Tag: is digital audio better than analog

What is digital audio?


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What is digital audio?

Digital Audio

Digital sound is nothing more than a combination of numbers.

DIGITAL AUDIO

With a certain algorithm, sound, such as air pressure, is converted into data streams and encoded for further processing and playback. Depending on the algorithm used, the music file has one format or another, one or another extension.

Remember that along with digital sound, there is analog sound, which is represented by a continuous electrical signal that reflects the change in the sound wave. The analog to digital sound conversion is a setting of the numerical value of the amplitude at a given time with a given density of values. Consequently, the more values ​​that are recorded, the more reliable and accurate the image of the digitized sound fragment is recreated. With such digitization, very voluminous data matrices emerge that, depending on the format used, differ in the sound quality / volume ratio of the final file.

Perhaps the main advantage of digital audio over analog is the ability to store and copy data indefinitely without losing the original quality (whereas when copying from one analog medium to another, a decrease in recording quality is quite noticeable).

The most widespread and popular digital audio format today is MP3 (MPEG Layer 3). It was developed, after a series of intermediate formats and investigations, started in 1987, by the Fraunhofer Institute in Germany.

The developers of the format were faced with the task of simplifying and reducing the cost of shipping long musical fragments. As you know, one minute of a stereo signal from a CD (16 bit, 44.1 kHz sample rate) takes up about ten megabytes of memory. At the same time, unlike text or graphic files, the audio signal cannot be compressed without loss of quality. Thus, modem transmission of an uncompressed composition from an audio CD lasting 3 minutes at a data transfer rate of, say, 24 kbps will take several hours. Scientists at the Fraunhofer Institute managed to achieve multiple file size compression: on average, one minute of a compressed audio signal in MP3 format takes about 1 megabyte. The principle of compression is based on the removal of “unnecessary” sounds from the music file, to which the human ear is immune, or which duplicate each other.

The main factor that determines the relationship between file size and sound quality within a given format is the bit rate. Bit rate is an indicator of how much information a second of sound encodes. The higher it is, the less distortion and the closer the encoded composition is to the original. The most common on the Internet are compositions with 128 and 192 Kbps bitrates. The maximum bitrate supported by programs and devices that work with MP3 is 320 Kbps. In practice, only an expert or a professional who works with sound can notice the differences between an MP3 file with a 320 bit rate.

To optimize the size of MP3 music files while maintaining decent quality, a variable bit rate (abbreviation VBR – variable bit rate) is used. In this case, the encoding program divides the file into fragments of different spectral saturation and encodes them with a suitable bit rate. Most modern MP3 players support variable bit rate playback. A significant advantage of MP3 files is that they can contain the name of the artist, the name of the track and the album, the year of its release, etc. The set of this data is called ID3 tags. Most modern gamers can read and display them on the screen.

In 2001, Swedish Coding Technologies and Thomson Multimedia developed the MP3 Pro codec. It is MP3-based and as a result is fully MP3 backward compatible and only partially forward compatible. It uses SBR (Spectral Band Replication) technology, so the codec provides good quality at low bit rates. However, the encoding quality at medium to high bit rates is inferior to that of most other codecs. For this reason, this format is mainly used for broadcasts on the Internet and demonstrations of fragments of new musical compositions.

Another type of MP3 was the development of MP3 Surround, recently introduced by the creators of MP3: the Fraunhofer Institute. This format repeats all the characteristics of multi-channel sound, while still being compatible with standard stereo MP3: information describing the spatial characteristics of the sound is recorded on an additional track. By playing files of this format on special equipment capable of reading this track, you can obtain surround sound that conforms to the Surround 5.1 standard.


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Digital audio encoding

Digital audio encoding

Digital audio encoding

In fact, one or another digital form of representation of analog audio signals is already a coding method – a sequence of numbers that describes an analog audio signal is itself a digital code.

Digital Audio Encoding

However, the encoding that we are going to talk about now is something else. Now let’s look at the methods of encoding digital audio signals.

A digitized audio signal “in its pure form” is a fairly accurate, but not the most compact, way of recording the original analog signal.

