From the list of audio file formats e. What is the AUDIO extension? How to fix problems with AUDIO file formats

It is immediately clear that we should talk ONLY about the hidden characteristics and do not include any details. Maybut Lifehacker will conduct a more objective investigation. And today we’ll try to uncover the same evidence as before.

Є analogue that figure.

Analogue - good, but not very good and not easy. Therefore, analogue noses, regardless of high vinyl sales, will not turn around.

Audio figures can be of three main types:

• in a format that does not have vikoryst’s constraint;
• in a format that is like a vikoryst’s squeeze without waste (lossless);
• in a format that has a vikory squeeze due to waste (lossy).

On the graphs there is a good AudioCD, a version of OGG with a variable bitrate of 350 kbps and an MP3 with Lame vicors. The lower the graph is scaled, the closer the sound is to the original. It's a great picture to come out. Regardless of the fact that MP3 may clearly cut high frequencies, it is replaced by OGG, in which a drop below 2 kHz is visible.

The frequency-hour division of the sound speaks about no less important speech. With a steady bitrate of 320 kbps, MP3 may be comparable to the original recording. It seems that now everything is falling into place. Ale... In truth, one is still more confused.

We are currently in need of lossy, if lossless is available

Healthy glutton.

On the right is that most analog recordings do not accommodate the same amount of information that would need to be saved in high-quality formats. Don’t forget that the typical sampling frequency for CD is 44.1 kHz, quantization is less than 16 bits.

The front graphs cannot demonstrate the high fidelity of MP3 transmission. Even for an audio cassette, a magnetic strip (which is, of course, not a master tape), the characteristics of an AudioCD are unattainable. For mass studio use, the ability to record analog sound, as evidenced by AudioCD, has recently appeared. There is no need to digitize into FLAC (and especially into WAV) a concert recording or recording from the pre-digital era, especially those made from magnetic media. They do not contain these spectra and as much information as containers can store without constraint.

What has changed today

A rare sound engineer will use a digital master recording (from which reproduction is then carried out on physical media), vicoristic and current technologies again. Therefore, the chance that a 24-bit track is actually inferior to a 16-bit track is extremely high.

Analogue high-acidity recording on high-acidity-owned modern instruments is even more complex - as fans do not have a similar sound. This is, for example, Jack White, ex-leader of the White Stripes. When some of your recordings go up to the lo-fi variation, and discover the dim sound characteristics of the track, it becomes quite satisfying for gourmets.

Once you find the ideal output, then only training your hearing or listening on clear audio equipment will allow you to find the correct file. І already spiraling onto the center (і), warto create an offensive arch:

Necessary and sufficient for mid-priced users is AAC, without any (and for the absence of output codes that can be encoded in AAC) - MP3 with a constant bitrate of 320 kbps, creations with the additional Lame codec 3.93 (recommended keys for decoding: cbr -b320 - q0 -k -ms).

It is important to note that recordings are primarily taken from a high capacity, say, recorded on DVD-Audio, SACD or recordings, primarily collected in DSD (or a similar format) with a high bitrate.

I want some special features from lossless. And we know about them just now.

Today there are about three dozen of the most advanced digital audio formats. It is now necessary to create as many types of audio files as possible to save one type of content and as a whole you learn from this material.

Enter

Of course, many people are willing to use their home computer not only as a work device, but also as a multimedia center on which you can watch movies or family photos, and also listen to your favorite music. Although, of course, compact digital players or mobile phones are better suited for listening to musical compositions, rather than being controlled from them, a computer can not only play music.

No matter how great the task of recalled memory is, your music player, which is worth everything, will hardly be able to save your entire music library in it. Moreover, with the help of a PC you can create, edit, arrange and play music. It’s also important to remember that today there are about three dozen advanced digital audio formats, and most players are far from universal, and even more than a dozen of them are created today.

So is it really necessary to create so many music formats to save one type of content? It’s all about the fact that the sound of most of the episodes is saved in a “sweet” look, since one piece of compressed composition takes up about 10 MB on the hard drive. On the one hand, this is not much, but on the other hand, if you are a music lover and your collection consists of hundreds or even a thousand songs, it becomes clear that the sound needs to be compressed in order to change the place it occupies on the electronic nose іх information.

To compress music files, various special algorithms are used to determine the structure and features of the presentation of sound data or so-called digital audio formats files. All audio formats can be divided into three groups: audio formats without compression, without compression without waste, and with compression without waste.

Without a squeeze

One of the most widespread formats that belong to this type is the well-known WAV. The sound of files with such extensions is preserved without any restrictions or changes. True, the place to save compressed files requires a lot more, and the greatest use of WAV is found only in professional audio and video add-ons, where the sound before processing is not to blame for wasting it. Saving the most important musical compositions seems to be an unjustifiable waste of money.

To create WAV files, you do not need any special software, since this format is accepted by all media players, including those installed in the Windows system with the standard Windows Media audio file player.

Another format that is used to save uncompressed audio is one developed by Apple called AIFF (Audio Interchange File Format). As you already guessed, it is most often used on Macintosh computers running Mac OS X systems.

Squeeze without wasting (lossless)

Algorithms that compress audio files without wasting money follow the principle of ultimate archivers. It is possible to ensure the highest level of compression (from 40 to 60%), while it is practical not to interfere with the harshness of the sound. It also means that in this case the encoded data can be restored to its original form. Therefore, perfect compression without waste most often stagnates in cases where it is important to preserve the identity of the compressed data to the original.

The most popular audio formats in this group are FLAC (Free Lossless Audio Codec), APE (Monkey's Audio), WMA (Windows Media Lossless) and ALAC (Apple Lossless Audio Codec). Each of them has its own advantages and disadvantages. For example, the APE codec gives much greater compression gain, while FLAC gives greater compression. By the way, all serious music lovers save their music collections in lossless formats, since the data in them is not deleted from the audio stream, and files created using these codecs can be listened to online on high-quality sound equipment.

To create these files without wasting formats, as a rule, third-party players (except WMA) are used, such as MPlayer, foobar, AIMP, Winamp, VLC and others, since all codecs are already built into them. Another option is to install a package of additional codecs (for example, K-Lite), after which listening to files in lossless format becomes accessible practically from any audio programmer.

Constraint due to expenses

This is the most popular group of algorithms that ensure the maximum (up to 10 times or more) level of sound compression. Although it is a change from previous formats, here the audio file is used up, and how strongly it is necessary to lie below the stage of its compression.

