Record audio files with different sound quality. Audio formats: types and their descriptions

I recently received the following letter:

Hello site, MP3 is the most popular audio format, but there are so many others such as AAC, FLAC, OGG and WMA that I'm not really sure which one I should use. What is the difference between them and which one should I use to store my music?

The question is quite popular, I will try to answer it simply but clearly.

We've already talked about the difference between lossless and lossy, but in short, there are two types of audio quality:

• lossless: FLAC, ALAC, WAV;
• lossy: MP3, AAC, OGG, WMA.

The lossless format preserves full audio quality, in most cases CD-quality, while the lossy format compresses files to save space (of course, the audio quality is degraded).

Uncompressed data storage formats: FLAC, ALAC, WAV and others

• WAV and AIFF: Both WAV and AIFF store audio uncompressed, meaning they are exact copies of the original audio. The two formats are essentially the same quality; They just store data a little differently. AIFF is made by Apple, so you may see it more often in Apple products, while WAV is pretty much universal. However, since they are uncompressed, they take up a lot of unnecessary space. If you don't edit audio, you don't need to store audio in these formats.
• FLAC: Free Lossless Audio Codec (FLAC) is the most popular lossless audio storage format, making it a good choice. Unlike WAV and AIFF, it compresses the data slightly, so it takes up less space. However, it is considered a format that stores lossless audio, the quality of the music remains the same as the original source, so it is more efficient to use than WAV and AIFF. It is free and open source.
• Apple Lossless: Also known as ALAC, Apple Lossless is similar to FLAC. This is a lightly compressed format, however, the music will be preserved without loss of quality. Its compression is not as efficient as FLAC, so your files may be a little larger, but it is fully supported by iTunes and iOS (while FLAC is not). So, if you use iTunes and iOS as your main software for listening to music, you will have to use this format.
• A.P.E.: APE - has the most aggressive compression algorithm for lossless music storage, that is, you will get maximum space savings. Its sound quality is the same as FLAC, ALAC, but there are often compatibility issues. In addition, playing this format puts a much higher load on the processor to decode it, since the data is highly compressed. In general, I would not recommend using this format unless you are limited in available memory and have software compatibility issues.

Compressed audio storage formats: MP3, AAC, OGG and others

If you just want to listen to music here and now, chances are you'll be using a lossy format. They save a ton of memory, leaving you with more room for songs on your portable player, and if high enough, they will be indistinguishable from the original source. Here are the formats you are likely to encounter:

• MP3: MPEG Audio Layer III, or MP3, is the most common lossy audio storage format. So much so that it has become synonymous with downloadable music. MP3 is not the most efficient format of all, but it is certainly the most well supported, making it the best choice for compressed audio storage.
• A.A.C.: Advanced Audio Coding, also known as AAC, is similar to MP3, although it is slightly more efficient. This means you can have files that take up less space but have the same sound quality as MP3. The format's best evangelist today is Apple's iTunes, which made AAC so popular that it has become almost as widely known as MP3. I've only had one device in a very long time that couldn't play AAC, and that was a few years ago, so you can safely use this format to store your music.
• Ogg Vorbis: The Vorbis format, known as Ogg Vorbis due to its use of an Ogg container, is a free alternative to MP3 and AAC. Its main feature is that it is not limited by patents, but you as the end user are not affected at all. In fact, despite its openness and similar quality, it is much less popular than MP3 and AAC, which means that fewer programs support it. Thus, we do not recommend using it to avoid software compatibility issues.
• WMA: Windows Media Audio is Microsoft's own proprietary format, similar to MP3 or AAC. It doesn't offer any advantages over other formats, and it's also not very well supported outside of the Windows platform. We do not recommend that you rip CDs to this format unless you know for sure that all music will be played on the Windows platform, or on players compatible with this format.

So what should you use?

Now that you understand the difference between each format, which should you use to rip or download music? In general, we recommend using MP3 or AAC. They are compatible with almost every player, and both are indistinguishable from the original, if . Unless you have special needs that dictate otherwise, MP3 and AAC are your best bet.