Judge for yourself. To obtain complete information about the original analog signal in the frequency range 0-20 kHz (in the audible frequency range), the analog signal must be sampled at a frequency of at least 40 kHz. Therefore, the CD – DA standard (the standard for recording data on audio CDs familiar to all) establishes the following encoding parameters: recording of two or one channel in PCM format with a sampling frequency of 44.1 kHz and a 16-bit quantization bit depth. One hour of music in this format takes up approximately 600 MB of space (60 minutes * 60 seconds * 2 channels * 44100 samples per second * 2 bytes per sample = approximately 605 MB). Taking into account that, for example, the music collection of an ordinary music lover may have 5,000 tracks with an average length of about 3 minutes each, the amount of memory required to store it in its original digital form is quite significant. Awesome. Therefore, storing relatively large amounts of audio data, ensuring fairly good sound quality, requires the use of various “tricks” to compress the data.

In general, all existing methods for encoding audio information can be conditionally divided into only two types.

1. Lossless data compression (“Lossless Encoding”) is a method of encoding (compacting) digital audio information, which enables one hundred percent recovery of the original data from the compressed transmission (the term ” original data “here means the original form of the digitized audio data). This method of data compression is used in cases where one hundred percent absolute preservation of the quality of the original audio data is required. Lossless compression algorithms that exist today can reduce the volume of data occupied by 20-50% and at the same time guarantee a 100% recovery of the original digital material from the compressed data. The operating mechanisms of such encoders are similar to the operating mechanisms of general data archivers, such as ZIP or RAR, but at the same time they are specially adapted to compress audio data …. Lossless encoding While it is ideal in terms of preserving the quality of audio materials, it cannot provide a high level of compression.

2. There is another more modern way to compact data. This so-called lossy data compression (Engl. “Lossy encoding”) The purpose of encoding is to achieve the highest data compression rate by all means while keeping sound quality at an acceptable level. The idea behind lossy encoding is based on two simple underlying considerations:

original digital audio data is redundant: it contains a lot of unnecessary information that is useless to the ear, which can be removed, thereby increasing the compression ratio;
Requirements for the sound quality of audio material may vary and depend on specific purposes and areas of use.
Lossy encoding is therefore called “lossy”, which results in the loss of some of the audio information. Such encoding leads to the fact that the decoded signal, when reproduced, sounds similar to the original, but in reality it is no longer identical to it. Most lossy coding methods rely on the use of the psychoacoustic properties of the human auditory system, as well as various tricks associated with resampling and resampling the signal. In frequency, during the compression process, the encoder analyzes the audio data to identify various details of the sound that can be ignored. Disguised frequencies, inaudible and inaudible sound details can be sacrificed for a higher compression ratio. Where intelligibility is only important in sound (for example, in telephony, where the presence of frequencies above 4 kHz is not necessary), the audio information during the encoding process undergoes a serious “simplification”, which, together with the use of successful “smart” quantifiers and “greedy” data compression algorithms.

Digital audio formats

Digital audio formats

Digital Audio

The digital audio format is a format for presenting audio data used in digital audio recording, as well as for additional storage of recorded material on a computer and other electronic media, so-called audio media.

digital audio

The audio file (a file containing a sound recording) is a computer file consisting of information about the amplitude and frequency of sound, saved for later playback on a computer or player.

Varieties of digital audio formats.

There are several concepts of audio format.

The digital representation of the audio data depends on how the digital-to-analog converter (DAC) quantizes. In sound engineering, two types of quantization are currently the most common:

pulse code modulation

sigma delta modulation

Quantization bit depth and sample rate are often specified for various audio recording and playback devices as a digital audio rendering format (24-bit / 192 kHz; 16-bit / 48 kHz).

The file format determines the structure and presentation characteristics of the audio data when stored on a PC storage device. To eliminate the redundancy of the audio data, audio codecs are used, with the help of which the audio data is compressed. There are three groups of audio file formats:

uncompressed audio formats like WAV, AIFF

lossless compressed audio formats (APE, FLAC)

lossy compressed audio formats (mp3, ogg)

Modular music file formats are highlighted. Created synthetically or from prerecorded live instrument samples, they are primarily used to create modern electronic music (MOD). Also, this can be attributed to the MIDI format, which is not a sound recording, but at the same time, using a sequencer, it allows you to record and play music using a certain set of commands in the form of text.

Digital audio media formats are used for both mass distribution of sound recordings (CD, SACD) and professional sound recording (DAT, minidisc).

For surround sound systems, sound formats can also be distinguished, which are mainly multichannel sound accompaniments for movies. These systems have complete format families from two major competitors, Digital Theater Systems Inc. – DTS and Dolby Laboratories Inc. – Dolby Digital.

