbps CBR mp3 was undoubtedly considered the worst; the one of 160 Kbps VBR would be the one of better quality, surpassing even the one of 320 CBR (the variable bit rate would be higher than the fixed one). This is very curious since it is assumed that an original CD would house a quality between 192 and 256 Kbps. According to the study cited, the mp3 of 160 would have more quality than the compact, which seems absurd, so I tend to think that the Most people, after 160, do not distinguish some sounds from others, also taking into account that the subconscious can deceive us and make us imagine what it is not.

In short, the best way to compress, saving on storage and with optimal sound, would be thanks to the 192 Kbps VBR bitrate. From there, onwards, it is very difficult to appreciate nuances and alterations.

We must also consider something logical: if we recompress a file of 128 and convert it to 192 Kbps, improvements will not be achieved and we will lose space. If we do the opposite, go from 192 to 128 Kbps, we will reduce the quality somewhat but reduce the weight of the file.

I think this experiment obviates interesting data such as knowing the player or equipment from which the music was broadcast (computer, speakers, headphones, hi-fi equipment, etc.), listening time (our senses do not lend the same attention for 5 minutes than during 20) or musical style (electronic, rock, classical, etc.), since all these factors can greatly influence the final result.

Even, and this is proven: the volume (if the file is normalized) and the dynamics (a normalization like the one that only Mp4Gain does) manage to make one perceive the music as with higher quality.

Decibels: Understanding Decibels part 2

As you know the level of a digital audio signal should not exceed 0 db, in case it does distortions will occur. Normally the signal has an average level of -x db; but it is its highest peaks that should not exceed 0 dbs. That is, you can have a signal with an average level of -5 dbs but with a level of peaks of -2db, which prevents those peaks from raising the signal by more than 2 db.

decibels

Normalize is to increase the volume of a shot until the highest point of it is 0 dB.
It is assumed that when recording, for example, a voice shot, you have regulated the gain so that it does not saturate (0 dB pass). Most likely, that shot has not reached its maximum volume. When you normalize you climb to the maximum just before the highest peak saturates.
It is the same as raising the fader of the corresponding track until you reach 0 dB, but the normalize option calculates it automatically.
Necessary? Well, I guess not always. But it doesn’t hurt to know if you’ve reached the highest possible volume of a track.

There are cases in which it is very useful. Imagine that when you make a song you record a guitar shot and then start applying equalizers, compressors, etc. But you decide to make more shots, to see if they come out better (almost always this way). If each recording has different volumes, you must adjust everything for each shot (for example the compressor threshold). But if you have normalized after each recording, just replace one audio jack with another, since its volume is the same. In some posts you can read things like: “test set the threshold to -20 dB”. The one who gives that advice assumes that the track is normalized, because if it actually sounds much lower, what that compressor does will have nothing to do. In addition, some functions, such as noise reducers, are optimized for standard shots.
If you get used to normalize the newly recorded tracks, the values ​​that you will use in all types of plugins (or hardware tackle) will be similar. You can create your presets and apply them quickly.
It is assumed that normalizing does not change the quality of the recording.
Some people think that the background noise is also increased, but that is normal given that the volume is rising. The important thing is that everything goes up in the same proportion. Maybe you would have to do tests to check

What is a decibel?

A decibel (dB) is a unit of measure that is used to express, in a logarithmic way, the relationship between two values. In the world of acoustics it is about the relationship between two levels of sound pressure.

Decibel is composed of two words:

deci: that means one tenth.
bel: which is a unit named after Alexander Graham Bell, the inventor of the telephone.
The bel is a unit of sound and a decibel (dB) is one tenth of a bel.
So, in acoustics, a decibel is a unit used to measure the sound pressure level (SPL) based on a reference level.

The human ear is sensitive to a wide range of sound levels ranging from 0 dB, which means total silence for the human ear, up to 130 dB, which causes pain.

To get an idea:

-Smooth human breathing, which is heard only when we are very close, reaches about 10 dB.
-A normal conversation can be around 60 dB.
-A vacuum reaches up to 80 dB.
-An ambulance siren is around 120 dB when we are close to it.
-A volume of 140 dB can cause damage to the ear, if it is supported for a period of time.
-Exposure to 150 dB can burst the eardrums. And the sound above this level can be very harmful and even lethal.

