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What is the best audio format. Audio file formats

In the world of music, there are a huge number of music formats, their modifications and versions, created by giants of the music industry and small companies that have gained public recognition in the electronic world.

For these purposes, various physical methods for storing audio data have been developed, for example: vinyl records, magnetic tape, CDs, DAT, MD (minidisc), DVD or converting notes into music formats (MIDI), in the same way many different computer methods have emerged audio data storage – digital: OGG, Mp3, Flac, Wav formats.

It is impossible to review and discuss all sound formats, codecs, their advantages and disadvantages, so in my article I will try to talk about the most popular audio file extensions that you come across.

Why can't we use any universal audio file encoding format? Because to implement various functions you need your own format. For example: for playing 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 over the Internet, in addition, there are a number of operating systems that are most widespread in the world . These include: Amiga, Macintosh, NEXT and personal computers with the Windows operating system.

In addition, the work of a dj, sound engineer, cj, video engineer or a simple music lover is quite different in nature. This may require that your audio data be saved in your own way. For example, audio for a CD must be stored using 16 bits and a sampling rate of 44.1 kHz. However, to download audio over the Internet, we are better off using a different bit depth and sampling rate, since each minute of 16-bit, 44-kilohertz audio takes up approximately 10 MB, i.e. an average track lasting 5 minutes will be 50 “meters” - this is too much data for the average user. This article provides brief information about the most popular music formats.

A.A.(Audible Audio Book File) – the format is closed, developed by Audible. It is used to record audiobooks that are sold through Audible and iTunes. It is possible to slow down or speed up the speed of listening to files - digital pitch, the ability to leave bookmarks when listening to audio books, file protection when delivering sound recordings via the Internet.

A.A.C.(Advanced Audio Coding) – an audio file format with less quality loss during encoding than Mp3 with the same dimensions. Encoding music without loss of original quality using the ALAC profile. AAC is a family of MPEG4 audio coding algorithms. Unlike the hybrid mp3 filter bank, AAC uses MDST technology (modified cosine transform) - this means that the listener receives better sound quality than MP3 encoding with the same or lower bitrate. Possible AAC file extensions: [.m4a], [ .m4b ], [ .m4p ] .

AAC is also a wideband audio coding algorithm that uses two basic coding principles to greatly reduce the amount of data required to transmit high-quality digital audio. This format is one of the highest quality, using lossy compression, supported by most modern equipment, including portable ones.

As of 2009, it is much less widespread than MP3 and other alternative solutions. AAC (Advanced Audio Coding) was originally created as a successor to MP3 with improved encoding quality. The AAC format, officially known as ISO/IEC 13818-7, was released in 1997 as the seventh new member of the MPEG-2 family. There is also an AAC format known as MPEG-4 Part 3.

Advantages of AAC over MP3:

– up to 48 audio channels;

– greater coding efficiency at both constant and variable bitrates;

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

– more flexible Joint stereo mode.

ADXis an ADPCM-based proprietary lossy audio compression and storage format developed by CRI Middleware specifically for use in video games. The most characteristic feature is the ability to loop a sound recording, which makes the format convenient for use as background music in various games that support this media container. It is supported by many SEGA Dreamcast games and some PlayStation 2 and GameCube games.

Unlike MP3, it does not use the psychoacoustic model of reducing the volume of sound data (reducing its complexity). Instead, the ADPCM model uses a relative prediction error data record to store samples, which means greater preservation of the original signal after encoding; Essentially, ADPCM compression, rather than using full resize samples of the audio recording, provides samples of the signal's deviation from the previous value that are much smaller in size, typically 4 bits. To the human ear, this deviation is at the noise level, making the loss of quality barely noticeable.

AIFFis a standard file format for saving audio data on the Macintosh platform. If you ever need to transfer audio files between a personal computer and a Macintosh computer, use this format. It supports 8- and 16-bit mono and stereo audio data. Files in this format may or may not contain a Mac-Binary header. If a file of this type does not contain a Mac-Binary header, it most likely has an aif extension. If a file of this type contains a Mac-Binary header, Sound Forge will open it but identify it as a Macintosh Resource file (see next section). In this case, the file most likely has the extension snd. Note When files are saved on Macintosh computers, a so-called Mac-Binary header is added to them. This is a small piece of information written at the beginning of a file that identifies the file type for the Mac OS operating system and other applications. This is a way for Macintosh computers to tell you what a file contains: text, graphics, or audio data, for example.

AMR(Adaptive multi rate) [ . amr] - variable rate adaptive encoding. An audio file encoding standard specifically designed for signal compression in the speech frequency range. Standardized by ETSI (European Telecommunications Standards Institute). The use of AMR makes it possible to provide high network capacity with simultaneously high quality voice transmission. AMR has a wide range of speech encoding/decoding speeds and allows you to flexibly switch to different modes depending on environmental conditions or network load, ensuring crystal clear voice transmission in any conditions.

A.P.E.– (Monkey's Audio) [ . ape] – developer Matthew T. Ashland – lossless digital audio format ( 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, and BeOS. Monkey's Audio files use the following extensions: .ape for storing audio and .apl for storing metadata. This format is not free, because its license seriously restricts distribution.

AppleLossless[. m4 a] – is an audio codec developed by Apple Inc to compress digital music without data loss. Apple Lossless data is stored in an MP4 container with the .m4a extension. Although Apple Lossless has the same file extension as AAC, it is not AAC, the codec is similar to other Lossless codecs such as FLAC, etc. An iPod with a dock connector (not shuffle) and the latest firmware can play files in the Apple Lossless format. It does not use any digital rights management (DRM), but given the nature of the container, it is believed that DRM may apply to ALAC.

Tests have shown that ALAC-compressed files are approximately 40% to 60% the size of the originals, depending on the type of music, similar to other Lossless formats. Additionally, the speed at which it can be decoded makes it useful for performance-constrained devices such as the iPod.

Apple Lossless Encoder was introduced as a component of QuickTime 6.5.1 on April 28, 2004, and as a feature of iTunes 4.5. The codec is also used in AirPort Express in the AirTunes implementation.

A decoder for the Apple Lossless format is now available in the open source libavcodec library. This means that any media player based on this library, including VLC and MPlayer media, can be able to play Apple Lossless files.

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

WITHRed Book audio specification:

– the maximum time of all recordings is 79.8 minutes;

– minimum track time - 4 seconds (including a 2-second pause);

– maximum number of tracks - 99;

– maximum number of reference points (track sections) - 99 without time restrictions;

– must be present International Standard Recording Code (ISRC).

DTS– (Digital Theater System), essentially it’s Dolby Digital , or rather its competitor. Format DTS uses minimal compression level than Dolby , so in fact it sounds better, which is proven in practice DVD discs on which tracks are recorded DTS or DD format.