To increase the clarity of digitized sound, the following display is most often used: bitrate- The velocity of the sound stream that emerged after compression and damping in kilobits per second (kbps). As we have already said, the average volume of compressed audio takes up about 10 MB, which corresponds to an audio stream of approximately 1400 kbps. After encoding with the expense of your data, the bitrate may drop to 56 kbps. In this case, it is important that in order to preserve the natural sound, the speed of the stream must be no lower than 192 or 256 kbit/s. If the bitrate of the stream becomes 320 kbps or more, then the difference in what most people hear between compressed and uncompressed audio is practically clear.

The most popular format here is definitely the famous and popular MP3, developed by members of the MPEG (Moving Picture Experts Group). It is most widely used for encoding audio files that are located on the Internet and various file hosting services due to the ability to quickly change the size of the data that is transferred, even when the connection speed is low. It is not at all important.

Other popular formats in this series include AAC (Advanced Audio Coding) and OGG Vorbis. In this case, being less popular, their algorithms are more thoroughly compressed than their main competitor. So, with a new file size, they will ensure that the sound quality is equal to that of MP3. Another significant advantage of these formats is the ability to encode up to 48 audio channels for AAC and 255 for OGG, versus just two for MP3.

This means that the WMA format is the property of Microsoft, which was initially created to save and broadcast audio information without waste, and encoding without wasting energy has recently reached a new level, starting with Windows Media Audio 9.1. Nominally, this format will provide a lower level of compression than MP3, allowing retailers to offer it as an alternative to competing AAC and OGG algorithms. The truth is that the wide range of WMA is important for its closure and availability on many platforms (operating systems). This increased support for the digital copyright management system (DRM) does not add popularity to Microsoft's creation.

Despite the fact that MP3 loses out to its competitors both for the efficiency of compression and for the clarity of sound, it still continues to be deprived of the most popular audio format. The secret of such success, one can say, can be called the banal inertia of thought, which has been the result of many years before the majority of manufacturers, equipment manufacturers and software distributors. Moreover, MP3 files can be listened to on anything that can process digital sound - a mobile phone, a personal computer with any popular operating system, a portable audio player, a current music center or a DVD player.

And although other formats still cannot boast of such encouragement, everything is not so bad with them. Thus, AAC is widely supported by Apple for its algorithms for storing audiobooks, podcasts, music from the iTunes store and ringtones. So for regular Macintosh computers, iPad tablets, iPhone smartphones and iPod players, this format can be used as a “real” format.

WMA files can be easily opened on any PC running the Windows operating system, which is the most widely used in the world. However, many manufacturers of portable audio players and stationary optical disc players also support this format. And for listening to files in OGG Vorbis and AAC formats on Windows systems, you will have to install special codecs. I don't want it to be a problem. Installing the all-new K-Lite Codek Pack will allow you to process almost any audio files on your computer using your favorite programmer.

Visnovok

Finally, let's see what kind of software you need to turn your home computer into a universal tool for working with audio files. For clarity, we will divide all the add-ons into a number of main groups.

Players - serve for the easy creation of sound files, and are also often used for cataloging and organizing music collections. The thickness of the table is great, you can’t beat it. Anyway, to make it a little easier for you to select, we will point out, in our opinion, the twelve most popular: Windows Media Player (installed in the system), Winamp, KMPlayer, iTunes, GOM Player, JetAudio, VLC Media Player (VideoLAN), AIMP, BSPlayer , Real Player, WinDVD and Foobar2000.

Convertory - Add-ons that allow you to convert from one format to another. For this purpose, you can use most popular players without having to worry about using special programs. I want to be able to cope without someone in such situations.

Riperi (graberi) - allow you to extract digital audio information from optical media (Audio-CD, DVD) and save it in different formats. Regardless of the number of powerful grabbers, the most popular in this field is the EAC (Exact Audio Copy) add-on, which allows you to make exact copies of discs. Other popular rippers include: Audiograbber, Reaper, Easy CD-DA Extractor and others.

Editors - programs designed for creating, recording and editing sound data. This group will have access to simple programs that allow you to perform basic operations on an audio file (verify, trim, merge, normalize, etc.), as well as other monsters for professional work. with sound. Among small editors, you can see the Nero WaveEditor add-on for its modest size and yet high functionality. The most popular professional solutions for sound processing include: Adobe Audition, Sound Forge, Cubase, Sony Vegas Pro and others.

Of course, purely theoretically, all these necessary functions can be combined in just one program, but in practice, creating a single add-on for all tasks is not always easy. It is practically impossible to achieve clear execution of all tasks using one program.

At any time, a richer mother has at hand a number of specialized accessories that take up less space and cope with their tasks more efficiently.

It is the main audio format for many digital audio systems and is considered the standard format for audio files on personal computers. In addition, there is a solid set of specifications that will be replenished in the remaining hours. Its name is Microsoft RIFF/WAVE – Resource Interchange File Format/Wave – resource transfer file format/format, created by Microsoft and Intel engineers. WAV stands for Waveform Audio File Format.

WavPack (extended WV)

WavPack also includes a unique “hybrid” mode, which provides all the advantages of compression without waste with an additional bonus: instead of creating a single file, in this mode a relatively small high-quality file with waste (.wv) can be created. It can be programmed by itself, as well as the file “corrections” (.wvc), which (combined with the front.wv) allows you to completely update the original. For some consumers, this means that they will never have the opportunity to choose between pressure without wasting or wasting power.

The format is very suitable for developers. Of course, the plugin for Winamp can be downloaded from the official website of the codec http://www.wavpack.com for Adobe Audition (!) and Nero Burning Rom. A standalone plugin for XMMS – an analogue of Winamp for Linux. In addition, there is a project underway to develop DirectShow filters for WavPack - which will allow you to change the format in any Windows program, including Windows Media Player. You can download the remaining version. WavPack uses a frontend from third-party vendors. Yogo hitaemo here.

From a technical point of view, there is also nothing to understand. It supports multi-channel sound, 32-bit separate audio stream, sampling frequency – up to 192 kHz (!).

Format, such as the output codes of the programs for compression, and the open ones. It is a great pity that the compilation encoder is no longer available for the Windows platform, so you will no longer be able to independently recompile the available output codes for your operating system.

The site, like the codec itself, is updated regularly, which is good news.