Some details of the sample rate

For many years it was thought that the sample rate or sampling frequency did not decisively influence the final quality of the digital audio; There are currently several engineers who record in 44.1K or 48K without really knowing why they do it. With the advent of new and better computers, interfaces, ports and protocols, 88.2K, 96K and up to 192K entered the discussion table on the best sample rate to use. It has always been the subject of discussion between engineers and audiophiles; some argued that they did hear the difference between different sample rates and others that did not, and the topic has been subjected to millions of A / B tests with very high quality equipment, causing all kinds of opinions found and uncompromising, fights and friendships of years broken

samplerate

While this is a basic issue of digital audio, it is always surrounded by a halo of mystery, mysticism and magic (like every sound theme), which is well worth clarifying.

 What is the sample rate?

This topic, although it occurs in the first or second class of digital audio, is not always understood correctly. In scholastic thinking, sample rate is defined as the amount of audio samples transported and taken per second. Since this is a unit of measurement over a second and with events that occur cyclically, the Hertz (1 / Frequency) is used as a unit. Obviously we cannot talk about this subject without referring to the Nyquist sampling theorem, which was tested by Shannon almost twenty years after its publication and in which it is stated that for a signal of limited bandwidth (B) (for example, a vibraphone reaches 14.917Hz), the sampling frequency must be twice its bandwidth (2 * B). Then, taking the previous example, we can say that: 2 * B → 2 * 14.917Hz → The sampling frequency for 14.917Hz should be 29.834Hz. This would be equivalent to 29,834 samples per second (1/29, 834) to be able to regenerate the signal of a vibraphone without error. Hence, it is taken that the highest frequency that human beings listen to is 20kHz and if we apply Nyquist it should be 40kHz, but it takes 44.1kHz to meet the demanding ears and for a matter of multiples.

44.1K or 48K to 88.2K or 96K, the correct division

At the dawn of the digital audio era, Nyquist was used to use the sampling resolution of 44.1K, used at that time audio CD format that played at 16bit / 44.1kHz. With the advent of DVD and Blu Ray as video and audio formats, resolutions such as 24Bits / 48K or 24Bits / 96kHz began to be used. Although for many years there were recordings that were made in 24Bits / 88.2kHz or 24Bits / 96kHz, at a certain time of mastering, before sending it to the disk duplicator, the audio suffered a mutilation that reduced it to 16Bits / 44.1kHz as It was ordered by the CD format. This process should be carried out with equipment specially designed for this function and in stages so that the audio did not suffer a very noticeable cut and the bad conversion was evidenced. Although the old and dear Dither was applied since then to compensate for this process (something like “grain” in the cinema. Watch a film without “grain” and it will look like HD even though it was filmed in 1980 on tape and goes to notice until the makeup of the actor and the assembly of the special effects, something otherwise disagreeable).

Generally, to prevent the audio from mutilating or applying several conversions that degrade it, it was decided at what resolution to record before pressing the REC button (we will not mention those that come down directly with your DAW from 24Bits / 96kHz to 16Bits / 44.1kHz in one step to export the audio … there is a place reserved especially for them in hell). If the audio was going to end on CD, a 88.2kHz sample rate was generally applied, since at the time of mastering, with the symmetric re-sampling at “half”, it was 44.1kHz.

Sounds better?

The subjective point of this is that we expect recordings to “sound” better at a higher sample rate. The reality is that if we record in high sample rates, with very good sampling, our sound will not “sound better”, but will be more detailed. Obviously, if our sound source is bad, our microphones and preamps too and so on, no matter how much we record at 192K, the result will not be the best. Now, if we use a good sound source, good audio chain and a good converter, everything will be obviously good. But don’t confuse; We are talking about detail here, not if it will sound more “warm,” “fat,” or “full-bodied.” This translates into a more homogeneous capture of the entire frequency spectrum, both audible and non-audible.

sample rate

CPU, disk and plug-ins

Obviously, having a higher sample rate means that our processor must do more calculations, since it has to process more samples (or audio samples). Depending on the amount of plug-ins that we use before a multitrack in high resolution, our use of both DSP and native processors (the computer equipment), will increase significantly, making it very difficult or impossible to work. There are several options to overcome this problem, from buying more processor or DSP, using fewer processes or external equipment (hybrid mixing), to borrowing a machine. The only option that should never go through our minds is to lower the resolution of the audio, process and upload it again. The serious problem that comes with this is a cut in the audio, which is not reversible and what is limited and trimmed, so it stays.

Another aspect to consider is that the storage speed must be in accordance with the audio resolution we use. Suppose we want to record at 24Bits / 96kHz; The transfer rate would be: 2304kbits / second. Now, calculating the amount of tracks, we should use a disc that really reaches us in speed for this transfer rate (topic to be developed in another article).