DTS This is a digital theater system - a family of digital multichannel sound recording systems created by the Digital Theater System company to demonstrate digital soundtracks in cinemas synchronously with rental film copies. In addition to accompanying film film copies, both systems ( DTS and Dolby Digital ) in a simplified form are used on optical video discs for home viewing. DTS uses less compression than Dolby , but none of the systems has absolute superiority. Benefits debate DTS or Dolby Digital have not stopped to this day. Format DTS Stereo almost identical Dolby Surround. DTS Supports both 5.1 channel and 7.1 channel audio options. DTS in home theaters allows full bitrate (1509.75 kbps).

FLAC(free codec from the Ogg project)[.flac] – (English Free Lossless Audio Codec - free lossless audio codec) - a popular free codec for audio compression. Unlike lossy codecs Ogg Vorbis, MP3 and AAC, FLAC does not remove any information from the audio stream and is suitable for both listening to music on high-quality sound reproduction equipment and archiving an audio collection. Today, the FLAC format is supported by many audio applications. To store basic types of metadata, the basic decoder uses tags ID 3 v 1 and ID 3 v 2, so they can be freely added and edited.

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

The interface allows you to uniformly encode in digital form such data as keystrokes, adjusting volume and other acoustic parameters, choosing timbre, tempo, tonality, etc., with precise timing. The encoding system contains many free commands that manufacturers, programmers and users can use at their discretion. Therefore, the MIDI interface allows, in addition to playing music, to synchronize the control of other equipment, for example, lighting, pyrotechnics, etc.

A sequence of MIDI commands can be recorded on any digital medium in the form of a file and transmitted via any communication channels. The playback device or program 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) of lossy audio compression defined in the MPEG-1 standard. Used in DAB digital broadcasting and the legacy Video CD standard, which was used to distribute films on optical compact discs in the 1990s and existed before DVDs became widespread.

The MPEG-1 Audio Layer 2 encoder evolved from the MUSICAM (Masking pattern adapted Universal Subband Integrated Coding And Multiplexing) audio codec developed by CCETT, Philips and IRT in 1989 as part of the EUREKA studies of 147 European intergovernmental developments for digital radio broadcasting systems for stationary, portable and mobile receiving devices. The main parameters of MPEG-1 Audio were inherited from MUSICAM, including filter bank, time domain processing, audio frame size, etc. However, after further improvements, the MUSICAM algorithm was not used in the final version of the MPEG-1 Layer II standard.

MP3 (MPEG Layer 3) [ . mp3 ] – the third audio track encoding format, MPEG, is a licensed file format for storing audio information. At the moment, MP3 is the most famous and popular of the common lossy digital encoding formats for audio information. It is widely used in file-sharing networks for the evaluation of music. The format can be played in almost any popular operating system, on almost any portable audio player, and is also supported by all modern models of stereo systems and DVD players.

The MP3 format uses a lossy compression algorithm designed to significantly reduce the size of data required to play a recording and provide a playback quality very close to the original (according to most listeners), although audiophiles report a noticeable difference. When creating an MP3 at an average bitrate of 128 kbps, the resulting file is approximately 1/10 the size of the original audio CD file. MP3 files can be created with high or low bitrate, which affects the quality of the resulting file. The principle of compression is to reduce the precision of certain parts of the audio stream, making it virtually inaudible to most people's ears. This method is called perceptual coding. In this case, at the first stage, a sound diagram is constructed in the form of a sequence of short periods of time, then information that is not discernible to the human ear is removed from it, and the remaining information is stored in a compact form. This approach is similar to the compression method used when compressing images into JPEG format. Many music gourmets prefer to compress music with maximum quality - 320 kbps , or switch to other formats, for example FLAC , where the average bitrate is ~1000 kbps.

MusePack[. mpc] – unlicensed file format for storing audio information, distributed over GNU General Public License.

Musepack uses frequency banding, so it belongs to the so-called subband codecs. The main feature is the precise tuning of psychoacoustics, which allows you to work with pure VBR encoding (variable bit rate encoding). The main goal of Musepack is the transparency of the sound of encoded music.

In modern formats, such as MP3, Vorbis, AAC, AC3, WMA, a second dct conversion is performed, which allows them to achieve better quality at medium and low bitrates, but does not allow them to achieve good results at higher ones. MusePack does not perform a second DCT conversion, which allows you to achieve unsurpassed quality at bitrates above 180.

Just like in AAC and some other modern formats, Musepack pairs channels by frequency bands, which has a slight impact on quality, but allows you to save a lot on size. In MP3, channels are paired not by frequency bands, but for the entire band, dividing the signal into frequency subbands, then decomposing the signal into a series of cosines (MDCT - a special case of the Fourier transform) and recording the rounded (quantized) values ​​of the coefficients obtained after the conversion (quantization occurs in accordance with the psychoacoustic analysis performed). MPC, after dividing the signal into frequency subbands, simply requantizes (based on psychoacoustics) the amplitude signal in each subband and writes the resulting rounded (quantized) values ​​to the output stream. The same fact explains the high speed of compression and decompression of MPC.

MOD– format developed for the Amiga platform. Each MOD file contains digitized recordings of the real sounds of instruments, so-called samples, somewhat similar to the MIDI structure. A Cj or a composer writing in MOD format uses a program called a tracker, in which he indicates which instrument should sound at what time, in what note and octave - this sequence of notes is recorded in a list - a track, and several parallel sounding tracks form a block , called a pattern. A set of patterns forms a module - a file in MOD format with the .mod extension. One tracker line corresponds to one real channel in which the cj can play or edit numbered notes. Notes can be assigned various “ornaments” - for example: tremolo, glissando, etc.

OGG [.ogv], [.oga], [.ogx], [.ogg] – an open standard 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 developing this format and codecs for it. Like all technologies developed under the auspices of Xiph.Org, the Ogg format is an open and free standard with no patent or licensing restrictions.

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

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

RealAudio[. ra],[. ram] – Prop standard for streaming and media file format owned by " RealNetworks Products and Services." RealAudio first introduced as part of the package RealAudio 10, codec for audio compression without loss of quality.

Among the advantages of this codec are support for streaming and very fast decoding. The disadvantages include closed code and lack of multi-channel functionality. Available for Microsoft Windows, Macintosh and GNU/Linux.

RKAU[.rka] – Among all audio codecs, RKAU occupies a very special place. Firstly, it is the smallest (only 25kB!) and fastest encoder. Secondly, in addition to the fact that it is a lossless audio compression program, it provides lossy compression modes that provide a greater degree of compression than all known lossless algorithms. However, due to the peculiarities of the algorithm underlying rkau, the distortions introduced by the codec are not in the spectral region (as in the case of psychoacoustic models of MP3, MP+, AAC and others encoders), but in the real region. That is, they have, roughly speaking, a nonlinear nature, like the distortions of most paths. In this case, there is no loss of small details and microplanes of the phonogram. However, if you “overdo it” in this regard, the sound can become completely indigestible: hard noise-like artifacts will appear in the sound, and the sound itself will acquire a pronounced coloration.