However, there is something to be said for storing your music in a lossless format like FLAC. While you probably won't notice higher quality, lossless is great for storing music if you plan on converting it to other formats later, since converting a lossy format to another lossy format (such as AAC to MP3) will result in files appear to be of noticeably lower quality. Therefore, for archival purposes we recommend FLAC. However, you can use any lossless format, as you can convert between lossless formats without changing the quality of the file.

As real field research has shown, the good idea to rank the top ten best of the best audio formats turned out to be a fundamentally impossible task.

The competition conditions are too different for unequal participants. In addition, some corruption schemes or lobbies of transnational corporations in the field of audio recording interfere with our good cause of helping people choose the best audio product.

The most popular MP3 format in the world has become a leader in popular love solely because of multi-billion dollar investments in promotion. And if you take the sound quality, it’s so-so. And even in terms of compression and saving disk space, it’s also not the most highly compressed.

Therefore, a compromise decision was made - to divide the experimental subjects into three groups and compare and identify leaders within the groups.

Three types of audio formats

• Without compression.
• Lossless compression.
• Lossy compression.

Uncompressed audio recording formats show their best only on high-quality, professional sound reproduction equipment.

If you have a budget tablet or smartphone in your hands, then wonderful music will sound in your gadget, but you won’t hear it simply because the software and hardware resource and speakers or headphones are not capable of reproducing such high quality sound.

On the other hand, if you run an MP3 audio recording through professional stereo systems and amplifiers, you will hear such noise and grinding in the speakers that, again, this kind of use will be completely pointless.

Audio rating by type of sound-reproducing equipment

1. For professional equipment – ​​uncompressed audio formats.
2. For semi-professional equipment – ​​compressed audio formats. But no losses.
3. For budget equipment – ​​compressed and lossy audio formats.

In the first case, the equipment is so expensive that worrying about saving money on media is simply ridiculous.

In the second case, the owner of an Apple device for a thousand dollars will also obviously be able to spend a couple of hundred bucks on voluminous memory.

In the third case, since you’ve barely managed to scrape together enough money for a cheap smartphone, saving on the size of stored music is very important. Well, no one is going to listen to a symphony orchestra on the phone on Hi-Fi anyway. Maybe download a ringtone from the classics for fun, to look like a cool pepper in the eyes of tomatoes.

This is where the overture ends, let’s begin presenting the theme.

This includes uncompressed formats.

• PCM – pulse code modulation. The original analog sound is sampled “as is”, without any changes.

PCM is the most common audio recording format used on CDs and DVDs. Multi-channel Dolby, surround, with high-quality speakers, the sound is almost one-on-one with a live performance.

If you like to sit down in front of your home theater and immerse yourself in empathy for the main and secondary characters of the movie, this is it.

WAV

Quite an ancient format, developed back in 1991. Well, that's how the old masters always thought about high quality.

Many people consider WAV to be an uncompressed format. But in fact, this is a container and it can also contain compressed files.

In most cases, WAV contains uncompressed PCM audio. Therefore the quality is high. But for one minute of recording, about 32MB of memory is spent.

Fairly good compatibility on Windows and Mac.

AIFF

Analogue of WAV from Apple developers. It’s also a container and also most often contains sound in PCM format. Good compatibility with Windows.

Lossy compressed audio formats

Truly popular formats for everyone.

MP3

According to the MPEG-1 Audio Layer 3 standard. It appeared back in 1993 and instantly won everyone’s love precisely for its efficiency in memory consumption.

• You can store the entire discography of your favorite band on one CD.
• Throw a few discs into the glove compartment and you can enjoy music all the way from Kaliningrad to Vladivostok.
• During this time, you can listen to all the books of all writers worthy of listening to them.

The MP3 format is such a sound eunuch, from which everything that I didn’t want was cut out, but the ability to accumulate and save began to appear. So MP3 is a very economical format.