DTS – Digital Theater System, in essence – is not Dolby Digital, or rather its competitor. The DTS format has a minimal level of compression, less than Dolby, which actually makes it sound better, which makes it practical for DVD discs that record tracks in DTS or in DD format. DTS in home theaters has a maximum bitrate of 1,536 kbps (full bitrate), such sound is clearer than AC-3 format in Dolby Digital. DTS - contains 6 large audio tracks and supports the system with 7.1 sound, at this bitrate DTS 4.0 - sounds as spacious as "Dolby" 5.1.

Windows Media Audio (WMA)

The file format is licensed and developed by Microsoft to save and transmit audio information.

Nominally, the WMA format is characterized by excellent compression, which allows it to “compress” the MP3 format and compete in terms of parameters with Ogg Vorbis and AAC formats. However, as was shown by independent tests, and also with a subjective assessment, the range of formats is still not clearly equivalent, and the advantage over MP3 is clear, as confirmed by Microsoft.

The codec is part of the Windows Media Audio package. Vіn bezkoshtovny, prote format, as well as the program for coding is closed. Obviously, the version is only for the Windows platform.

In principle, WMA Lossless supports all the necessary functions: tags, high-frequency sampling, multi-channel audio (including 7.1), broadcasting audio through a network, etc. In addition, decoder support has been introduced into Windows Media Player, eliminating the need for users to download anything from the Internet to create a music file.

MP3 - (MPEG audio encoding format) - a file format that is licensed to save audio information.

The most popular squeeze format for today. The MP3 format (MPEG Layer 3) was developed after a number of intermediate formats by the Fraunhofer Institute in Germany. Vzagali, format.MP3 based on the deception of the human ear. After several studies, it became clear that human hearing has the power to adapt before the appearance of new sounds, which is detected at an increased threshold of sensitivity. Therefore, some sounds of the building are masked (in order to work subjectively inaudible) and others. Axis and in this format, some of the sounds, which, according to the underlying theory, are inaudible, simply taken from the usual sound. After that, the “fabricated product” is coded according to the Hoffman method. It is important to note that in the MP3 format, programs that compress sound from the original are not standardized, so that a competent programmer can implement his own compression scheme. And decoders are also subject to standards, which means that the quality of the MP3 format will not always remain in the hands of the programmer that plays the file. Due to the different abilities and similarities of the implementers of different coders, some of them are better at handling symphonic music, others at rock and metal, others at rap and rave etc.

JointStereo, which is one of the features of MP3, means that instead of encoding stereo as two independent channels, audio encoding is carried out. the center channel and the difference that differs from the output stereo channels. Adding a lot of storage to the sound in the stereo channels, however, and its encoding in the front channel allows you to add additional fluid for more detailed encoding of the difference, which leads to a certain enhancement of the viscosity.

It’s important to know about Variable Bit Rate and VBR. This means that the encoder changes the compression level by degrees, depending on the nature of the sound. This approach can lead to a change in the size of the file or, with increased bitterness, while the size of the file allows you to achieve a shorter sound.

The MP3 Pro codec, which appeared in 2001, was created by Coding Technologies in collaboration with Thomson Multimedia. It is based on MP3. This one uses SBR (Spectral Band Replication) technology, for which the codec provides high brightness at low bitrates. However, the encoding capacity at medium and high data rates is compromised by the capacity of many other codecs. As a result, MP3 Pro is more suitable for broadcasting on the Internet and demonstrating fragments of new musical compositions.

Vorbis is a free format for compressing sound at a cost, which officially appeared in 2002 rock. The psychoacoustic model that is being developed in Vorbis, the principles of operation are close to MP3 and similar, mathematical processing and practical implementation of this model are completely different, which allowed the authors to announce their format not stale from all the predecessors. To save audio data in the Vorbis format, the Ogg media container is most often used, such a file is called the extension.ogg and is called subordinate to Ogg Vorbis.

As of 2006, there are significantly fewer extensions, lower than MP3. According to all estimates, there is another for the popularity of the format of sound compression at a cost. Widely used in computer games and file sharing services for transferring musical works.

Vorbis offers a more clearly psychoacoustic model than its competitors, which gives better clarity of performance at equal flow strength.

The format does not separate the player with more than two audio channels (stereo - left and right). Supporting up to 255 adjacent channels with a sampling rate of up to 192 kHz and a resolution of up to 32 bits (which other cost-constrained formats do not allow), Vorbis is ideal for encoding 6-channel audio DVD-Audio.

Moreover, the Vorbis format is “sample accurate”. This guarantees that the audio data before encoding and after decoding does not interfere with any additional or lost samples. This is easy to appreciate if you encode non-stop music (if one track moves step by step to another) - the result will preserve the integrity of the sound.

The format has initially expanded due to the possibility of streaming communication. This allows the format to have a significant side effect - in one file you can save a large number of compositions with valid tags. When you select such a file, the player will display all the songs that were imported from different files.

The format has a flexible tag system. The header tags are easily expanded and allow you to include texts of any complexity (for example, the text of a song), interspersed with images (for example, a photograph of an album cover). Text tags are saved in UTF-8, which allows you to write multiple words at once and avoids possible coding problems.

Ogg Vorbis uses a variable bitrate, in which the remaining values ​​​​are not limited by any hard values, and can be changed to 1 kbps. Please note that the format is not strictly limited by the maximum bitrate, and with maximum adjustments to the coding, you can vary it from 400 kbps to 700 kbps. The sampling frequency is equally flexible - users can choose between 2 kHz and 192 kHz.

Vorbis is breaking up the "Xiphophorus" audio format in order to replace all paid proprietary audio formats. Regardless of the fact that it is the youngest format among all MP3 competitors, Ogg Vorbis is gaining widespread support on all major platforms (Microsoft Windows, GNU/Linux, MacOS, PocketPC, Palm, Symbian, DOS, FreeBSD, BeOS, etc.), as well as There are a large number of hardware implementations. Popularity today significantly outweighs all alternative solutions.

FLAC (English: Free Lossless Audio Codec) is a popular free codec for compressing audio. By replacing codecs using Ogg Vorbis, MP3 and AAC, it does not remove any necessary information from the audio stream and is suitable for both personal listening and archiving of audio collections. Today, the FLAC format is supported by a wealth of audio add-ons.

FLAC is a member of the Xiph.Org family of codecs. Before speaking, before entering the new ogg vorbis – one of the most lossy algorithms for music compression. As a container for audio data, it is obvious that OGG (files with extensions.ogg) and another open-source container is Matroska (files with extensions.mka).

It is immediately clear that both the format and the FLAC algorithm are open again. The stinks are not patented, so they can absolutely harmlessly get involved in any kind of programs. There is also a wide support for FLAC in the program - no matter how serious the program, there is a plugin for FLAC. In addition, there are hardware mp3 players with support for the FLAC codec.