In the hierarchy of audio codecs, the rkau program stands completely apart. It is so original that it has no analogues among other audio data compression algorithms. The small size of the encoder program (25kB) and high speed of operation with compression rates similar to other lossless algorithms make rkau an undisputed leader. And although OptimFROG, discussed in the previous part of the article, can be considered the most effective lossless encoder, rkau is only slightly behind it in terms of efficiency. However, when the “lossy” compression mode is activated, rkau, even in the highest quality mode, leaves all lossless algorithms far behind, approaching in efficiency programs based on the psychoacoustic model (MP3, MP+, AAC, VQF and others). In this case, the loss of microplanes and nuances of the original audio material, characteristic of MPEG-like algorithms, does not occur, and the artifacts that inevitably arise can only be noticed on very high-quality equipment with repeated comparative listening.

Shorten[.shn] – is a format used to compress audio data. This form of file compression is used for CD-quality compression, tp gjnthm audio files (44.1 kHz, 16 bit, stereo PCM ). This format is still used by some people because it is legal to sell concert recordings in which are encoded as Shorten files.

Speex [. spx] is a free speech compression codec that can be used in voice-over-Internet applications ( VoIP ). It is highly likely that it has no patent restrictions and is licensed under the latest version of the license BSD (without the third article). Codec compressed Speex data can be stored either in audio data storage format Ogg , or transmit directly using packets UDP/RTP.

Developers contrast their development with other open codecs, for example, the codec Vorbis , claiming that it is the codec Speex best suited for voice over a network where data packet delivery is unreliable. At the same time, the authors of the development specifically emphasize that the codec is suitable for use in networks with unreliable packet transmission, that is, either the packet arrived or it did not.

Speex belongs to the class of so-called Code Excited Linear Prediction (CELP) )-codecs, that is, codecs built on the basis of the so-called Linear Predictive Coding LPK. LPK uses a digital filter with only feedback connections (the so-called “autoregressive filter”) to approximate a segment of a speech signal. The coefficients of this filter are “adjusted” to the signal segment using the Levinson procedure (in Western literature - Levinson-Durbin). CELP - modification of the LPK provides for the presence of the so-called. “code book”, which contains predefined sets of single pulses exciting the LPC filter.

Speech signal in codec Speex is divided into non-overlapping segments of 20 ms duration (160 samples at 8 KHz). In this case, to evaluate the excitatory set, the above segment is divided into four subsegments of 5 ms duration, respectively. On each of the subsegments, exciting sets of impulses are searched for both the current subsegment (from the code book) and the two previous subsegments. Unlike other codecs, in order to avoid patent restrictions, Speex does not use algebraic coding, but only vector coding. The excitations of the two previous subsections are added with variable weights, in contrast to a number of other codecs, which use variable time positions.

According to the developers, Speex optimized for high quality speech at low speeds. Codec Speex also allows for variable signal compression and supports signals with different bandwidths: ultra-wideband (32 kHz sampling rate), wideband (16 kHz) and narrowband (8 kHz).

SO(Tom's lossless Audio Kompressor) [ . so] – Audio codec and lossless digital audio compression format. It has a high compression ratio and encoding and decoding speed. Distributed free of charge along with a set of software for encoding and playback, as well as plug-ins for popular players: Winamp, foobar2000, etc. Developed by Thomas Becker, Germany. Relatively new codec. The first final version 1.0 was published on January 26, 2007.

The format continues to be actively developed (latest version 1.1.1) and is currently, according to a survey on the hydrogenaudio.org forum, one of the three most popular lossless audio compression formats (after FLAC and WavPack)

TTA(True Audio) – a free audio codec that compresses music files without loss in real time. The codec is based on adaptive predictive filters and has all the improved characteristics like most modern encoders. The compressed file size will be 30% - 70% smaller than the original music file. TTA format supports ID3v1 and ID3v2 tags. Using the True Audio codec, you can place up to 20 audio CDs on one DVD-R disc.

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

VQF files are approximately 30-35% smaller than MP3s with the same sound quality. A 128 Kbps stream for MP3 files corresponds to a 80 Kbps stream for VQF files. These advantages also have a downside. Decoding also uses 30% more CPU than MP3 decoding. This determines increased requirements for the computer on which you plan to play such files.

Tests show VQF's superiority in all respects at lower frequencies and much less waveform distortion with a large dynamic range (real music). However, in terms of the roll-off of the upper frequencies of the audio spectrum, VQF is 2-3 dB inferior to MP3 at frequencies above 15 kHz. This, however, is easily compensated for by adjusting the player’s equalizer, which objectively puts VQF a step higher in sound quality compared to 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, with the same sound quality. This gives VQF a significant advantage over the MP3 format. But the process of encoding, decoding (decoder) VQF, takes about 30% more PC processor resources than Mp3 audio.

Tests show TwinVQ's superiority in all respects at lower frequencies and much less waveform distortion with a large dynamic range (real music). However, in terms of the roll-off of the upper frequencies of the sound spectrum, TwinVQ is 2-3 dB inferior to MP3 at frequencies above 15 kHz. This, however, is easily compensated by adjusting the player’s equalizer, which objectively puts TwinVQ a step higher in sound quality compared to MP3.

Vorbis [. ogg] is a free lossy audio compression format that officially appeared in the summer of 2002. In functionality and quality it is similar to such codecs as AAC, AC3 and VQF, which are superior to MP3. The psychoacoustic model used in Vorbis is similar in operating principles to MP3 and the like, but the mathematical processing and practical implementation of this model are significantly different, which allowed the authors to declare their format completely independent from all predecessors.

Ogg Vorbis uses a variable bitrate by default, but the latter is not limited to any fixed values, and it can vary by even 1 kbps. It is worth noting that the maximum bitrate is not strictly limited by the format, and with maximum encoding settings it can vary from 500 to 1000 kbps. The sampling rate has the same flexibility, giving users any choice from 2 to 192 kHz.

Vorbis was developed by the Xiphophorus community to replace all paid proprietary audio formats. Despite the fact that it is the youngest format of all MP3 competitors, Ogg Vorbis has full support on all popular platforms (Microsoft Windows, Linux, Apple Mac OS, PocketPC, Palm, Symbian, DOS, FreeBSD, BeOS, etc.), and There are also 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] – developed jointly with IBM . Uncompressed audio recording format (stereo or mono). So just one minute of stereo sound recording made with CD quality (sampling frequency 44.1 KHz) contains 60 s x 44100 Hz x 2 channels = 5,292,000 samples. Each sample can have 8 or 16 bits. Thus, in the 8 bits per sample version, one minute of sound will take 42,336,000 bits = 5,292,000 bytes (about 5 MB) in memory.