In these times, storage size is not a problem, but speed is. Having three terabyte disk drives are generally used for 5400 rpm dish disks; the least that should be used if they are not solid state disks, would be 7200 rpm plate disc drives. Obviously, with 5400 rpm discs, we would have a third reduction in the final transfer speed and reading and writing possibilities called “iops” (in out per second or in and out per second), which have a certain number, depending on the disk, capacity and arrangement of the same (RAID) which, depending on how much we demand in the resolution of the audio, amount of channels, processing (plug-ins) and expected latency (if we record with real-time monitoring), we will surely face some problems like “clicks” and / or “pops” in our audio.

Clock

The importance of using a good clock (or clock) and being in sync with all the elements that belong to our audio chain is vital. Recall that a few articles ago we have exposed this topic in detail, but it should be reinforced in this article. Several ADC and DAC converters of economic interfaces do not perform sampling and quantization in the correct or expected manner; External clocks or protocols such as Dante help the synchronization between several devices to be correct and improve the audio quality. Much of the final quality of our work in audio is in this part of the process and it is important that if we take our work and passion seriously, we begin to pay attention to these kinds of details that are generally overlooked.

Audio Formats: Everything musicians should know to choose the right file

What is the best audio format? It is a very frequent question. Surely you’ve already raised it.

The answer is simple. It all depends on your needs. Whether you’re sending demos, building your digital music distribution, or archiving your songs, the file format is very important.

So, to help you choose the best file format for your music, we have collected all the essential information about the audio formats.

And even more important, which one is better in each situation.

Compression: the first impression

Audio formats depend on compression.

I don’t mean the compression you apply to a song in your DAW software. I am talking about file compression.

Compression makes a file smaller, to save space when streaming, downloading or storing.

But what happens when you compress?

There are 3 types of file compression:

Uncompressed (I know that “uncompressed” is not a type of compression, but I add it to make everything clearer), without loss and loss.

Uncompressed and lossless files retain the original data intact. But files with loss delete certain data from the original file to reduce the file size.

So the more compressed a file is lost, the more information is lost.

Compressing with loss does not mean that all your drums are going to be deleted. It simply removes the audio that the human ear is not able to hear. Maybe only dogs notice the difference …?

In any case, if you really want to hear what disappears when you compress a file, watch this MP3 conversion experiment.

About compression types
Here is a simple way to understand each type of compression:

An uncompressed file is an exact copy of the original. No information is lost. Uncompressed files are like an original picture.
Lossless files are slightly smaller files, but they keep the original information intact. A lossless file is like an original painting, but it is folded in two until you look directly at it.
Files with loss are the most compressed. Some of the original information is lost during compression. Files with loss are smaller versions than the original — the photo is still there, but some details have disappeared.


formats_c

Now that you know what compression is, you may be wondering how each type of file is compressed.

Do not worry. Here we go.

How each type of file is compressed

Uncompressed Formats

Uncompressed formats are not compressed (obviously). The most common uncompressed formats are WAV and AIFF.

These are the formats that you usually export from your DAW. If you duplicate a song to WAV, it is an exact and uncompressed copy of the original.

Lossless Formats
LANDR: A space to create. More details
Lossless files are compressed. But although they are compressed, they retain all the original information as a WAV. They simply unfold at the time of opening.

The most common lossless format is FLAC. Apple also has its own lossless format, called ALAC, used in iTunes.

The FLAC format makes the files lighter than WAVs, but they retain all the original information. Although the size of these files is usually very large.

Formats with loss

Lossy files are the most common audio format. The most used is the MP3. But there are other types, such as OGG, WMA and AAC.

The drawback of files with lossy compression is that it deletes some data from the original file.

But the benefit is that they are smaller, open faster and take up less space.

Files with loss can be high and low resolution, depending on the amount of compression. The higher the quality, the less information will be lost.

The truth about bitrates

The quality of an audio file is determined by its bitrate (bit rate).

The bitrate corresponds to the information processed per second. And that is what 320 or 192 means in MP3 files.

Thus, an MP3 with a bitrate of 320 has 320 kilobits per second — or kbps.

WAV and AIFF usually have 1411kbps.

A higher bitrate means more information per second. And more information per second means better sound. Simple, right? Now you understand the basic points of compression, file types and bitrate, right?

Perfect. Let’s continue.

Now comes the million dollar question …

 

In what situation do I use each format?
If I talked about each of the audio formats, we would be here for days. Surely you have other responsibilities, and a lot of music to produce. So I will be brief and concise. These are the best uses for each of these formats. We talk about WAV, MP3 320 and MP · 192.