FLAC supports tags in the FlacTags format. The ability to encode multi-channel sound is a significant advantage, according to Monkey's Audio. The format supports any sampling frequencies in the range from 1 Hz (!) to 65.535 Hz. Audio capacity 4 (!) Up to 32 bits.

It is important that when paired with lossless codecs, FLAC uses system resources most efficiently when decoded (created) audio. Unfortunately, there is no significant advance in the hour of encoding (pressure).

The FLAC website is regularly updated, new versions of the codec are released. By the way, FLAC is insanely in the lead in development activity. It is entirely possible to break it down in the future in the main format.

AAC (English: Advanced Audio Coding) - an audio file format with less loss of capacity when encoded, less than MP3 for the same size. The format also allows compression without losing the output code (ALAC AAC profile).

AAC was initially designed as a successor to MP3 with a colored coding. AAC format, officially recognized as ISO/IEC 13818-7, published in 1997, as a new family, part of the MPEG-2 family. It also uses the AAC format as well as MPEG-4 Part 3.

This file type is standard for Apple Macintosh systems and audio processing systems based on it. Apple AIFF stands for Audio Interchange File Format - an audio interchange file format that is somewhat similar to WAV. Yogo, those who are permitted at once by the sound of the sound hwilee of the dodatkov іnformasy, zokrem of the Sample of WaveTable (sounded the sound of iz at the parameters of the synthesizer), and the yaks of the pidsumic result. I want to immediately create files on Apple computers in almost any format, including MP3.

Some companies create their own proprietary formats that support other players. Apple Lossless was created by the same company for the iPod as an alternative to FLAC, which is used on other players. If you don't use iPod, you may not encounter this format at all.

Monkey's Audio (APE)

Monkey's Audio or APE (English version) ape - Mavpa) is a popular format for digital audio coding at no cost. It can be used everywhere without any costs at once, with open source code and a set of software for coding and creation, as well as plugins for popular players. Monkey's Audio files have the following extensions: .ape to save audio and .apl to save metadata. Regardless of the open source code, Monkey's Audio is not valid because its license imposes copyright restrictions.

The average bitrate in an audio file is 600-700 kbps; Upgrade to 128 kbps MP3. The average pressure is 40-50% dependent on the genre of music: while classical and jazz works are compressed at the highest level, then compositions in the style of trash-metal or anything similar to “electronic noise” will show the best result. For cost-intensive codecs, with acceptable luminosity, the compression becomes close to 80%.

There are several steps of compression. Maximum compression can be achieved with just the right decisions, regardless of the great hour of compression. However, it is necessary to take into account the resource consumption of the system that creates the file - for a maximally compressed file, the cost is significantly high.

The APE format will provide support for tags to search for songs in a music collection. Another advantage is checking the integrity of the file before decryption. We strive to update the original wav file with the compressed .APE.

Monkey's Audio runs a graphical frontend for Windows, an otherwise seemingly manual window program for handling the coding process. Other codecs require support from the command line and frontends of third-party vendors. Most importantly, the Monkey's Audio frontend can be combined with other codecs - Rkau, Wavpack, Shorten and lossy codecs mp3 and ogg vorbis.

A little about the shortcomings. The Monkey's Audio codec is only available on Windows. However, the website says that “versions for Mac and Linux are already being released.” Before speaking, the site itself has not been updated for a long time, which is not a good sign. There is also daily support for manufacturers of hardware players.

MIDI (English Musical Instrument Digital Interface - digital interface of musical instruments) is a standard for hardware and software that allows you to create (and record) music by recording/recording special commands, as well as a file format that allows the same commands. The device or program it creates is called a MIDI synthesizer (sequencer) and is actually an automatic musical instrument.

In addition to other formats, it saves not digitized sound, but sets of commands (notes to be played, sent to instruments to be played, values ​​of changed sound parameters), which can be created differently in I will build the creation. The use of the MIDI format as a data presentation format allows the implementation of devices that automatically generate arrangements for given chords, as well as 3D sound visualization programs. In addition, such files tend to be orders of magnitude smaller in size, with less digitization and an equalized sound quality.

We will look at different audio file formats:

WAVE (.wav)- The most expanded sound format. Used in Windows OS to save sound files. It is based on the RIFF (Resource Interchange File Format) format, which allows you to save more data in a structured view. To record sound, a variety of compression methods are used, so fragments of sound files can be used in great detail. The simplest compression method is Pulse Code Modulation (PCM), but it does not provide sufficient compression.

AU (.au,.snd)- The audio file format that is used on Sun workstations (.au) and on the NeXT operating system (.snd). With the advent of a wide expansion of the Internet, at an early stage there was development of a standard format for audio information.

MPEG-3 (.mp3)- audio file format, one of the most popular today. There will be divisions to save sounds that are important in human language. Vikorist is used to digitize music recordings. Previous versions of the format: MP1 and MP2. When encoded, psychoacoustic compression is established, if the melody contains sounds that are poorly perceived by the human ear. Early versions provide the highest compression and are less capable of using computer resources at the time of creation. The characteristics of the processor greatly affect the sound quality - the weaker the processor, the more harmful the sound.

MIDI (.mid)- Digital interface of musical instruments (Musical Instrument Digital Interface). This standard is divided into the beginning of the 80s for electronic musical instruments and computers. MIDI means the exchange of data between music and sound synthesizers of various players. The MIDI interface is a protocol for transmitting musical notes and melodies. However, MIDI is not digital sound - it is a shorter form of recording music in numerical form. A MIDI file contains a sequence of commands that record actions, such as pressing a key on a piano or turning a knob. These commands, which are sent to the device for creating MIDI files, affect sounds; small MIDI commands can affect the sound or the sequence of sounds on a musical instrument or synthesizer, so MIDI files take up less space. єм (unit of sound sound per second), nіzh sound.

MOD (.mod)- a musical format, in which images of digitized sound are saved, which can be used as templates for individual notes. Files in this format begin with a set of sound symbols, followed by notes and information about the problem. A skin note is created using one of the sound patterns placed on the ear. Such a file is remarkably small and has a structure that is based on notes. This makes it easier to edit for additional programs that have a traditional musical recording. The entry into the MIDI file sets the sound, which allows it to be created on any computer platform.

IFF (.iff)- Interchange File Format – a format, in its entirety, for the Amiga computer platform. Narasi is also available on compact discs in the CD-I format. Its structure is very similar to the RIFF format.