WavPack[.wv], [.wvс] – Free, open-source audio codec for audio compression without loss of quality. Designed by David Briant.

WavPack format allows you to compress (and decompress) 8-, 16-, 24- and 32-bit audio files in .WAV format. It also supports surround sound streaming and high sampling rates. Like other lossless compression methods, compression efficiency depends on the source data, but it typically ranges between 30% and 70% for general popular music, slightly higher for classical music and other sources with a wider dynamic range.

WavPack also includes a unique "hybrid" mode that provides all the benefits of lossless compression with the added bonus: instead of creating a single file, this mode creates a relatively small high-quality (more precisely, specified at encoding) lossy quality (.WV) file that can play on its own, as well as a “correction” file (.WVC), which (in combination with the previous .WV) allows you to completely restore the original. For some users, this means they will never have to choose between lossless and lossy compression.

WMA(Windows Media Audio) [ . wma] – a licensed file format developed by Microsoft for storing and broadcasting audio information. Initially, the WMA format was positioned as an alternative to MP3, but today Microsoft opposes it to the AAC format (used in the popular online music store iTunes).

Nominally, the WMA format has good compression capabilities, which allows it to “bypass” the MP3 format and compete in terms of parameters with the Ogg Vorbis and AAC formats. But as has been shown by independent tests, as well as by subjective assessment, the quality of the formats is still not clearly equivalent, and the advantage even over MP3 is clear, as claimed by Microsoft. It is especially worth noting that early versions of the format (or its implementation) had problems at low bit rates. Also, many music lovers and owners of digital players do not like the WMA format for its low error resistance. If during encoding/transferring a WMA file some part of it is damaged, then playback of the file becomes impossible, both after the point of damage and several tens of seconds before it. (For comparison, if an MP3 file is damaged, you can still play it from the beginning to the very point of damage, then skip a few seconds and play it further to the end; sometimes errors of a few bytes in an MP3 file are subtle or not noticeable at all. ) However, this format is constantly evolving, so it can be assumed that the quality will be optimized.

Most portable audio players support WMA format along with MP3. This format is very poorly supported on alternative platforms (due to its closed nature).

Microsoft included support for digital rights management (DRM) (protection system) in WMA. Its main consequence is the inability to listen to protected compositions on computers other than the one on which the composition was downloaded from the music store.

The latest versions of the format, starting with Windows Media Audio 9.1, provide encoding without loss of English quality. lossless, multi-channel surround sound encoding and voice encoding.

In this article I want to give an incomplete list of the most common music formats. We are more familiar with some, and with some less, for example, those who use the Windows operating system on their computer are practically unfamiliar with the AIFF file format for Mac OS - an analogue of the more well-known WAV format. But that's not the point

Today there are a “great variety” of music formats; they differ from each other in different sound compression algorithms, while the degree of compression itself is expressed by such a concept as bitrate.

Uncompressed formats are not compressed. They simply unfold during opening. Although the size of these files is usually very large. The disadvantage of lossy compressed files is that it removes some data from the original file. But the advantage is that they are smaller, open faster and take up less space.

Lossy files can be high or low resolution depending on the compression ratio. The higher the quality, the less information will be lost. The bitrate corresponds to the information processed per second. Higher bitrate means more information per second. And more information per second means better sound. Now you understand the basics of compression, file types and bitrates, right?

The lower the bitrate, the worse the sound quality of the compressed, transcoded file. The audio bitrate is measured in kilobytes per second. To make it more clear what sound is depending on its bitrate, below is a table that sheds light on this issue:

  • 800 bps - 800 bits/sec - the minimum quality for the voice to be recognizable.
  • 8 kbps - 8 kbit/s - quality of voice transmission over the phone.
  • 32 kbps - 32 kbps - AM quality.
  • 96 kbps - 96 kbps - FM quality.
  • 128–160 kbps - 128-160 kbps - quality standard.
  • 192 kbps - 192 kbit/s - DAB quality (Digital Audio Broadcasting) digital radio broadcasting. Becoming the new standard for MP3 music. At this bitrate, only professionals can notice the difference in sound.
  • 224–320 kbps - 224-320 kbps - quality close to CD quality.
  • 1411 kbps - 1411 kbps - PCM audio format, similar to CD “Compact Disc Digital Audio”.

Of course, you need to remember and understand that the sound will depend on another characteristic of digital sound, such as the sampling frequency, which is responsible for representing the spectrum of the signal.

If we talked about every single audio format, we'd be here for days. Of course, you have other responsibilities and a lot of music to produce. This is the best use for each of these formats. They take up a lot of hard drive space. For a simple reason: it has the best of both worlds.

They are compressed, making them easier to handle in terms of size. But they also offer pleasant and rich sound. If you listen to streaming music, this will most likely be the case. They are useful when transferring multiple files at once, browsing an entire directory, or sharing and linking to tracks quickly.

  • 8,000 Hz - telephone, enough for speech, Nellymoser codec;
  • 1025 Hz;
  • 22,050 Hz - radio;
  • 44 100 Hz - used in Audio CD;
  • 48,000 Hz - DVD, DAT.
  • 96,000 Hz - DVD-Audio (MLP 5.1)
  • 192,000 Hz - DVD-Audio (MLP 2.0)
  • 2,822,400 Hz - SACD Super audio CD 5.1

The most common format, especially on the Internet, is MP3. It is created using a compression algorithm in such a way that while reducing the size of the data required to play back the recording and ensure playback quality, the loss of sound quality is minimal. The file size depends on the degree of compression. Thus, when creating an MP3 with an average bitrate of 128 kbps, the resulting file is approximately 1/10 the size of the original CD-Audio file.

By the way, don't forget any format

Take the test and judge for yourself. Choosing the right format depends on each context. So think about what sound you're sharing and where you're doing it. Are you using the correct format? So make smart choices and use the right format. Digital audio can be saved in different formats. Each of them corresponds to a specific file extension that contains it.

It is not an audio format itself, so its functions are discussed separately. There are a huge number of audio formats. Typically the format type corresponds to the file extension. Some file types are assigned a specific codec. Simply put, a format can be compared to a container in which audio or video can be stored using a given codec. If you don't know which program to use to open a format or other audio, we recommend using our audio converter. It is compatible with almost all existing formats.

For comparison, I will provide information about the Wav format, which supports high quality sound. At a sampling frequency of 44100 Hz, its bitrate is 1411 kb/s and 1 minute of a recorded file in this format takes up approximately 10 m of hard disk space.

So, what are the most common audio formats today:

This group of formats records and compresses audio in a way that preserves its exact original quality when it is decoded. With lossy compression, the sound undergoes certain modifications. For example, compression cuts out audio frequencies that are inaudible to the human ear. When it is decoded, the file will be different from the original in terms of the information stored in it, but it will sound almost the same.