The format is also called the number of channels in multichannel sound systems (5.1; 7.1). This system was originally developed for movie theaters, but has since been expanded for home theater systems.

Why are AV hard drives used in digital recording?

Why are AV hard drives used in digital recording?

AV Hard drives

 

AV HARD DRIVE

The class of AV (audio / video) hard drives means their ability to
read and write streams of data efficiently and smoothly, without pauses. Reserve Army-
some disks ship with a larger internal buffer and are not interrupted
They read / write the process thermal calibration positioning system.
For digital recording systems with insufficient performance and
amounts of RAM to smooth out possible irregularities in the operation of the
discs, AV discs are the only possible output.

Note that the presence of the abbreviation AV in the designation of the disc
it does not mean that it belongs to the Audio / Video class; must be
It must be explicitly mentioned in the passport of the disc.

However, the specified feature is generally necessary only when working
bot with high-quality video information, whose speed
it is approximately 10 megabytes per second per channel. In the case of sound
systems output the rate of a single 16-bit channel stream with a frequency
The 48 kHz sample rate is two orders of magnitude lower and is only 94 kilograms.
bytes per second. At the same time, almost no workstation
to ensure simultaneous operation with hundreds of channels, as well as
the disk cannot process so much data in parallel,
located in different parts of it. In real applications, multichannel
burning disc to disc, most of the overall disc costs
The howling subsystem relies on head movement between recording areas,
and nothing in the data transfer itself. The low speed of sound flows.
kov makes it more convenient and reliable to store them in the computer’s RAM,
disc thermal calibration compensation within 0.5 – 1 s, instead of
use of expensive and rare AV class discs. Also, it is far from
All conventional discs, thermal calibration has a remarkable effect on the
data stream number.

“Broken” data transmission can also occur when using “unintentional”
correct “operating system (DOS, Windows without 32-bit driver
faith on disk, etc.), insufficient number and size of file buffers
get rid of the operating system and the burning program, the use of low-class discs with
transfer rate of the order of 1-2 megabytes per second and lower, incorrect
connect a disc, etc. In any case, these situations are usually
talk about misconfiguration and hardware and software configuration
parts of the system.

What methods are used to compress digital audio effectively?

What methods are used to compress digital audio effectively?

Compress Digital Audio

COMPRESS DIGITAL AUDIO

Currently, the most famous are Audio MPEG, PASC and ATRAC. All of them
use the so-called “perceptual
encoding) in which information is removed from the sound signal,
perceptible to the ear. As a result, despite the change in shape and spectrum
signal, your hearing perception is practically unchanged, and the degree
Compression accounts for the slight reduction in quality. Such encoding
refers to lossy compression methods, when
it is no longer possible to accurately reconstruct the original waveform from the compressed signal
shape.

 

The techniques to eliminate part of the information are based on the characteristics of the human being.
who to listen to, called masking: if there is a high
strong peaks (dominant harmonics) weaker frequency content
hear in the immediate vicinity of them practically no
accepted (masked). When encoding, the entire audio stream is divided
is divided into small squares, each of which becomes a spectral
presentation and is divided into several frequency bands. Within the stripes there are
performs the definition and removal of masked sounds, after which each frame
it undergoes adaptive coding directly in spectral form. All
these operations can significantly reduce (several times) the volume
data while maintaining acceptable quality for most listeners
I read.

Each of the encoding methods described is characterized by a bit rate
the bitrate with which the compressed information should come
on the cable box when the audio signal is restored. Decoder converts
a series of instantaneous signal spectra compressed into a conventional digital waveform
shape.

MPEG Audio – A group of MPEG standardized audio compression methods
(Moving Pictures Experts Group – a group of experts to process motion
images). MPEG audio methods exist in various
types – MPEG-1, MPEG-2, etc .; currently the most common
not MPEG-1 type.

There are three layers of MPEG-1 audio for stereo compression.
your signals:

1 – 1: 4 compression ratio with a data stream of 384 kbps;
2-1: 6..1: 8 at 256..192 kbps;
3 – 1: 10..1: 12 at 128..112 kbps.

The minimum data rate in each layer is defined as 32
kbps; specified bit rates maintain signal quality
roughly at the level of a CD.