WAV
The WAV is at the top of the podium. It is the Ferrari of audio formats. The WAV offers a cleaner and sharper sound than the other compressed formats. If you share demos with a record label, show your work for a possible audiovisual project or send your music to a blog, you need a mastered WAV.

1512-38_mixtape-700×366

The WAV is a guarantee that your best sound represents you.

When you master your music, always use the WAV as the delivery format.

WAVs can also be converted to other formats later, so it is the right format for conversion later.

The only drawback of WAVs is the large size of the files. They take up a lot of hard disk space. So your computer, your phone, your iPod or your Dropbox will fill up very quickly if you only use WAV.

But when it comes to your own music, it is important to always have a WAV copy of each of your tracks.

Most platforms require WAV to upload your music for distribution. For example: iTunes and Amazon ask for high-quality WAV to upload music to their services.

The 320 MP3
The MP3 of 320 is the most frequent type of file. For one simple reason: It has the best of both worlds.

They are compressed, so they are easy to handle in regards to their size. But they also offer a pleasant and rich sound.

If you listen to music in streaming, it is very likely that it is 320. For example: everything you hear in high quality on Spotify is at 320kbps.

The MP3 of 320 is a good way to share your best sound saving space on the hard drive and avoiding long waits during the upload and upload.

MP3 192
The 192kbps MP3 is the draft horse. They are fast and dirty MP3, for when you have to share something easily and quickly. They are useful when transferring a handful of files at once, check your entire catalog or share and reference tracks quickly.

A lower bitrate causes more degradation than an MP3 of 320 with loss, but sometimes it is difficult to feel the difference. Take the test and judge for yourself.

The MP3s of 192 are the perfect tool for musicians who need an efficient and fast way to share or listen to their music in streaming.

Useful tip: if you use your own streaming player on your web page, an MP3 of 192 will make your page load faster.

Don’t forget any format by the way
Each format has its uses. Choosing the right format depends on each context.

So think about what sound you share and where you do it. Are you using the right format?

Mastering in WAV format is the best bet to share your music. Once you have the mastered WAV, you can convert it in any other format into a periquete.

Formats are important in the era of streaming. So make a smart choice and use the right format.

Most used audio formats

In this article we will talk about the most popular Audio Formats and used for both Music production and Entertainment.

Lossless Audio Formats (lossless)

Also known as Loosless, they are audio formats without loss of sound, its main feature is that it resembles original audio with excellent quality.

Its main disadvantage is the size of the file since when capturing it and converting it into digital audio format, files from 50 to 100 megabytes become depending on the number of minutes and resolution.

Normally it is usually captured with a resolution of 16 bits at 44,100 hertz which is the audio quality handled by a compact disc.

Although today it is usually captured at 48,000, 92,000 and up to 192,000 herts that can vary to 16, 24, 32 and up to 64 bits.

WAV (audio file format)

Undoubtedly this is the best known and most standard format, it was created by Microsoft and is called audio formats without signal loss.

Its main feature is that it closely approximates the capture audio signal, which gives us great sound quality. Therefore it is an audio format widely used for audio editing with Music Production Softwares.

Its main disadvantage is its weight is that 5 minutes of audio is approximately 50 megabytes, which becomes a format that takes up a lot of storage space.

AIFF (audio interchange file format)

This format was created by Apple, it is also an audio format without signal loss.

It closely resembles the original sound at the time of capture but also occupies almost the same space as the Wav.

CDA (audio cd track)

Perhaps you have seen this format before, however it is not exactly an audio format rather it is a shortcut which is part of the audio data of a compact disc. This format is created when you burn a CD with some software such as Nero or Alcohol 120%.

When exploring an original compact disc on your computer you will find these files with that extension.

However, when copying these files to your computer you will not be able to reproduce them since they are only small shortcuts that do not contain the music information, they are only used by stereos, homemade modulars or cars.

In order to have these files you will have to convert this format to one that is readable by your computer, which can be Wav or Aiff or other audio formats with some CDA conversion software to other formats.

FLAC (free losseless audio codec)

This audio format is the most popular since it has no signal loss but with compression, which greatly decreases the file size apart from the fact that it is possible to reverse the compression to have the original version again.

It works very similar to the ZIP file but optimized for audio with the advantage that you don’t have to unzip it in order to listen to it.