Find out more about the most common audio formats

Some of the most common loss formats. However, this has been compromised by some independent tests. It usually provides better audio quality with the same file size. It does not change the audio sequence, and the audio encoded in this format is identical to the original. It is often used to reproduce sound in high-quality audio systems. Playback compatibility on devices and players is limited, so it is often converted to other formats before playback on the player if desired.

  • AAC (Advanced Audio Coding) - other names - MPEG-2 AAC and MPEG-2 NBC. The result of the evolution of MP3 files. With a lower bitrate, they are not inferior to MP3 quality.
  • AIFF - file format for Mac OS, uncompressed data. High sound quality.
  • ASF (Advanced Streaming Format) is a standard format for OS Mac. Large file size with high sound quality comparable to AudioCD quality.
  • AudioCD (CDA) - analog audio, high quality sound.
  • FLAC (Free Lossless Audio Codec) - free audio codec, audio compression up to 50 percent without loss of sound quality.
  • Liquid Audio (LQT, LA1) is a secure format for paid music downloads over the network.
  • MP2 (MPEG-1, Layer2) is an obsolete audio format, predecessor of MP3.
  • MP3 (MPEG-1, Layer3) is an audio format that provides acceptable sound quality with a high compression ratio. One of the most popular formats in the world.
  • VQF is an audio format, an outdated analogue of MP3.
  • WAV is a standard Windows file, high quality audio is supported. Takes up a lot of disk space.
  • WMA (Windows Media Audio) is a promising format from Microsoft. With smaller file sizes and lower bitrates, it is as good as MP3.

    • As a rule, today the term “audio” refers to everything related to sound, be it playback, processing, mixing, mastering or listening to recordings. But few people know that audio formats have constantly undergone many significant changes since their inception, either for the better or for the worse. The trouble is that, compared to the initial formats, the creators of the new formats tried to improve the sound quality, and this invariably affected the size of the playback file. Reducing the size, on the contrary, led to a loss of quality. But it was not always so.

    What devices support it?

    There are several formats that support high-resolution music. The quality of each format may vary depending on how it was recorded and at what frequencies.

    There are many other brands already flirting with handheld devices that are supposedly capable of delivering high-definition audio. It's good to note that playing high-resolution audio is not enough with a file. It requires adapted audio electronics and supports these speeds and bit frequencies. Of course, a high-end output device is also required.

    The first audio format in computer games

    The very first mention of computer sound came from the creation of games, primitive at that time, in which the sound was reproduced through the system speaker. But no matter how hard the developers of such software (software) tried, they could not achieve the required quality, compatible with reel-to-reel or cassette recorders or records.


    High-definition digital music has been around for a long time. Why does it seem to be becoming fashionable now? In addition to traction, they may have devices such as one, the answer to the market. The perpetrator of this disappearance does not have to look very far for him. Except in special cases, for most users, having their smartphone listen to music is more than enough. Even players who have settled, as they see, reduce the terrain they break through year after year.

    That is why many manufacturers have started looking for a solution on how to change the audio format so that the sound is natural. Frankly, this led to further competition that we have now. This applies not only to the reproduced material, but also to studio sound, live performances, quality or adjustment of basic parameters in terms of knowledge of physics, acoustics, etc.


    We come to the millionth question. This requires a sensitive ear and some education. If you're a high-end enthusiast, you'll probably recognize the difference first, but for ordinary mortals who just listen to music with a background meter, it's probably too much of a hassle to pay the price difference just to have that kind of quality.


    It's not just about the music player. Those that match this equipment don't come cheap. To top it off, there is music that is sold in high definition formats, which is also a little more expensive.

    Finally, this is the technical section. There isn't even a consensus among audio experts themselves as to whether these devices make that important difference. At least on paper, it appears that high-definition music is heard better, but there's a strong marketing component behind it. Are we willing to pay to be seen by the public as music lovers?

    The emergence of the WAV format

    It is believed that the first full-fledged quality of audio formats was associated with the advent of the .wav file standard and extension (this abbreviation was derived from the English word “wave” or wave). It was precisely he who became the first-born who could be processed in computer programs at a professional level.

    As usual in most of the multimedia files that we work with almost on a daily basis, we refer to video with photos or audio files, depending on the type of use we are going to make of them, we must use in some specific format, so it is also important know the main differences between them.

    But here's what we're going to talk about, these are a few alternatives that we're going to present and these will be very helpful to you when it comes to trying and converting the different types of audio files that we usually come across. If you want to know the main differences between them, we recommend that you take a look at this post that we bring to you.

    Such files already had their own characteristics: sampling frequency, sound depth, bitrate and much more. This sound was compatible even with what could be obtained after processing a regular audio CD using certain tools such as a conventional equalizer. But the size was clearly unjustified. For example, a three-minute track could take from 20 to 50 MB.


    With this program we refer to a suitable alternative to carry out conversions between multiple audio files, in addition to extracting audio from video files in the most common formats. That being said, it has a very intuitive user interface that makes it valid even for beginners in these tasks. To get started, just add this file, select the output format and click the "Convert" button.

    First of all, one of the main characteristics of this tool is that, in addition to being able to convert between different audio files, which is what interests us in this case, it also has functions for converting videos and images. Thanks to the functionality it offers, we will be able to tailor our music to play as optimally as possible on mobile devices, using a simple process. To do this, simply drag and drop the files you want to convert into the application's simple interface.

    CDs

    The audio CD format, more precisely the .cda extension, appeared at almost the same time.

    Unlike “wave” files saved on the hard drive, it cannot be edited. Today you can open it in an audio processing program, change the format by audio transcoding and save it in any place other than a CD.

    After specifying the output format and the device on which we will play them, the conversion will begin. For all these tasks, we just need to drag and drop the elements we are working with on your user interface. For example, if we play back a file at twice the speed, a simple way to interpolate would be to play back one sample out of every two.

    If you put an object at 50% of the pitch, it should double the number of samples, and a simple way to do this is to find the average of the two actual samples. Well, this signal has to be decompressed, and this takes up proportionately more CPU load, which can become "saturated". And a less saturated processor means less risk of crashing. We will analyze the most popular compression formats that will allow us to compress any files without losing information. What's fast? The most powerful?

    MP3 codec

    With the advent of the LAME MP3 Encoder codec, the music industry experienced a real shock, because such files “weighed” tens of times less than the same WAV file. Even a five-minute composition with maximum compression rarely exceeds the size of 5-7 MB. Agree, a significant breakthrough, not to mention, made it possible not only to adjust the above characteristics, but also some additional parameters in the form of ID3 tags, which contained information, say, about the artist, the name of the album and tracks, and the release date.