AIFF (.aiff) - Audio Interchange File Format – a format for exchanging audio data, available on Silicon Graphics and Mac computer platforms. Much like the Wave format, the prote, in addition to the new one, allows you to digitize sounds and patterns. There are plenty of programs to open files in this format.

RealAudio (.ra, .ram)- Format, divisions for the creation of sound on the Internet in real time. Distributed by Real Networks (www.real.com). The quality that comes out, in short, corresponds to the middle audio cassette, for clear recording of musical works in the mp3 format is more important.

In the world of music there are a number of musical formats, their modifications and versions, created by the giants of the music industry and small companies that have been gaining more recognition in the electronic world.

For these purposes, a variety of physical methods for storing audio data have been developed, for example: vinyl records, magnetic tape, compact discs, DAT, MD (minidisc), DVD, or converting notes in music formats (MIDI), thus the application Ilos bezlich different computers 'Uther methods for saving audio data - digital: OGG, MP3, Flac, Wav formats.

It is impossible to look at and discuss all the audio formats, codecs, their advantages and disadvantages, so in my article I will try to inform you about the most popular extensions of audio files that you are familiar with.

Why can’t we choose any universal format for encoding audio files? Because for the implementation of various functions, a different format is required. For example: for creating a CD in a CD drive, for recording music or sound effects in video games, for recording a movie track or video clip, for playing in mobile phones or transferring files via the Internet, in addition, There are a number of operating systems that have achieved the greatest expansion in the world. These include: Amiga, Macintosh, NEXT and personal computers with the Windows operating system.

In addition, a robot dj, sound engineer, cj, video engineer or a simple music lover are subject to severe criticism for their work. Why would you need your audio data to be saved in your own way? For example, the sound for a CD can be saved at a 16-bit bit rate and a sampling frequency of 44.1 kHz. In the interest of sound via the Internet, it is better for us to use different bit depth and sampling frequency, so the skin of 16-bit, 44-kilohertz sound takes up approximately 10 MB, then. The middle track has a value of 5 hvilins and becomes 50 “meters” - this is a very great gift for the average koristuvach. This article provides short information about the most popular music formats.

A.A.(Audible Audio Book File) - The format will be closed, split by Audible. Suitable for recording audiobooks that are sold through Audible and iTunes services. It is possible to increase or speed up the speed of listening to files - digital pitch, the ability to remove bookmarks when listening to audio books, file protection, when delivering audio recordings over the Internet.

A.A.C.(Advanced Audio Coding) – audio file format with less loss of capacity when encoded, less than MP3 for the same size. Coding music without spending money on the original using an additional ALAC profile. AAC is a family of MPEG4 audio encoding algorithms. Instead of a hybrid set of mp3 filters, AAC uses the vicoristic MDST technology (modified cosine transformation) - which means that the hearing removes the harshness of the sound, compared to MP3 encoding with the same or less bitrate ytom. Possible extensions for AAC files: [.m4a], [ .m4b ], [ .m4p ].

AAC is also a broad-based audio encoding algorithm that combines two basic encoding principles to greatly reduce the amount of data required to transmit high-quality digital audio. This format is one of the most obvious ones, which reduces the cost pressure and allows for greater daily use, including portable ones.

As of 2009, there are significantly fewer extensions, including MP3 and other alternative solutions. AAC (Advanced Audio Coding) was initially created as a successor to MP3 with improved coding. AAC format, officially recognized as ISO/IEC 13818-7, published in 1997, as a new family, part of the MPEG-2 family. It also uses the AAC format as well as MPEG-4 Part 3.

Advantages of AAC over MP3:

- Up to 48 audio channels;

- greater coding efficiency both at steady and variable bitrates;

- Sampling frequencies from 8 Hz to 96 kHz (MP3: 8 Hz - 48 kHz);

- Greatest Joint stereo mode.

ADX– based on the ADIKM proprietary format, there is a reduction in costs and savings in sound recording, development of CRI Middleware specifically for use in video games. The most characteristic feature is the ability to loop a sound recording in order to create a manual format for use as background music in various games that support the entire media container. This is supported by SEGA Dreamcast games for PlayStation 2 and GameCube.

In addition to MP3, no one is stuck with the psychoacoustic model of changing data about sound (changing its complexity). The ADPCM model is designed to save data records by reducing the transfer function, which means greater security of the output signal after encoding; In essence, the compression of ADPCM instead of the replacement of new oversize images of the sound recording gives the images a boost to the signal from the previous value, which may have a much smaller size, which is 4 bits. For the human ear, such delicacy is equal to noise in order to prevent the loss of ice.

AIFF– This is a standard file format for saving audio data on the Macintosh platform. If you need to transfer audio files between a personal computer and a Macintosh computer, select the format itself. It supports 8- and 16-bit mono and stereo audio. Files of this format may or may not have the Mac-Binary header. If a file of this type does not have a Mac-Binary header, it means that it has the aif extension. If a file of this type contains the Mac-Binary header, Sound Forge will open it and identify it as a file in the Macintosh Resource format (div. next section). In this case, the file that is superior to everything has the extension snd. Note When files are saved on Macintosh computers, they are referred to as the Mac-Binary header. This is a small piece of information that is written to a file that identifies the file type for the Mac OS operating system and other programs. In this way, Macintosh computers tell you what to put in a file: text, graphics, or, for example, audio.

AMR(Adaptive multi rate) [ . amr] - adaptive coding with variable speed. A standard for encoding audio files specifically for compressing signals in the radio frequency range. Standardization by ETSI (European Telecommunications Standards Institute). The AMR network allows you to ensure high capacity of the network with simultaneously high transmission capacity. AMR has a wide range of language encoding/decoding capabilities and allows you to seamlessly switch between different modes depending on the minds or preferences, ensuring crystal-clear voice transmission.

A.P.E.– (Monkey's Audio) [ . ape] – by Matthew T. Ashland – digital sound format without wasting power ( lossless ). The Monkey's Audio codec is released only for the Microsoft Windows platform, although there are a number of unofficial codecs for MacOS, Linux, BeOS. Monkey's Audio files have the following extensions: .ape – for saving audio and .apl – for saving metadata. This format is no longer valid, because The license is seriously limited by its extension.

AppleLossless[. m4 a] – This is an audio codec, developed by Apple Inc, for compressing digital music without wasting data. Apple Lossless data is saved in an MP4 container with extensions. m4a. Although Apple Lossless has the same file extension as AAC, it is not AAC, a codec similar to other Lossless codecs, such as FLAC and others. It does not rely on digital rights management (DRM), but due to the nature of the container, it is important that DRM can be stacked before ALAC.