The main advantage is that it is supported on everything that plays and sings.

A.A.C.

An advanced audio encoding method. The younger but more advanced brother of MP3. Has slightly improved sound characteristics and a higher compression ratio.

Applies to Android, iOS, iTunes, YouTube, Nintendo and the latest versions of PlayStation.

Also a popular format, but for a little more advanced people. Which is reflected in the title.

OGG

In general, it is not a format, but a container and, in fact, the name OGG does not indicate anything regarding the sound contained in it.

However, most often it contains the Vorbis codec.

• Significantly improved sound quality relative to other lossy audio formats.
• It is possible to record files with the same sound quality with less weight.

An even more economical format than MP3.

The problem is that the OGG format is free, so no one invests money in its promotion. So it may not be supported everywhere and incompatibilities may arise.

Then you will have to convert to MP3.

WMA

A proprietary format from Microsoft, therefore, although it is an improved version of MP3 and OGG, it is not widely used and is not supported on most devices and platforms.

Advice. If it is possible to use WMA instead of MP3, use the former. More economical and more pleasant to hear.

For owners of advanced, expensive devices, branded mobile and desktop computers, complete with high-quality headphones and speakers.

The disadvantage of such formats is that the file sizes of the same recording duration will be approximately two or three times larger.

However, although lossless compression is declared, do not confuse it with Hi-Fi audio recordings. There are still minimal losses, albeit noticeable only for musicians.

FLAC

Free lossless audio codec. Its advantage is its wide popularity, almost like MP3.

• The compression rate is up to 60% of the original file.
• Supported on most software platforms and devices.

Can be a beneficial alternative when burning CDs. Almost indistinguishable in sound, but a bonus in the form of saving almost half of the disk space.

ALAC

The format is for owners of Apple brand devices, as it may not be supported on others.

Slightly less good than FLAC in terms of compression ratio.

But Apple owners simply have no choice - the free FLAC format is not available on iOS and iTunes.

WMA Lossless

Improved version of the above WMA. Slightly inferior to FLAC and ALAC. It has a significant advantage over ALAC, since WMA is well supported on Windows and Mac.

However, it is not widely used, so if you use many different devices, incompatibility is likely.

Verdict

Well, we’ve looked at all the most famous, common and best-performing audio formats and briefly talked about the features of their use.

So now you can confidently decide in which case and into which format it is better to convert your sound recordings, music and audiobooks.

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 sizes. 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 hearing. 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 the company " 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 other 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 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.

Sound is a physical natural phenomenon that propagates through air vibrations and, therefore, we can say that we are dealing only with wave characteristics. The task of converting sound into electronic form is to repeat all of its wave characteristics. But the electronic signal is not analog, and can be recorded through short discrete values. Even though they have a small interval between each other and are practically imperceptible, at first glance, to the human ear, we must always keep in mind that we are only dealing with the emulation of a natural phenomenon called sound.

This recording is called pulse-code modulation and is a sequential recording of discrete values. The capacity of the device, calculated in bits, indicates how many values ​​simultaneously in one recorded sample the sound is taken from. The higher the bit depth, the more closely the sound matches the original.

Any sound file can be presented, so that you can understand it more clearly, as a database. It has its own structure, the parameters of which are usually indicated at the beginning of the file. Then there is a structured list of values ​​for certain fields. Sometimes instead of values ​​there are formulas that allow you to reduce the file size. To make it completely clear to you, I will say that writing a file to your hard drive is similar to how you create tables in Microsoft Excel. Naturally, these files can only be read by specialized programs that contain a reading block.

PCM stands for pulse code modulation, which is translated as pulse code. Files with this exact extension are quite rare (I have only seen them in the 3D Audio program). But PCM is fundamental to all audio files. I would not say that this is a very economical method for storing data on a disk, but I think that you will never get away from this, and the volume of modern hard drives already allows you to ignore a couple of tens of megabytes.

