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MP3, FLAC, WAV, ALAC: the differences between audio formats


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Digital audio formats

Digital Audio

Today, most people listen to music completely digitally. The differences between digital audio formats like WAV, FLAC, MP3, and ALAC are not clear to everyone. We put the facts together.

Digital audio formats

While vinyl is booming and CD sales are slowly but surely falling, today’s music is often heard without any physical medium. Whether you use your smartphone or digital audio player, you can move forward with digital audio formats on the go. After all, no one today wants to carry a Discman and multiple CDs with them when they typically have a powerful pocket computer in the form of a smartphone that can play digital music files. But what are the differences between the individual file formats and what are their advantages and disadvantages?

WAV and AIFF: the uncompressed ones

The Wave container format (.wav) was developed by Microsoft. Saves uncompressed audio content, so files require a lot of storage space (2 minutes can take 20MB of space. WAV is especially important when recording and editing audio content. The downside of .wav files is that they don’t metadata is required (about, Title Artist) can be stored,
the equivalent developed by Apple AIFF (.aif) Due to the fact that Apple computers are very common in music production, this audio format is very common there.

MP3, AAC, WMA, Ogg-Vorbis – compressed to save space, but not lossless

The MP3 file format (.mp3, named for the MPEG-1 Audio Layer 3 compression codec) developed by the Fraunhofer Institute in the 1980s is probably the best-known digital audio format. It gave the MP3 player its name, and for a long time music was digitized almost exclusively as MP3, for example, on the extremely popular and now illegal file-sharing networks around the turn of the millennium. The advantage of MP3 is the small amount of storage space required: on average, it takes up one-tenth the size of the original file. However, one disadvantage that should not be neglected is that it is lossy – frequencies that are inaudible to humans are removed to drastically reduce the memory required. To what extent this affects the sound, you can compare Flac with MP3 Read.

AAC (Advanced Audio Coding) is a successor to the MP3 format, offering slightly better sound quality. Apple continues to mainly offer songs in this audio format on the iTunes store.

WMA stands for Windows Media Audio (.wma), as the name suggests, a development by Microsoft. .Wma is also a lossy compression file format.

A somewhat rarer audio format is Ogg-Vorbis (.ogg), where Vorbis is the music compression technology and .ogg is the container format. Like MP3, .ogg is also lossy, but requires less storage space and better quality.

FLAC / ALAC / WMA lossless – the lossless

Lossless formats were developed to preserve all sound information while keeping the amount of memory required small. With all file formats, the required memory is reduced to about half the original file. With audio conversion software, the file can be converted to other lossless formats, something unthinkable with lossy formats. This is why lossless file formats are popular for archiving music collections in a space-saving way.

FLAC – Free Lossless Audio Code (.flac) is a free audio format, so it is not owned by any major corporation. ALAC: Apple Lossless Audio Codec (.alac) is Apple’s lossless file format, while Microsoft also has its own development on the market with WMA Lossless.


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Basics of digital audio

Basics of digital audio:

Before the computer can record, manipulate, and reproduce sound, sound must be transformed from an audible analog form to a computer-acceptable digital form, using a process called analog-to-digital conversion (ADC). Once the sound data has been stored as bytes in the computer, the power of the computer’s CPU can be used to transform this sound in thousands of ways. Finally, when you are ready to listen to the result, the digital-to-analog conversion (DAC) process transforms the sound bytes back into an analog electrical signal from the speakers.

Sampling: Analog to Digital Conversion

Given an analog signal, discrete values ​​of its amplitude are taken at small time intervals, obviously the more reliable the reproduction the more samples per second are taken. These obtained values ​​are assigned a digital value that the computer can understand and process as required. We can use 8 or 16 bit words, thus obtaining 256 or 65536 different combinations and obtaining higher resolution.

 

SAMPLE FREQUENCY: According to the Nyquist theorem, it is possible to accurately repeat a waveform if the sampling frequency is at least twice the frequency of the component with the highest frequency. The highest frequency that the human ear can perceive is close to 20 kHz, so the 44.1 kHz sampling rate of sound cards is more than enough. This value is the one used today by CD audio players.

SAMPLE SIZE: The sample size controls the dynamic range that can be recorded. For example, 8-bit samples limit the dynamic range to 256 steps (50 dB range). In contrast, a 16-bit sample has a dynamic range of 65,536 steps (90 dB range) a substantial improvement. The human ear perceives a whole world of differences between these two sample sizes. Ears are more sensitive to detecting differences in pitch than intensity, but are even more sensitive to the strength of sound.

From the previous processes we can get an audio file, such as (and since it is the best known), a WAV audio file. It is the own format of Windows. They can be 8 or 16 bit with sampling rates of 11,025 kHz, 22.05 kHz, or 44.1 kHz and generally have good sound quality.

Digital audio compression

It could be assumed that all you have to do to get good sound is to record at the 44.1 kHz speed limit with 16-bit (2-byte) samples. The only problem that appears if recording in stereo, sampling simultaneously on the left and right channels at 44.1 kHz, a one minute sound sample needs a 10.58MB storage space. This involves using large disk spaces to store these sound files. Many compressed file formats (codecs) have been developed that enable high-quality recording without the need for so much disk space.

Most common audio formats:

With the simple objective of listing a series of codecs used by different operating systems to perform audio compression. Later, a more complete description of the most used is made: MP3.

Therefore, some of the most used are:

Advanced Audio Coding (AAC): used by Apple computers. More efficient than MP3.

Audio for Unix (AU): Acoustic standard for the JAVA programming language.

Windows Media Audio (WMA)

Ogg Vorbis: It is free, open and not patented.

Atrac: compression and playback technology for minidisc.

 

The codec par excellence: the MP3

Its origin and current

The abbreviations MP3 respond to the abbreviation of MPEG (Moving Picture Expert Group) 1 Layer 3, which is a perceptual coding algorithm. This among others was developed by the Moving Picture Expert Group (MPEG) (http://www.cselt.it/mpeg/) together with the Fraunhofer Institute of Technology (http://www.ipa.fhg.de/english/ ).

Moving Picture Expert Group is an ISO / IEC research committee. MPEG is in charge of the international development of compression, decompression, processing and encoded rendering standards for movies, audio and the combination of both. It is a non-profit institution created in 1988, which brings together 300 experts from 20 countries three times a year.

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.