    What is file compression? What does squeeze mean?

    And the most used? Have you ever encountered a file that was too busy and didn't know how to get it to do less, like mail it to a friend? File compression allows us to reduce the file size. This will take up less hard drive space and be easier to send. Depending on the type of file used and the type of compression, its size will be reduced more or less.

    What compression formats are there and which ones are the most popular?

    As we just said, there are several types of compression methods. For example, it is a compression method used to compress video, audio or image files. The main feature of this compression method is that when compressed there are approximations, so the media file is reduced in size. This method looks for patterns that repeat in addition to other more advanced methods. This is achieved by reducing the file size without losing information or quality, although obviously the file size is not reduced. Unlike the previous case, this information is not lost. . When it comes to formats and compression methods, we have a wide range.

    This type has become the most popular. Look, almost the entire Internet is filled with this universal format. In general, we can say that the MP3 audio format has become a real revolution in sound. It remains one of the most popular and most in demand to this day, despite the fact that it is being replaced by other types of audio. But more on that later.

    AIFF files

    Audio formats have another variety. The so-called .aiff format was originally created for use on Macintosh computer systems.

    Only much later did a transformation occur that predetermined the compatibility of sound formats with their use on platforms with different operating systems.

    OGG format

    Music in audio.ogg format is also quite common. This standard was developed by Vorbis. However, it is worth noting that it has a number of significant disadvantages. Firstly, this is an unjustified load on the computer’s system resources, despite its minimal size. Secondly, the use of your own codecs and decoders, which the system may not automatically install. For example, when working in FL Studio Producer Edition (or XXL) in versions below 9.x.x, there was a folder with an installation file in .inf format, which had to be activated for installation after installing the main application manually (otherwise presets in this format simply would not were lost).

    Nevertheless, audio formats of this type are now available, and the sound itself looks very good.

    AMR standard

    As for this format, it is perhaps one of the most low-grade. Its origin is associated with the advent of the first clumsy mobile phones, which still could not set ringtones in .mp3 format.


    At that time, AMR could still replace natural sound with a certain amount of loss of quality. But this quality cannot be compared with what is offered by more “advanced” formats.

    MIDI

    Oddly enough, MIDI can also be classified as what is commonly called “audio formats”. Although it is generally accepted (and many, in fact, still think so) that the MIDI system is just a set of commands, one can argue with this. The abbreviation MIDI is actually a system for recording and editing certain keystrokes, pitch, tempo, key, effects, etc.

    However, there are files with the .mid or .midi extension that can be easily played in modern sequencers or studio recording programs using a standard set of sounds in the GM (General MIDI), GS (which is the same) format from Roland, or XG (Extended MIDI) from Yamaha Corporation. The first two sets contain 128 standard sounds, not counting effects, the third contains almost three times as many.

    FLAC

    Now we come to one of the most modern and unique formats of our time. Music in FLAC audio format is becoming increasingly common today. This is due to the quality that true music lovers pay attention to first of all.

    If you look at it, this format was created on the basis of the already known MP3. But if previously distribution into separate tracks was used, this is not the case in this format (for the time being). The structure consists of one or two files, one of which is informational. Only specialized software audio players can reproduce this format. The most famous can be called AIMP. Only when the main file is opened does a list of tracks recorded in the main container appear. In such a player, switching between tracks is done in the same way as in any other. But there is no chance of accidentally deleting a particular composition (as already mentioned, information about them is contained in a single file).

    Format compatibility

    Naturally, all audio formats today are compatible with each other. In other words, any standard home DVD player or software player will handle this without difficulty. The same applies to audio processing programs. Semi-professional and professional programs recognize all formats known today (even despite the specifics of operating systems). Audio editors, sequencers, additional modules such as VST, RTAS (for Windows systems) or AU (for Mac OS X) are capable of working with such formats in the so-called cross-platform mode.

    Format conversion

    There are several ways to change audio. For example, you can open a “native” format and save the file in another. You can do it even simpler. There are special converters for this. In them you can simply load the desired file of the initial format from the list, and then simply select the final one. As they say, just nothing.

    Audio Quality Processing

    It's another matter when the question concerns changing some frequencies of the source file. You can’t do this without specialized software packages. It is with their help that you can change the quality of audio files. In this case, you can change not only the standard sampling frequency of 44100 Hz, increasing it, say, to 96000 Hz, but also adjust the depth from the same 16 to 24 or 32 bits. And we’re not even talking about the fact that you can also configure the bitrate, that is, the reproducible bandwidth expressed in kilobits per second. The standard value is 128 kbit/sec. The bitrate can be changed at your discretion, but the best sound quality is achieved at around 320 kbps. Of course, not every person is able to perceive the difference between the standard sound and the maximum characteristics. However, it is worth trying once to play an audio track with different data on good equipment. Here the difference will not be long in coming.


    Moreover, in addition to all these parameters, you can edit much more. Just look at the use of software equalizers, limiters, compressors, crossovers, normalizers, de-essers, etc., etc. Each such module allows you to customize the sound, as they say, “for yourself.” And absolutely all formats known today can be processed by programs of this type.

    Final comparison

    Let's try to make some comparison between the formats used (although this is not all that there is in the world of sound).

    So! The WAV format, although “heavy”, can still be used as intermediate files during subsequent conversion in some audio reactors. These types of audio file formats are most often present when saving open projects or when recording live instruments in the studio. It is clear that the sequencer will then process the incoming information in the form of an audio stream. And then you can change the format of the audio file or save it as a preset or track as you wish.

    Formats such as audio CDs are also irrelevant today. If we take AIFF or OGG into account, they are better used in virtual studios. There is no need to talk about the AMR format at all. MIDI is useful only to musicians who know a lot about it.

    It is believed that the best audio format today is still FLAC. According to many experts and musicians, it is not just the most “advanced”, but even revolutionary compared to what existed or exists today.

    However, it is worth noting that MP3 cannot be discounted, because almost all encoded audio on DVDs or MKV files is in this format. The only difference is in the version of the codec and decoder. But the audio and video industry is not standing still in its development. It is very likely that we will soon see something new.

    Today, there are about three dozen common digital audio formats. Why it was necessary to create so many types of sound files to store one type of content and how to manage all this you will learn from this material.

    Introduction

    Surely many users prefer to use their home computer not only as a workhorse, but also as a multimedia center on which they can watch movies or family photos, as well as listen to their favorite music. Although compact digital players or mobile phones are certainly more suitable for listening to music, unlike them, a computer can do more than just play music.

    No matter how large the amount of built-in memory your music player has, most likely it will not be possible to store your entire music library in it. Moreover, you can create, edit, organize and search music using your PC. Also, do not forget that today there are about three dozen common digital audio formats, and most players are far from omnivorous and are capable of playing only some of them.