Tests have shown that ALAC files yield approximately 40% to 60% of the size of the originals due to the type of music, like other Lossless formats. In addition, the flexibility with which decoding can occur is useful for productivity-intensive devices such as the iPod.

Apple Lossless Encoder is presented as one of the components of QuickTime 6.5.1 January 28, 2004 and as a feature of iTunes 4.5. The codec is also tested in AirPort Express AirTunes.

A decoder for the Apple Lossless format is now available in all versions of the libavcodec library. This means that any multimedia programr based on this library, including VLC and MPlayer multimedia, can play Apple Lossless files.

CDDA(Compact Disc Digital Audio) - audio compact disc, an international standard for storing digitized sound on compact discs, introduced by Philips and Sony. Sound information is represented by pulse code modulation with a sampling frequency of 44.1 kHz and a bit rate of 1411.2 kbps, 16 bit stereo.

ZSpecification of audio in the Red Book standard:

- The maximum hour of all recordings is 79.8 hours;

- The minimum hour for a track is 4 seconds (including a 2-second pause);

– maximum number of tracks – 99;

- The maximum number of points per track (parts of a track) is 99 without limiting the hour;

- guilty but present International Standard Recording Code (ISRC).

DTS– (Digital Theater System), essentially the same as Dolby Digital , Or rather, his competitor. Format DTS Vikorista minimal level of squeeze, lower Dolby , so in fact it will sound better to put it into practice DVD discs on which tracks are recorded in DTS or DD formats.

DTS This digital theater system is a family of digital multi-channel sound recording systems created by the Digital Theater System company for the demonstration of digital phonograms in cinemas synchronously with rental film copies. Cream accompanying spitting film copies, offending systems ( DTS and Dolby Digital ) in simple terms, vikorist on optical video discs for home viewing. DTS Vikoristovu less rhubarb squeeze, lower Dolby But there is no absolute advantage in every system. Argue about advantages DTS or Dolby Digital don't stop until this day. Format DTS Stereo practically identical Dolby Surround. DTS Supports both 5.1-channel and 7.1-channel sound options. DTS home theaters allow higher bitrates (1509.75 kbps).

FLAC(Vilny codec from the Ogg project)[.flac] - (English: Free Lossless Audio Codec - free audio codec without spending) - a popular free codec for audio compression. In addition to codecs using Ogg Vorbis, MP3 and AAC, FLAC does not remove any necessary information from the audio stream and is suitable for listening to music on high-quality sound production equipment, so and for archiving an audio collection. Today, the FLAC format is supported by a wealth of audio add-ons. To preserve basic types of metadata, the basic decoder decoder wikirist tags ID 3 v 1 and ID 3 v 2, they can be freely added and edited.

MIDI(Musical Instrument Digital Interface) - Digital interface of musical instruments. This digital audio recording standard is a format for exchanging data between electronic musical instruments.

The interface, however, allows you to digitally encode data such as key pressure, adjusting the volume and other acoustic parameters, selecting timbre, tempo, tonality, etc., with precise reference to the hour. The coding system has no independent commands, which developers, programs and contributors can use at their own discretion. Therefore, the MIDI interface allows, in addition to recorded music, synchronization of equipment with other equipment, for example, lighting, powder technology, etc.

The sequence of MIDI commands can be recorded in any digital file or transmitted via any communication channels. The device or program it creates is called a MIDI synthesizer (sequencer) and is actually an automatic musical instrument.

MP2 (MPEG-1 Audio Layer II or Musicam) [ . mp2 ] – one of three formats (level 2) that compresses sound at the cost of the MPEG-1 standard. It is based on digital radio broadcasting DAB and the old Video CD standard, which was used in the 90s to distribute films on optical CDs and DVDs.

MPEG-1 Audio Layer 2 encoder developed from the MUSICAM audio codec (Masking pattern adapted Universal Subband Integrated Coding And Multiplexing - universal subband and multiplexing with adaptation to the masking pattern), developed by CCETT, Philips and IRT in 1989 as part of EUREKA 1 tracking digital systems Radio communication for stationary, portable and mobile devices. The main parameters of MPEG-1 Audio were reduced from MUSICAM, including the filter bank, time-area processing, audio frame size, etc. However, after further refinement, the MUSICAM algorithm did not comply with the final version of the MPEG-1 Layer II standard.

MP3 (MPEG Layer 3) [ . mp3 ] – The third audio encoding format is MPEG - a file format that is licensed to save audio information. At the moment, MP3 is the most popular and popular format for digital encoding of audio information with waste. It is widely used in file-sharing networks for the evaluation of transferring musical works. The format can be easily downloaded in any popular operating system, on any portable audio player, and is also supported by all current models of music centers and DVD players.

The MP3 format uses a cost-saving algorithm to reduce the amount of data needed to make a recording and ensure that it is created even close to the original (in the opinion of most listeners), I want to talk about music lovers' remarkable vibrancy. When an MP3 is created with an average bitrate of 128 kbps, the resulting file is approximately 1/10 the size of the original file from the audio CD. MP3 files can be created at high or low bitrate, which affects the quality of the resulting file. The principle of constraint refers to the reduced accuracy of certain parts of the sound stream, which is practically imperceptible to the hearing of most people. This method is called koduvannyam priinyattya. In this case, at the first stage, there will be a diagram of sound in the view of a sequence of short intervals of hours, then it will display information that is not perceptible to the human ear, and information that is lost will be saved Available in a compact form. This approach is similar to the compression method used when compressing images in JPEG format. There are plenty of musical gourmands who will be able to compress music with a maximum brightness of 320 kbps , or switch to other formats, for example FLAC average bitrate ~1000 kbps.

MusePack[. mpc] – non-license file format for saving audio information that is being expanded GNU General Public License.

Musepack is stagnant in its distribution of different frequencies, which includes so-called subband codecs. The main feature is the precise adjustment of psychoacoustics, which allows you to work with pure VBR encoding (encoding with a variable bitrate). The main mission of Musepack is to provide insight into the sound of coded music.

In current formats, such as: MP3, Vorbis, AAC, AC3, WMA, another dct transformation is generated, which allows them to achieve better results at medium and low bitrates, but does not allow them to achieve high results at higher rates . MusePack does not work with another dct conversion, which allows you to reach unconverted capacity at bitrates higher than 180.