All three levels use the input split spectral transformation
changing the frame in 32 frequency bands. The most optimal in relation
data volume and sound quality recognized as level 3 with bit rate
128 kbps and a data density of approximately 1 Mb / min. When compressed from a bottom
at what speeds the forced limiting of the frequency band starts to
15-16 kHz, and channel phase distortions also occur (effects such as
phaser or flanger).

MPEG audio is used in computer sound systems, CD-i / DVD,
CD-ROM “audio”, digital radio / television and other systems
massive sound transmission.

PASC (Precision Adaptive Subband Coding – Precise Adaptive Intraband
coding) – a special case of Audio MPEG-1 Layer 1 with a speed
Stream 384 kbps (1: 4 compression). Used in the DCC system.

ATRAC (Adaptive TRansform Acoustic Coding – acoustic coding
adaptive transformation) is based on stereophonic sound
16-bit quantized format with a 44.1 kHz sample rate.
When compressed, each frame is divided into 52 frequency bands, resulting in
transmission speed: 292 kbps (1: 5 compression). Applied in the system

What interfaces are used for digital audio transmission?

What interfaces are used for digital audio transmission?

Digital Interfaces

S / PDIF (Sony / Phillips Digital Interface Format – digital information format
terface from Sony and Philiрs) – digital interface for home radio
team.

Digital Audio Interfaces

AES / EBU (Society of Audio Engineers / European Broadcasting Union – Society
sound engineers / European Broadcasting Association) – digital engineering
terface for studio radio equipment.

Both interfaces are serial and use the same form
marking mat and coding system: BMC code with automatic synchronization
(Biphasic brand code: code with a double change representation of a unit
phase) and can transmit signals in PCM format of up to 24 bits
at sample rates up to 48 kHz.

Each signal sample is transmitted as a 32-bit word (frame), in which
rum 20 digits are used to transmit the count, and 12 – to form
synchronization preamble, transmission of additional information and
parity bit. 4 bits of the service group can be used to
extension of the sample format to 24 bits.

192 consecutive frames form a block, the beginning of which is marked
special preamble code of the first frame.

In addition to the parity bit, the service part of the word contains a validity bit
(Validity), which must be zero for each valid answer
accounts. If a word is received with a single bit of Validity or with a violation
parity in the word, the receiver interprets the entire sample as wrong and
you can choose to replace it with the old value or interpolate
based on multiple adjacent valid reads. Counts
marked invalid can transmit CD players that
DAT recorders and other devices, yes, when reading information from
the media could not be corrected during read errors
Ki.

The service part of the word also includes the C bits (Channel Status – Status
channel) and U (user bit). Constant price
kidney of each of these bits, taken one at a time from each block frame,
forms a 192-bit word of block service bits, where information is transmitted
information about the title of the work, track number,
device, CD subcodes, etc. S / PDIF transmits
copy protection settings (SCMS).

The standard encoding format is designed to transmit one and two
channel signal, however, when service bits are used to
By encoding the channel number, a multi-channel signal can be transmitted.

On the electrical side, S / PDIF provides a coaxial connection
cable with characteristic impedance of 75 ohms and RCA connectors (“tulle
pan “), signal amplitude – 0.5 V. AES / EBU provides connection
2-wire shielded symmetrical cable with transformer
decoupling via RS-422 interface with signal amplitude 3-10 V, connectors –
Cannon XLR 3-pin. There are also optical options
transceivers: TosLink (plastic fiber) and AT&T Link
(fiberglass).

How sound is encoded

How sound is encoded

How sound is encoded

Sound is a wave that travels more frequently in air, water, or other medium with a continuously changing intensity and frequency.

How sound is encoded

A person can perceive sound waves (air vibrations) with the help of hearing in the form of sound, while distinguishing between volume and pitch.

The higher the intensity of the sound wave, the louder the sound, the higher the frequency of the wave, the higher the pitch of the sound.

We previously wrote in more detail about the human perception of sound, you can read it here.

How audio is encoded (digital encoding and audio processing)
Dependence of the loudness, as well as the tone of the sound on the intensity and frequency of the sound wave.

Hertz (denoted by Hz or Hz) is a unit of measurement for the frequency of periodic processes (eg, oscillations).
1 Hz means an execution of said process in one second: 1 Hz = 1 / s.

If we have 10 Hz, this means that we have ten executions of said process in one second.