    So why was it necessary to create so many music formats to store one type of content? The thing is that the sound in the vast majority of cases is stored in a “compressed” form, since one minute of an uncompressed composition takes up about 10 MB on the hard drive. On the one hand, this doesn’t seem like much, but on the other, if you are a music lover and your collection consists of several hundred or even thousands of songs, then it becomes clear that the sound needs to be compressed in order to reduce the space it takes up on electronic storage media.

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

    Without compression

    One of the most common formats of this type can be considered the famous WAV. Sound in files with this extension is stored without any compression or changes. True, much more storage space for uncompressed files is required and therefore WAV is most widely used only in professional audio and video applications, where the sound before processing should not have a loss in quality. Storing ordinary musical compositions in this form is an unjustified wastefulness.

    To play WAV files, you do not need any special software, since this format is understood by all media players, including the standard Windows Media audio player built into the Windows system.

    Another format worth mentioning that is used to store uncompressed audio is one developed by Apple called AIFF (Audio Interchange File Format). As you might have guessed, it is most commonly used on Macintosh computers running Mac OS X systems.

    Lossless compression (lossless)

    Algorithms that compress audio files without loss work on the principle of conventional archivers. Providing not the highest level of compression (from 40 to 60%), they have virtually no effect on sound quality. It is also worth noting that in this case, the encoded data can be completely restored to its original form. Therefore, the use of lossless compression is most often used 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 pros and cons. For example, the APE codec provides slightly greater compression gains, while FLAC is more common. In general, all true music lovers store their music collections in lossless formats, since they do not delete any data from the audio stream, and files created using these codecs can be listened to even on high-quality audio equipment.

    To play lossless compressed formats, as a rule, third-party players (except WMA) are used, such as MPlayer, foobar, AIMP, Winamp, VLC and others, since all the necessary codecs are already built into them. Another option is to separately install a package of additional codecs (for example, K-Lite), after which listening to files in lossless format becomes available from almost any audio player.

    Lossy compression

    This is the most popular group of algorithms that provide the maximum (up to 10 times or even more) degree of audio compression. True, unlike previous formats, here the audio file loses quality, and how much - directly depends on the degree of its compression.

    To determine the quality of digitized sound, the most commonly used indicator is bitrate- the speed of the sound stream obtained after compression and measured in kilobits per second (kbps). As we have already said, on average, a minute of uncompressed audio takes about 10 MB, which corresponds to an audio stream of approximately 1400 kbps. After lossy encoding, its bitrate can drop to 56 kbps. At the same time, it is worth considering that in order to maintain natural sound, the flow rate must be at least 192 or 256 kbit/s. If the bitrate of the stream is 320 kbit/s or more, then the difference in sound for most people between compressed and uncompressed audio practically disappears.

    The most popular format here is definitely the famous and beloved MP3, developed by specialists from the famous MPEG (Moving Picture Experts Group). It is most widely used for encoding audio files posted on the Internet and various file hosting services due to the ability to significantly reduce the size of transferred data, which is important when the network connection speed is low.

    Other well-known formats in this series are AAC (Advanced Audio Coding) and OGG Vorbis. At the same time, being less popular, their compression algorithms are more advanced than those of their main competitor. So, with the same file size, they provide better audio quality compared to MP3. Another serious advantage of these formats is the ability to encode up to 48 audio channels for AAC and 255 for OGG, versus only two for MP3.

    It is worth noting that the WMA format, a property of Microsoft, was originally created for storing and broadcasting audio information in lossy compressed form, and lossless encoding was added to it not so long ago, starting with Windows Media Audio 9.1. The format nominally offers better compression rates than MP3, giving developers the opportunity to compare it as an alternative to competing AAC and OGG algorithms. True, the widespread use of WMA is hampered by its closed nature and limited use on many platforms (operating systems). And built-in support for digital rights management (DRM) does not add to the popularity of Microsoft's brainchild.

    Despite the fact that MP3 is inferior to its competitors, both in terms of compression efficiency and sound quality, it still continues to be the most popular audio format. The secret of such success can probably be called banal inertia of thinking, since over many years most users, hardware manufacturers and software developers have become accustomed to it. That is why MP3 files can be listened to on anything that can play digital sound - be it a mobile phone, a personal computer with any popular operating system, a portable audio player, a modern music center or a DVD player.

    And although other formats cannot yet boast such support, things are not so bad for them either. So AAC has found wide support from Apple, which uses its algorithms to store audiobooks, podcasts, music in the iTunes store and ringtones. So for fans of Macintosh computers, iPad tablets, iPhone smartphones and iPod players, this format can be considered “native”.

    WMA files can be easily played on any PC running the Windows operating system, which is the most widely used operating system in the world. However, many manufacturers of portable audio players and stationary optical disc players also support this format. But to listen to files in OGG Vorbis or AAC formats on Windows systems, you will have to install special codecs. It's not a problem though. Installing the above-mentioned free K-Lite Codek Pack will allow you to play almost any sound files on your computer using your favorite player.

    Conclusion

    In conclusion, let's look at what set of software you will need to turn your home computer into a universal tool for working with audio files. For convenience, we will divide all applications into several main groups.

    Players - serve for direct playback of sound files, and are also often used for cataloging and organizing music collections. Their number is so huge that it is impossible to count. But still, to make your choice a little easier, here are, in our opinion, the twelve most popular: Windows Media Player (built into the system), Winamp, KMPlayer, iTunes, GOM Player, jetAudio, VLC Media Player (VideoLAN), AIMP, BSPlayer, Real Player, WinDVD and Foobar2000.

    Converters - applications that can convert from one format to another. For this purpose, you can use most popular players without resorting to special programs. Although in some cases this cannot be avoided.

    Rippers (grabbers) - allow you to extract digital audio information from optical media (Audio-CD, DVD) and save it in various formats. Despite the numerous variety of grabbers, the EAC (Exact Audio Copy) application has gained the most popularity in this field, allowing you to make the most accurate copies of discs. Other popular rippers include: Audiograbber, Reaper, Easy CD-DA Extractor and others.

    Editors - programs designed for creating, recording and editing audio data. In this group there are both fairly simple programs that allow you to perform basic operations with an audio file (cut, crop, merge, normalize, etc.), and real monsters for professional work with sound. Among the small editors, one can highlight the Nero WaveEditor application, for its modest size and at the same time quite high functionality. The most popular professional audio processing solutions include: Adobe Audition, Sound Forge, Cubase, Sony Vegas Pro and others.

    Of course, purely theoretically, only one program can combine all these necessary functions, but in practice, using a single application for all tasks is not always convenient. And it is almost impossible to achieve high-quality performance of all tasks from one program.

    In any case, it is much more convenient to have several specialized applications on hand, which take up less space and do their individual tasks better.