Research into economical storage of audio data on disk. If you come across this abbreviation, then know that you are dealing with difference RSM. The basis of this method is the completely justified idea that the calculations are much more cumbersome compared to the fact that you can simply indicate the difference values.

Adaptive DPCM. Agree that when specifying simple difference values, a problem may arise due to the fact that there are very small and very large values. As a result, no matter how super-accurate the measurements are, there is still a distortion of reality. Therefore, a scalability factor is added to the adaptive method.

The simplest storage of discrete data. I would say direct. One of the file types in the RIFF family. In addition to the usual discrete values, bit depth, number of channels and volume levels, wav can contain many more parameters that you most likely did not even suspect - these are: position marks for synchronization, the total number of discrete values, the order of playback of various parts of the audio file, and there is also space for you to place text information there.

Resource Interchange File Format. A unique system for storing any structured data.

This storage technology originates from Amiga systems. Interchange File Format. Almost the same as RIFF, only there are some nuances. Let's start with the fact that the Amiga system was one of the first in which they began to think about software-sampling emulation of musical instruments. As a result, in this file the sound is divided into two parts: what should sound at the beginning and the element of what comes after the beginning. As a result, the beginning sounds once, then the second piece is repeated as many times as you need and the note can sound indefinitely.

The file stores a short sample of the sound, which can then be used as a template for the instrument. In other words, a sample stitched into the synthesizer.

AIF or AIFF

Audio Interchange File Format. This format is common on Apple Macintosh and Silicon Graphics systems. Contains a combination of MOD and WAV.

AIFC or AIFF-C

The same AIFF, only with specified compression parameters (compression).

Again, the same race to save space. The file structure is much simpler than wav, but the data encoding method is specified there. The files weigh very little, which is why they have become quite widespread on the Internet. Most often you can find m-Law parameters 8 kHz - mono. But there are also 16-bit stereo files with frequencies of 22050 and 44100 Hz. This audio format is designed to work with audio on SUN, Linux and FreeBCD operating systems.

A file that stores messages to the MIDI system installed on your computer or device.

The most scandalous format in recent times. To explain the compression parameters it uses, many people compare it to jpeg for images. There are a lot of bells and whistles in the calculations, which cannot be listed, but the compression ratio of 10-12 times speaks for itself. If they say that there is quality there, then I can say that there is not much of it. Experts talk about sound contouring as the biggest drawback of this format. Indeed, if you compare the music with the image, the meaning remains, but the small nuances are gone. The quality of MP3 still causes a lot of controversy, but for “ordinary non-musical” people the losses are not clearly noticeable.

A good alternative to MP3, albeit less common. It also has its drawbacks. Encoding a file into VQF is a much longer process. In addition, there are very few free programs that allow you to work with this file format, which, in fact, affected its distribution.

Eight-bit mono format from the SoundBlaster family. You can find it in a large number of old programs that use sound (not music).

NSOM

Same as VOC (eight bit mono), but only for Apple Macintosh.

U-Law standard format. 8 kHz, 8 bit, mono.

Real Audio or audio streaming. A fairly common system for transmitting sound in real time over the Internet. The transfer speed is about 1 KB per second. The resulting sound has the following parameters: 8 or 16 bits and 8 or 11 kHz.

There are two types. One is the same AU for SUN and NeXT. The other is an 8-bit mono file for PCs and Macs with different sampling rates.

Compressed using special lossless audio codecs, it can be restored with absolute accuracy if desired.

If you take an ordinary Audio CD with analog audio, record it in WAV format for sound without compression, then compress the WAV using the lossless codec, then decompress the resulting audio file into WAV and burn the result to a blank CD, you can get two completely identical Audio CD.

The advantage of lossless for storing an audio collection is that the quality of the recordings is much higher than that of lossy codecs, and they take up less space than uncompressed audio. True, lossy files are smaller in size than lossless music files. Most modern player programs understand the lossless format. Those programs that are not able to play it can easily learn it using the lossless plugin. What are lossless audio formats?