Its main disadvantage is the incompatibility to reproduce it in any audio software or mobile devices, therefore you must have installed an application capable of playing it with the appropriate codec.

audio codecs

Audio Formats with Loss (loosy)

Also known as Loosy, they are audio formats with sound losses but with compression as the main feature since these files are reduced considerably such as a Wave or Aiff file could be compressed from 50 to 5 megabytes.

Being compressed by means of some audio compression codec eliminates all irrelevant and imperceptible information for the human ear.

You should practically have a fairly sensitive ear to be able to differentiate losses.

These formats are mainly used for portability in Usb memories, Compact Discs or audio devices such as Ipod, MP3, Cell phones, Tablets, Lap Tops and even Desktop Computers and some car stereos or CD, DVD, Blue Ray players, among others.

 

MP3 Audio Format

It is the most popular and commercial format used since the 1990s, precisely in 1993 created by the Fraunhofer IIS Institute by Karlheinz Brandenburg.

Its main feature is its low data weight and its compatibility with most audio playback software such as Itunes, Windows Media Player, the extinct Winamp among others.

However, it should be mentioned that this format when compressed eliminates audio information that may be almost inaudible by the human ear.

BITRATE (BIT RATE)

It is practically the one that determines the quality of the MP3 that can be from 8 to a maximum of 320 kbps (kilobytes per second).

It is usually compressed at 128 kbps which is a standard for internet radio, streaming platforms such as Soundcloud or Spotify that usually convert to this quality by default.

At the lowest amount of kbps, streaming is guaranteed to be fluid for most internet connections, but it also loses sound quality.

Although considering this happens to second term since many of the users usually reproduce it with headphones in the street, school, public transport, motorcycles or computer speakers in offices at low volume.

If you are a professional DJ I do not recommend mixing with this format since you will not get the best results in amplified sounds, preferably acquire the versions without loss of signal to strengthen your live presentations.

CBR AND VBR METHODS

The CBR (constant bitrate) method practically converts the audio with the constant dynamic range, while the VBR (bitware variable) method varies its dynamic range.

The advantage of the CBR method is that it converts with greater speed and a lighter audio file is obtained although its disadvantage is that its quality is diminished.

The advantage of the VBR method is that when the audio file is converted, a higher sound quality is obtained since it retains the dynamic range, that is, if your theme has sounds with lower and higher intensity, they will continue to obtain a more dynamic sound.

Another significant advantage is that you can include metadata such as ID3, which you can store information such as the title of the song, the name of the band or artist, year, album and a small image which can be the cover that when reproduced in Itunes or Windows Media Player will be displayed.

AAC format

It is a compression audio format with loss of signal which produces a sound very similar to the original.

The ACC format is an extension of MPEG-2 created by MPEG (moving picture experts group), it is often used for Internet radio broadcasting and by Apple for Ipods with Itunes software, also used by Ahead Nero, Winamp and Nintendo DSi.

Another use is for encoding MP4 videos and recently for the new STEMS audio format created by Native Instruments which is a multitrack format used by DJs.

WMA Audio Format

We cannot ignore the WMA (windows media audio) format that was basically the alternative to MP3.

On this format I will not deepen as it is very little used and incompatible in most mobile devices.

It is only compatible with devices such as Windows Phone, Xbox and Microsoft operating systems. However, if you download a multimedia application for example on your Android phone that supports WMA, then you can listen to it.

The WMA audio format has 3 versions:

WMA Pro: Supports multiple tracks like AC-3
WMA Lossless: Compress with loss of signal
WMA Voice: Aimed at human voices which is used
on Skype and WhatsApp.

AC3 Audio Format

The AC3 audio format also called Dolby Digital which is an alternative to MP3 since it allows the inclusion of multiple audio tracks generally 5.1 for Surround surround audio

Youtube to mp3: convert from youtube video to mp3

Youtube to mp3

The whole world wants to be able to download music videos from YouTube and convert them into mp3.
The idea is quite simple, since in YouTube all the artits upload all their ideas and almost always with HD and high musical quality.

But the next thing would be to convert them from mp4 to mp3, but our software does that automatically. You only select the songs you want to download, even a playlist or a complete channel and the software downloads it and converts it into the format you want.

Convert YouTube to mp3 with just one click?

Yes, just choose the video, the playlist or the channel and it gives you “download2 and the program does everything without you having to do anything.

The graphic interface of the program is very beautiful and modern and works very well.

And the program gets it for free, as a gift, when you buy Mp4Gain.

Mp4Gain is a program that normalizes the audio volume of the main audio and video formats.