So, just like in AAC and many other current formats, Musepack pairs channels across different frequencies, which slightly differs on the brilliance, but allows for significant savings on size. In MP3, the pairing of channels is carried out not by frequency ranges, but for all ranges of frequencies, dividing the signal into the subfrequency, then sorting the signal into a series of cosines (MDCT - the next step of Fourie’s transformation) and recording rounding (quantized) ) the values ​​of the withdrawals after the conversion of the coefficients (quantization) is required prior to the psychoacoustic analysis). MPC, after dividing the signal into frequencies, simply requantizes (using psychoacoustics) the amplitude signal in the skin fluid and removes the rounded value (quantization) and writes it to the output flow. This fact itself explains the great flexibility of compression and decompression of the MPC.

MOD- Release format for the Amiga platform. The MOD file contains digitized recordings of real instrument sounds, called samples, which have a similar MIDI structure. Or a composer writing in the MOD format creates a program called a tracker, which indicates which instrument itself, at what hour, with which note and octave is to sound - this sequence of notes is recorded in the list - a track, and a number of tracks, etc. o in parallel sound, establish a block, titles as a pattern. The set of parameters creates a module - a file in the MOD format, with extensions. One tracker line represents one real channel, in which numbered notes can be played or edited. The notes may have different “ornaments” - for example: tremolo, glissando, etc.

OGG [.ogv], [.oga], [.ogx], [.ogg] is the standard for the multimedia container format, which is the main file and streaming format for multimedia codecs of the Xiph.Org Foundation, as well as the name of the project that is developing this format and codecs for the new one. Like all technologies that are developed under Xiph.Org, the Ogg format is protected by a unique standard that does not involve patent or licensing restrictions.

Ogg is just a container. Music and video are compressed by codecs, and the processing result is stored in such containers. Ogg containers can store streams encoded with multiple codecs. For example, a file containing video and audio may contain data encoded with audio and video codecs.

The Ogg container can save audio and video from various formats (such as MPEG-4, Dirac, MP3 and others).

RealAudio[. ra],[. ram] – Promotional standard for streaming and format of media files that belong to the company " RealNetworks Products and Services." RealAudio ahead of ideas in the warehouse package RealAudio 10, codec to compress sound without sacrificing brightness.

Among the advantages of this codec is support for streaming, even decoding. The downsides include the closedness of the code and the lack of rich channels. Available for Microsoft Windows, Macintosh and GNU/Linux.

RKAU[.rka] – Among all audio codecs, RKAU occupies a special place. First of all, it’s the smallest (only 25kB!) and the best encoder. In another way, in addition to the fact that it is a lossless sound compression program, in a new transfer of the lossless compression mode, which will provide more, below lossless algorithms, the compression stage. However, due to the peculiarities of the algorithm that underlies rkau creation, the codec that is introduced lies not in the spectral region (as in the type of psychoacoustic models of encoders MP3, MP+, AAC and others), but in the real region. That is, apparently, the non-linear nature, as well as the creation of most tracts. In this case, there is no waste of other details and microplanes in phonograms. However, if you “overdo it” in this interpretation, the sound can become completely inaudible: the sound will show severe noise-like artifacts, and the sound itself will be clearly distorted.

In the hierarchy of audio codecs, the rkau program stands completely apart. The system is original and has no analogues to other audio compression algorithms. The small size of the encoder program (25kB) and the high speed of the robot at similar to other lossless algorithm levels make it difficult to bring rkau into the crazy leaders. And if you want the most effective lossless encoder, you can use OptimFROG, which is shown in the front part of the article, rkau stands out for its efficiency. However, when activating the compression mode “with waste”, rkau is set in the mode of greatest brilliance far behind all lossless algorithms, approaching the effectiveness of programs based on a psychoacoustic model (MP3, MP+, AAC, VQF and other). In this case, the loss of microplanes and nuances of the output audio material, which is typical for MPEG-like algorithms, is not observed, and artifacts that inevitably appear in this case can be noted only on even clear equipment with a large number of daily listening sessions.

Shorten[.shn] – this is a format that is used to compress audio data. This form of file compression is used to compress CD capacity, tp gjnthm audio files (44.1 kHz, 16 bit, stereo PCM ). This format is still being abused by certain people, which is why it is illegal to sell concert recordings that are coded as Shorten files.

Speex [. spx] - a useful codec for compressing a wireless signal, which can be used in voice-over-Internet add-ons ( VoIP ). Due to the high reliability of the wines, there are no regular patent exchanges or licensing under the remaining version of the license BSD (without third article). Squeezed by codec Speex The data can be saved in either the audio data saving format Ogg , or transfer directly for additional packets UDP/RTP.

Retailers should oppose their development to other open codecs, such as the codec Vorbis , firmly, what is the codec itself Speex best suited for transmitting voice over a network with unreliable delivery of data packets. In this case, the authors of the development specifically emphasize that the codec is suitable for use in the context of unreliable packet transmission, either the packet is received or not.

Speex to be elevated to the class of such ranks Code Excited Linear Prediction (CELP) )-codecs, or codecs generated on the basis of the so-called Linear Transmission Coding of the LPC. The LPK uses a digital filter for approximation of the cutoff of the signal signal only with turn-off links (the so-called “autoregressive filter”). The coefficients of this filter are “adjusted” in response to the signal of the additional Levinson procedure (in recent literature - Levinson-Durbin). CELP -modification of the LPC conveys the appearance of the so-called sound. “code book” to place song sets of awakening LPK filter single impulses.

Local signal at the codetsi Speex is divided into sections that do not overlap, with a duration of 20 ms (160 sections at 8 kHz). In this case, to assess the restlessness of a set of statements, the segment is divided into several subdivisions with a duration of 5 ms per day. On the skin there are oscillating sets of pulses from both the flow feeder (from the code book) and the two forward drives. For the substitution of other codecs in order to avoid patent infringement, Speex Not the vikoryst coding of algebra, but only vector. The damage to the two front bearings is due to changeable conditions, in addition to other codecs, where changing positions change over time.

For the application of the robbers, Speex optimized for picking up a high-acidity signal at low speeds. Codec Speex It also allows you to vary the level of signal compression and supports signals with different signal widths: ultra-wide (sampling frequency 32 kHz), wide-black (16 kHz) and high-black (8 kHz).

SO(Tom's lossless Audio Kompressor) [ . So] – Audio codec is a format that compresses digital sound without wasting money. There is a high level of compression and fluidity of encoding and decoding. It is easy to use everywhere with a set of software for coding and creation, as well as plugins for popular players: Winamp, foobar2000, etc. The first final version 1.0 was published on June 26, 2007.