Audio formats without loss of quality

A true music lover is unlikely to be satisfied with the sound of music recorded in Ogg Vorbis or MP3 compression formats. Of course, if you listen to audio recordings on household audio equipment, sound defects cannot be detected by ear, but if you try to play a compressed file on high-quality Hi-Fi equipment, sound defects will immediately become apparent. Of course, creating a collection of quality music on CD or vinyl records is not easy. There is a reasonable alternative to this path for lovers of high-quality sound - lossless music. It can be stored on a PC in a form that allows the original music parameters to remain unchanged, even if compression is applied. This way simultaneously solves the problems of high quality music and its compact storage, because audio equipment for listening (headphones, speakers, amplifiers) has a very affordable price.

Uncompressed audio formats without loss of quality:

• CDDA is an audio CD standard;
• WAV - Microsoft Wave;
• IFF-8SVX;
• IFF-16SV;
• AIFF;

Compressed formats:

• FLAC;
• APE - Monkey's Audio;
• M4A - Apple Lossless - high-quality music format from Apple;
• WV - WavPack;
• WMA - Windows Media Audio 9;
• TTA - True Audio.
• LPAC;
• OFR - OptimFROG;
• RKA - RKAU;
• SHN - Shorten.

FLAC format

The most common format is the format What distinguishes it from lossy audio codecs is that no data is removed from the audio stream when used. This makes it possible to successfully use it to play music on Hi-Fi and Hi-End equipment, as well as to create an archive of a collection of audio recordings.

The great advantage of the format is its free distribution. This is important for musicians who record their own music. The format has recently gained great popularity, thanks to which its support is included in the vast majority of media players.

APE format

Unlike FLAC, the APE format only has codecs and plugins designed for the Windows platform. For other platforms, there are expensive solutions from third-party software manufacturers. The algorithm is capable of achieving lossless compression of audio information by approximately 1.5-2 times. It includes three main encoding stages, of which only one is based on the use of properties inherent in sound for compression. The rest are similar to regular archivers. Despite the fact that the compression algorithm is distributed free of charge, license restrictions are such that it is practically inaccessible to amateur musicians.

Apple Lossless Format

High quality lossless music can be listened to using Apple's audio compression codec without sacrificing quality. This format was developed by Apple for use on its own devices. The format is compatible with iPod players that have special dock connectors and the latest firmware. The format does not use specific rights management (DRM) tools, but the container format contains such capabilities. It is also supported by QuickTime and is included as a feature in iTunes.

The format is part of freely available libraries, which makes it possible to organize listening to files in Windows applications. In 2011, Apple published the source codes of the format, which opens up broad prospects for the codec. In the future, it may become a serious competitor to other formats. The tests showed good results. Compressed files range in size from 40-60% of the size of the originals. The decoding speed is also impressive, which justifies its use for mobile devices whose performance is low.

One of the disadvantages of the codec is that the extension of the audio files matches the audio codec. This leads to confusion, because AAC is not a high-quality music format. Therefore, it was decided to store the data in an MP4 container with the .m4a extension.

Among other formats, it is worth mentioning Windows Media Audio 9 Lossless, which is part of the Windows Media application. It works with Windows and Mac OS X. However, users do not respond very favorably to it. There are often problems with codec compatibility, and the number of supported channels is limited to six.

WavPack format

WavPack is another freely distributed audio codec that compresses audio information without loss of quality. WavPack integrates an exclusive combined mode that allows you to create two files. One of the files in this mode is created with relatively little loss of quality.wv, which can be played independently. The second “.wvc” file corrects the previous “.wv” and, in combination with it, makes it possible to fully restore the original. Some users may find this approach promising, since there is no need to choose between two types of compression - both will always be implemented.

Also worthy of attention is a video codec with high-quality sound - lagarith lossless codec. It works quickly and efficiently.

Software for listening to lossless audio

Software players did not immediately learn to work with specific lossless codecs that can reproduce sound without loss.