You can store thousands of mp3s downloaded from YouTube or store them in other formats such as flac, ogg, m4a, etc.

Today converting to mp3 from YouTube is very simple if you use our free program.

What is the difference between MP3 and FLAC

What is the problem with the popular MP3 or why FLAC is of better quality? Actually, in addition to these 2 formats, there are others with different specifications that differentiate them from each other.

In this article we will talk about the most important formats and their characteristics, in this way you can choose the one that suits you.

What is bitrate?

The difference in the size of the files is noticeable

You’ve probably heard this term before, but if you don’t remember it or don’t know what it means, I’ll explain it. Bitrate is translated as the bit rate, that is, the amount of bits or data that are processed in a unit of time. When referring to audio, we usually use the measurement of kilobits per second (Kbps). For example, the music available on iTunes is usually 256 Kbps, which means that there are 256 kilobits of information stored in every second of the song.

The higher the bitrate of a file, the more space it occupies on your hard drive. Usually, a music CD uses a large amount of memory space, so since the early 1990s, researcher Karlheinz Brandenburg developed the audio compression standard we now know as MP3. When an audio file is compressed, the space destined for it is smaller, however, this means that it loses part of its sound quality. For that reason, the different audio formats available are divided into 2 categories: with losses and without losses.

Lossless Formats

Lossless or lossless formats, as they are known in English, retain the sound quality of the original source intact. This means if you compress a CD in any of these formats, your digital music files will have the highest quality.

FLAC. Free Lossless Audio Codec is the most used sound codec, making it one of the best options for storing music without losing quality. Unlike other formats such as WAV and AIFF, the compression it uses decreases the space it occupies. In addition, it is free and free software, so anyone can use it and is compatible with a considerable number of players.
WAV and AIFF. These 2 formats do not use compression, so it could be said that they are exact copies of the original sound source. Although both are of the same quality, what differentiates them is the way they store the data. AIFF was created by Apple, while WAV is used universally. The problem with these formats is that since they do not have any compression, they take up too much memory space. They are normally used in audio editing.
A THE C. Apple’s lossless format is very similar to FLAC because it uses compression, the detail is that it was created by the company founded by Steve Jobs. Although its compression method is not as efficient as that of FLAC (it occupies more space), the advantage is that it is compatible with iTunes and iOS devices.
Formats with losses

Lossy, or lossy, formats are the most common, most systems and players recognize them and compress a lot, which allows large amounts of music to be stored in a small space. One advantage is that if your bitrate is high enough, it will be difficult to distinguish the difference you keep with the original source.

MP3 Its name without abbreviation is MPEG Audio Layer III, and it is the most popular lossy format, so much that it has become the benchmark in the category of digital or downloadable music on the Internet. Although it is not the most efficient format of all, it is the one with the highest compatibility, so it is the first option to choose between lossy formats.
AAC Advance Audio Coding is similar to MP3, although a bit more efficient in compression, which means that your files take up less space while matching the quality of the MP3. Although not as popular, it has the advantage that Apple players support it.
WMA As the name implies, Windows Media Audio is the proprietary format of Microsoft. It is very similar to the 2 previous formats and does not really offer any advantage over them. The disadvantage is that it is not so popular, so it is not compatible with many systems.

Which one should you use?

After having reviewed the main formats of both categories and their characteristics, we conclude that the type of file to use will depend on the habits and tastes of the user. In general, MP3 and AAC are widely recommended, because they are compatible with most players and the quality of both becomes quite good if it is encoded with a high enough bitrate (192 Kbps onwards). If your demands in audio quality are not the highest, any of these 2 formats is ideal thanks to the little space they occupy.

Storing your music collection in FLAC will require many GB of storage

However, storing music for posterity in MP3 or AAC may not be the best idea, since converting from one format with losses to another with losses will produce files with much lower quality. On the other hand, FLAC is the best for preserving music for the future, because it will keep the quality intact and allow converting to other formats in the future, in addition to occupying less space than WAV or AIFF.

If your interest is to enter the world of audiophilia, without a doubt lossless formats are the option. Thus, it is necessary to consider that to notice the difference between a high quality MP3 file and a FLAC, you must have high fidelity headphones or speakers and have a trained ear. Of course, the storage issue is also important, the amount of gigabytes or terabytes you will need to store all your music will depend on the size of your collection.

In conclusion, the format you intend to use will depend on your musical demands, both in quantity of music and in its quality. You just have to remember that the important factors are compatibility and avoid converting between 2 formats with loss and, above all, enjoy music in the way you like best.