    We'll look at different audio file formats:

    WAVE (.wav)- the most widely used audio format. Used in Windows OS to store sound files. It is based on the RIFF (Resource Interchange File Format) format, which allows you to save arbitrary data in a structured form. Various compression methods are used to record audio because audio files are large. The simplest compression method is Pulse Code Modulation (PCM), but it does not provide good enough compression.

    AU (.au,.snd)- audio file format used on Sun workstations (.au) and the NeXT operating system (.snd). It became widespread on the Internet, at an early stage of its development it played the role of a standard format for audio information.

    MPEG-3 (.mp3)- audio file format, one of the most popular today. Was designed to store sounds other than human speech. Used for digitizing music recordings. Previous versions of the format: MP1 and MP2. When encoding, psychoacoustic compression is used, in which sounds that are poorly perceived by the human ear are removed from the melody. Early versions provide worse compression, but are less demanding on computer resources during playback. The characteristics of the processor directly affect the sound quality - the weaker the processor, the greater the sound distortion.

    MIDI (.mid)- digital interface of musical instruments (Musical Instrument Digital Interface). This standard was developed in the early 80s for electronic musical instruments and computers. MIDI defines the exchange of data between music and sound synthesizers from different manufacturers. The MIDI interface is a protocol for transmitting musical notes and melodies. But MIDI data is not digital audio—it's a shortened form of recording music in numerical form. A MIDI file is a sequence of commands that record actions, such as pressing a key on a piano or turning a knob. These commands sent to the MIDI file playback device control the sound, a small MIDI message can cause a sound or sequence of sounds to be played on a musical instrument or synthesizer, so MIDI files take up less volume (audio unit per second) than equivalent digitized files. sound.

    MOD (.mod)- a musical format, it stores samples of digitized sound, which can then be used as templates for individual notes. Files in this format begin with a set of sound samples, followed by notes and duration information. Each note is played using one of the sound patterns shown at the beginning. This file is relatively small and has a note-based structure. This makes it easier to edit using programs that simulate traditional music recording. It, unlike a MIDI file, completely defines the sound, which allows it to be played on any computer platform.



    IFF (.iff)- Interchange File Format – a format originally developed for the Amiga computer platform. Now also used on compact discs in the form of CD-I. Its structure is very similar to that of the RIFF format.

    AIFF (.aiff ) - Audio Interchange File Format - a format for exchanging audio data, used on Silicon Graphics and Mac computer platforms. In many ways it resembles the Wave format, but unlike it allows the use of digitized audio and templates. Many programs can open files in this format.

    RealAudio (.ra, .ram)- a format developed for playing sound on the Internet in real time. Developed by Real Networks (www.real.com). The resulting quality, at best, corresponds to a mediocre audio cassette; for high-quality recording of musical works, the use of the mp3 format is more preferable.

    4.3. MIDI and digital audio: advantages and disadvantages

    The WAVE format is one of many, but far from the only format for recording digital audio. Unlike MIDI data, digital audio data actually represents sound recorded in thousands of units called samples. Digital data represents the amplitude (or loudness) of a sound at discrete points in time. The sound of digital data does not depend on the playback device and therefore their sound is always the same. But you have to pay for this with large volumes of sound files.

    MIDI data is to digital data what vector graphics are to raster images. That is, MIDI data depends on audio playback devices, but digital data does not. Just as the appearance of vector graphics depends on the printer or monitor screen, the sound of MIDI files depends on the MIDI device used to play them. Likewise, the sound of a melody played on a concert piano will be different from the sound of the same melody played on a regular piano. Digital data, on the other hand, is identical and independent of the playback system. The MIDI standard is in this sense similar to the PostScript standard and allows you to control instruments in a clear language.

    Compared to digital audio, MIDI has the following advantages:

    § MIDI files take up less memory, and the size of these files does not affect sound quality. On average, MIDI files are 200 to 1000 times smaller than digital files and therefore occupy a small amount of RAM, disk space, and do not require large CPU resources.

    § In some cases, MIDI files sound better than digital audio files. In this case, the sound source of MIDI files must be of high quality.

    § You can change the length of MIDI files, changing the tempo of the sound while maintaining sound quality and volume. MIDI data can be easily edited, even at the individual note level. You can manipulate small segments of a MIDI song (with millisecond precision), which is not possible with digital audio.

    The main disadvantage of a MIDI file stems from its advantages. Since MIDI data is not itself a sound, playback will only be as accurate as the MIDI data playback device that is identical to the device that was used to create the original file. Even the sound of a MIDI instrument according to the General MIDI standard depends on the electronic playback device and the method used. MIDI audio is not used to reproduce speech.

    The main advantage of digital audio over MIDI audio is that the quality of digital audio reproduction is always consistent, and this is where MIDI audio is inferior to digital audio. There are two reasons why you should work with digital audio:

    § a wider selection of programs and systems that support digital audio;

    § To prepare and create digital sound elements, no knowledge of music theory is required, which cannot be said about MIDI data.

    Almost any computer user periodically listens to music on it, which is stored electronically. There are quite a lot of formats for storing music, each of them was developed for specific tasks:

    • Playback from CD;
    • Sound accompaniment of a computer game;
    • Audio track in ;
    • Streaming playback over the Internet;
    • Ringtones for mobile phones.

    Let's try to understand some of them, as well as...
    Basic definitions

    • Bitrate – the amount of information used during encoding to reproduce 1 second. The higher it is, the less distortion, and the sound matches the original as closely as possible.
    • Lossless – audio encoding without quality loss. When converting to lossless formats and back, we get absolutely the same sound.
    • Lossy – compression formats designed for the fact that a person simply physically cannot hear certain frequencies that are skipped during the conversion process. This can significantly save on disk space.

    Audio-CD

    The format that ushered in the era of digital sound after the transition from vinyl records. It was adopted as a standard in 1979 by Philips and Sony. In the audio-CD format, music can be physically stored only on optical media; when recording to a hard drive, the audio track must be converted.

    Thanks to the highest sound quality and the ability to play on any player, the format remains very popular, despite the fact that it is quite outdated.

    Flac

    Perhaps the most common format for storing losseless music. Compared to other codecs that provide lossless audio compression, flac, developed by xiph.org, is completely free and produces a minimal output file size.

    Mp3

    The most popular music format, adopted as an unofficial standard for any playback device. Its popularity is based on the fact that, thanks to cutting off frequencies that are inaudible to the ear, with almost the same sound quality, an mp3 file is 30% of the original lossless file.

    The first audio track in mp3 format appeared back in 1994. One of the reasons for its popularity is the ability to store a variety of additional information in audio file tags and the convenience of organizing a music library.

    Ogg

    A new lossy format, released in 2002 as a free alternative to paid formats. Unlike its predecessors, in particular mp3, it allows for the possibility of multi-channel encoding and storage of multi-channel audio. Most widely used in video games.