WinAmp Player

Capable of handling almost all music playback formats without lossless quality. What a good lossless player is can be understood by its example. It is able to correctly handle the processing of individual tracks in lossless format. This is a typical problem with FLAC or APE codecs. It consists of digitizing the entire audio disc at once and recording it in one file without dividing it into tracks. An additional file with the extension .cue is designed to solve the problem of dividing into tracks. It contains a description of the access parameters for each album track. An ordinary player plays the entire lossless file. The player for lossless AIMP perfectly reproduces most audio formats and recognizes tracks in a lossless file.

Digital players with lossless support

Users respond well to the digital players jetAudio, Foobar2000, Spider Player. There are no fundamental differences between them. The choice of any device is based on the subjective opinion of a music lover about the convenience of the interface for lossless playback. You can find out what a lossless format is by testing these players.

The Apple Lossless format is played using iTunes. In addition, this codec is supported by the popular video player VLC.

Owners of Apple-compatible computers can use two interesting programs: Vox and Cog.

They support the following lossless formats:

• Apple Lossless;
• FLAC;
• Monkeys Audio;
• Wavpack.

In addition to this, there are many useful features, for example, Last.fm services are supported.

Owners of Windows computers can use any application that is compatible with music codecs without loss of quality: Foobar2000 or WinAmp. Winamp requires special plugins. Lossless music plays well on iTunes and KMPlayer. An advantage of iTunes that other players do not have is the ability to support tags.

Lossless compatible devices

It is unlikely that the owner of a music library will want to spend time converting files from FLAC format to MP3 in order to be able to listen to recordings on his gadget. A smartphone or tablet has limited capabilities that are incomparable to a computer, but nevertheless, many mobile devices play lossless formats.

For example, owners of Android devices can use the andLess player. It is capable of playing FLAC, APE, uncompressed WAV and other formats supported by Android.

The situation is worse for owners of devices on the Blackberry platform. Only owners of the Bold 9000 and 8900 and later models can listen to the lossless format.

Owners of Apple devices can use the ALAC codec without any problems. It is supported by iPod (except shuffle), iPhone and iPad. For FLAC format, you can download FLAC Player from the App Store.

The FLAC codec is supported by Samsung Galaxy devices, some Sony Ericsson smartphones and iriver players.

Stationary devices from many manufacturers also received support for FLAC. Media players and media centers allow you to do without a personal computer when listening to songs without loss of quality.

It is still far from full support for absolutely all formats, but it is enough that the media player understands the FLAC codec - the most common codec for high-quality lossless music. What is lossless playback equipment?

Listening equipment

To truly enjoy the sound quality, you need special equipment: headphones, amplifiers, speakers. The easiest way, of course, is with headphones. If you intend to enjoy music while sitting at your computer, these are best suited. Users respond well to products from Koss and Sennheiser. Particular attention should be paid to the size of the membrane. The larger it is, the better the sound. It is important not to be deceived. Some manufacturers put a small membrane in large ear pads - such headphones look solid, but the sound is only suitable for listening to mp3s.

It is difficult to recommend anything to fans of high-quality sound equipment (Hi-Fi or Hi-End). The choice in this area is limited only by budget and tastes. Equalizer, amplifier, acoustics - the choice of these devices has many options. PC owners who are choosing a high-quality one are better off choosing budget monitor speakers from any well-known brand. Users respond well to the Microlab SOLO series acoustics. To make lossless music sound good, it is important to purchase acoustics with a subwoofer. unable to cope with the reproduction of the lower frequency band.

Results

New digital sound formats have made it possible for lovers of high-quality music to acquire their own libraries on large-capacity storage media and listen to their favorite compositions in high quality, saving quite a lot of money and quite a lot of space. The ideal option, of course, is a full set of Hi-End equipment, but budget options will also bring great pleasure to music lovers. After all, the experience of listening to music is incomparable to MP3 on plastic speakers.