What is the difference between the different audio formats, and which should I choose?


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What is the difference between the different audio formats, and which should I choose?

Lossless formats: WAV, AIFF, FLAC, Apple Lossless and others

In summary, there are two types of sound quality: lossless and lossy. Lossless music preserves the sound quality of the original source – in most cases, CD – intact, on the other hand, lossy music compresses the file to save space (in exchange for decreasing quality). Lossless formats include the following formats:

WAV and AIFF: Both are uncompressed formats, making them exact copies of the original sound source. The two formats are essentially the same quality; they simply store the data differently. AIFF was created by Apple – you will see it often in its products – but WAV is much more universal. However, since they are not compressed, they take up too much unnecessary space. Unless you are editing sound, we do not need to use this format.

FLAC: Free Lossless Audio Codec (FLAC). It is the most used lossless codec, it is a good option if we want to store our music without losing quality. Unlike WAV and AIFF, it uses compression, taking up less space. However, it is still a lossless format, which means that the sound quality is the same as the original source, so it is better for listening than WAV and AIFF. It is also free and free software, which is useful if you like to take a look at how it works.

-Lossless Apple (Apple Lossless): Also known as ALAC, it is similar to FLAC. It uses compression, although it is made by Apple. Its compression is not as efficient as that of FLAC, so the files will be slightly larger, but it is compatible with iTunes and iOS (FLAC not). Therefore, if you use iTunes or iOS as the main software for listening to music, you should choose this format.

APE: It is a very high lossless compression file, which means you will save more space. The quality is the same as FLAC, ALAC and other lossless files, but it is not compatible with most players. On the other hand, it makes the processor work more to decode because it is so compressed. Generally, I would not recommend using this format unless you are very concerned about space and have a compatible player.

Lossy formats: MP3, AAC, OGG and more

For day to day, we are more likely to use lossy formats. They save significantly more space, and if they have a high enough bitrate, it will be very difficult to distinguish the quality of the original source. The most commonly used formats are shown below:

MP3: MPEG Audio Layer III, or MP3 for short, is the most common lossy format. So much so that it has become synonymous with music downloads on the internet. It is not the most efficient f-format of all, but it is certainly the most compatible, making this the first choice of lossy sounds.

AAC: Advanced Audio Coding, also known as AAC, is similar to MP3, although slightly more efficient. Which means that the files take up less space and with the same sound quality as MP3. And, with Apple’s iTunes making it so popular, it’s as compatible as MP3.

Ogg Vorbis: The Vorbis format, often known as Ogg Vorbis due to the use of the Ogg container, is the free software version to MP3 and

AAC. Its main attraction is that it is not restricted by patents, but that does not affect you as a user – in fact, despite being open it is of similar quality, and much less popular than MP3 and AAC, so not all players support it. . I do not recommend it unless you are interested in being open source.

WMA: Windows Media Audio. Microsoft’s proprietary format, similar to MP3 or AAC. It really doesn’t offer any advantage over other formats, and it’s not very well supported.

So which one should you use?

Now that we have seen the differences between each format, which one should we use for our music? In general, we recommend using MP3 or AAC. They are compatible with most players, and the quality of both is very similar to that of the original source if encoded with a high bitrate. Unless you have specific needs, MP3 and AAc are the most recommended options.

However, there is something to be said for storing music in lossless formats like FLAC. Although we probably won’t notice higher quality, it is good for storing music if you plan to convert it to other formats later – as converting from one lossy format to another. Lossy (eg from AAC to MP3) will produce lower quality files.


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Introduction to digital audio

Introduction to digital audio

Digital audio is the representation of sound signals through a set
of binary data. A complete digital audio system usually begins
with a transceiver (microphone) that converts the pressure wave that represents the
Sound to an analog electrical signal.
This analog signal goes through an analog signal processing system, in
which can be made limitations on frequency, equalization, amplification and
Other processes such as compassion. Equalization aims
counteract the particular frequency response of the transceiver used of
so that the analog signal closely resembles the original audio signal.


After analog processing, the signal is sampled, quantified and encoded. The
sampling takes a discrete number of analog signal values ​​per second
(sampling rate) and quantification assigns discrete analog values ​​to those
samples, which means a loss of information (the signal is no longer the same
than the original). The encoding assigns a sequence of bits to each value
discrete analog The length of the bit sequence is a function of the number of
analog levels used in quantification. The sampling rate and the
number of bits per sample are two of the fundamental parameters to choose from
when you want to digitally process a certain audio signal.
Digital audio formats try to represent that set of samples
digital (or a modification) of them efficiently, so that it is optimized
depending on the application, either the volume of the data to be stored or the
processing capacity necessary to obtain the starting samples. In
in this sense there is a very extended audio format that is not considered audio
digital: the MIDI format. MIDI does not start with digital sound samples, but
stores the musical description of the sound, being a representation of the
score of them.
The digital audio system usually ends the reverse process to that described. From
the stored digital representation is obtained the set of samples that
represent. These samples go through a process of digital analog conversion
providing an analog signal that after processing (filtering,
amplification, equalization, etc.) affect the output transceiver (speaker)
which converts the electrical signal to a pressure wave that represents the sound.

Fundamental parameters of digital audio

The basic parameters to describe the sequence of samples it represents
The sound are:
ƒ The number of channels: 1 for mono, 2 for stereo, 4 for sound
quadraphonic, etc.
ƒ Sampling rate: The number of samples taken per second in each
channel.
ƒ Number of bits per sample: Usually 8 or 16 bits.
As a general rule, multichannel audio samples are usually organized in
frames A plot is a sequence of as many samples as channels,
each one corresponding to a channel. In this sense the number of samples per
second matches the number of frames per second. In stereo, the channel
Left is usually the first.