The format continues to be actively developed (the current version is 1.1.1) and currently, based on research on the hydrogenaudio.org forum, includes up to three of the most popular audio recording formats without waste (after FLAC and WavPack)

TTA(True Audio) - Cost-free audio codec that compresses music files without wasting time in real time. The codec is based on adaptive transfer filters and has all the reduced characteristics as most current coders. The compressed file size will be 30% - 70% smaller than the original music file. TTA format supports ID3v1 and ID3v2 tags. Vikorist's True Audio codec can accommodate up to 20 audio CDs on one DVD-R disc.

TwinVQ(Transform – domain Weighted Interleave Vector Quanization) - Vector quantization with transform domains and important digitization), developed in Japan in the laboratory NTT Human Interface Laboratories.

VQF files are approximately 30-35% smaller than MP3, but with the same clarity of sound. A stream of 128 Kbps of MP3 files is matched by a stream of 80 Kbps of VQF files. These are the challenges and the turning point. When decoding the processor is also 30% faster, MP3 decoding is lower. This means an increase in access to the computer where it is planned to program such files.

Tests show the superiority of VQF for all parameters at lower frequencies and less interference with signal shapes with a large dynamic range (real music). However, due to the collapse of the upper frequencies of the VQF sound spectrum by 2-3 dB, MP3 is sacrificed at frequencies higher than 15 kHz. This can be easily compensated for by adjusting the player’s equalizer, so it’s objective to put VQF on the side of the sound, on par with MP3.

VQF(Interleave Vector Quantization)– developed in Japan and based on TwinVQ technology. If we compare VQF and MP3, then the first format will be 30-50% more “compact”, but with the same clarity of sound. This gives VQF an advantage over the MP3 format. Also, the process of encoding, decoding (decoder) VQF takes approximately 30% more resources of the PC processor, not MP3 audio.

Tests show TwinVQ's superiority in all parameters at lower frequencies and less interference with signal shapes with a large dynamic range (real music). However, due to the collapse of the upper frequencies of the TwinVQ sound spectrum by 2-3 dB, MP3 at frequencies higher than 15 kHz is sacrificed. This can easily be compensated for by adjusting the player’s equalizer, so it’s objective to put TwinVQ on the side of more powerful sound on par with MP3.

Vorbis [. ogg] – a free format of compression of sound due to expenses, which officially appeared in 2002 rock. The functionality is similar to such codecs as AAC, AC3 and VQF, which is superior to MP3. The psychoacoustic model, which is being developed in Vorbis, is similar in principle to MP3 and similar, mathematical processing and practical implementation of this model are completely different, which allowed the authors to announce their format absolutely not stale from all the predecessors.

Ogg Vorbis uses a variable bitrate, in which the remaining values ​​​​are not limited by any hard values, and can be changed to 1 kbps. Please note that the format is not strictly limited by the maximum bitrate, and with maximum adjustments, the encoding can be varied from 500 to 1000 kbps. The sampling frequency is equally flexible - users can choose between 2 kHz and 192 kHz.

Vorbis is breaking up the "Xiphophorus" audio format in order to replace all paid proprietary audio formats. Regardless of the fact that it is the youngest format among all MP3 competitors, Ogg Vorbis is gaining continued support on all popular platforms (Microsoft Windows, Linux, Apple Mac OS, PocketPC, Palm, Symbian, DOS, FreeBSD, BeOS, etc.), as well as There are a large number of hardware implementations. However, despite all its advantages over competitors, the popularity of this format is still low.

WAV(Waveform audio format) [ . wav], [. wave] – split completely IBM . Recording format (stereo-mono) sound without compression. So, just one piece of stereo sound recording is divided into CD-yield (sampling frequency 44.1 KHz) with 60 s x 44100 Hz x 2 channels = 5292000 pixels. On the other hand, there can be 8 or 16 bits. Thus, the option has 8 bits per video, one sound occupancy in memory is 42336000 bits = 5292000 bytes (about 5 MB).

WavPack[.wv], [.wvс] – Cost-free audio codec with a closed code for compressing audio without wasting capacity. Destruction by David Bryant.

WavPack format allows you to compress (and edit) 8-, 16-, 24- and 32-bit audio files in .WAV format. It also boosts audio streams at high sampling rates. As with other methods of compression without loss of brightness, the efficiency of compression lies in the output data, but may lie in the range between 30% and 70% for most popular music, a little more for classical music similar to other devices with a wider dynamic range.

WavPack also includes a unique “hybrid” mode, which provides all the advantages of compression without wasting money with an additional bonus: replacing the creation of one file, in which mode a relatively small file of high value (more precisely, specified during encoding) is created. sti zi vtratoyu (.WV), which can can be programmed on its own, as well as a “correction” file (.WVC), which (in combination with the front.WV) allows you to completely update the original. For some consumers, this means that they will never have the opportunity to choose between pressure without wasting or wasting power.

WMA(Windows Media Audio) [ . wma] – The file format is licensed and developed by Microsoft to save and broadcast audio information. Initially, the WMA format was positioned as an alternative to MP3, but today Microsoft is introducing the AAC format (visored in the popular online music store iTunes).

Nominally, the WMA format is characterized by excellent compression, which allows it to “compress” the MP3 format and compete in terms of parameters with Ogg Vorbis and AAC formats. However, as was shown by independent tests, and also with a subjective assessment, the range of formats is still not clearly equivalent, and the advantage over MP3 is clear, as confirmed by Microsoft. It is especially important to note that early versions of the format (or its implementation) have little problems at low flow rates. Also, many music lovers and users of digital players do not like the WMA format due to its low resistance to damage. If, when encoding/transferring a WMA file, some part of it is corrupted, then opening the file becomes impossible, both after the scene of corruption and several tens of seconds before. (To be clear: if the MP3 file is corrupted, you can still open it until the point of corruption, then skip a few seconds and continue until the end; sound little noticeable to the ear or not noticeable to the eye.) However This format is gradually evolving, so it can be adjusted and optimized.

Most portable audio players support the WMA format rather than MP3. This format is very poorly supported on alternative platforms (through its closure).

Microsoft has approved WMA's digital rights management (DRM) support. The main disadvantage is the inability to detect the theft of a composition on other computers, in addition to the one on which the composition was purchased from a music store.

In the remaining versions of the format, starting with Windows Media Audio 9.1, the encoding was transferred without losing the English quality. lossless, multi-channel encoding of surround sound and encoding of voice.