The human ear can perceive sound at frequencies ranging from 20 vibrations per second (20 Hertz, low sound) to 20,000 vibrations per second (20 KHz, high sound).

In addition, a person can perceive sound in a wide range of intensities, in which the maximum intensity is 1014 times greater than the minimum (one hundred thousand billion times).

To measure the volume of sound, a special unit of “decibels” (dB) was invented and used.

A decrease or increase in sound volume by 10 dB corresponds to a decrease or increase in sound intensity by 10 times.

Characteristic sound Loudness measured in decibels
Lower limit of human ear sensitivity 0
Leaf whisper ten
Conversation 60
Horn 90
Jet engine 120
Pain threshold 140

Sound volume in decibels

Sync Audio Sampling

In order for computer systems to process sound, a continuous audio signal must be converted to a discrete digital form by time sampling.

For this, a continuous sound wave is divided into separate small time sections, for each section a certain value of sound intensity is set.

Therefore, the continuous dependence of the loudness of the sound at time A (t) is replaced by a discrete sequence of loudness levels. On the graph, this appears to replace a smooth curve with a sequence of “steps.”

How audio is encoded (digital encoding and audio processing)
Sync Audio Sampling

A microphone connected to the sound card is used to record analog audio and convert it to digital format.

The denser the discrete strips are located on the graphic, the better it will be to ultimately recreate the original sound.

The resulting digital sound quality depends on the number of sound volume level measurements per unit time, that is, the sampling frequency.

Audio sample rate is the number of audio volume measurements in one second.

The more measurements that are made in one second (the higher the sampling frequency), the more accurately the “ladder” of the digital audio signal repeats the curve of the analog signal.

Each “step” of the graph is assigned a certain value for the sound volume level. Loudness levels can be thought of as a set of possible N states (gradations), which require a certain amount of I information to encode, which is called audio encoding depth.

Audio encoding depth is the amount of information required to encode the discrete volume levels of digital audio.

If the known encoding depth, the number of digital audio volume levels can be calculated by the general formula N = 2 I.

For example let the audio encoding depth be 16 bit, in this case the number of audio volume levels is:

N = 2I = 2 16 = 65 536.

During the encoding process, each sound volume level is assigned its own 16-bit binary code, the smallest sound level will correspond to the code 0000000000000000, and the highest – 1111111111111111.

Digitized audio quality

Therefore, the higher the sample rate and depth of audio encoding, the better the digitized sound will sound and the better you can bring the digitized sound closer to the original sound.

The lowest quality of digitized sound, corresponding to the quality of telephone communication, is obtained at a sampling rate of 8000 times per second, a sampling rate of 8 bits, and by recording an audio track (“mono” mode).
But it should be remembered that devices that resemble speech synthesizers and speech coders are used to improve this sound in telephony. About speech coders, this article also

Digital audio encoding

Digital audio encoding

Digital audio encoding

To represent the vibrations of sound in digital form, the amplitude of the sound signal is measured at each specific moment of the sound.

DIGITAL AUDIO ENCODING

Since the waveform of sound is inherently continuous, for its accurate digital display it is necessary to measure the amplitude an infinite number of times per second and divide the amplitude scale by an infinite number of gradations. In reality, the number of measurements per second (sample rate) typically ranges from 10,000 to 96,000. Currently, the most common sample rates are 44100 Hz (the standard for CD-audio) and 48000 Hz (the main standard for CD-audio). DAT). The number of amplitude gradations (resolution) is generally taken equal to 28, 216, or 224 (depending on the number of bits allocated for this information).

Of course, distortion is unavoidable when sampling a continuous signal. The lower the sample rate and / or resolution, the closer the output waveform will be to rectangular. In this case, high-frequency distortions arise, which are partially suppressed by filters installed at the DAC output.

Digitized audio requires a large amount of memory. In fact, at a standard 44100 Hz sample rate and 16-bit resolution, the audio material (stereo) for one minute would be 10,584,000 bytes (approximately 10.09 MB). Also, the sound files are very poorly compressed by standard archive programs (zip, arj, etc.). Therefore, there are special compression algorithms for them. For example, a WAV file compressed with ADPCM takes about four times less space. However, distortion may occur. Therefore, it is better not to use audio compression algorithms in professional work.

What is digital audio and how does it work

What is digital audio and how does it work

Digital Audio

Regardless of the path chosen, after connecting the source, the sound from the source will be sent to a microprocessor called a digital audio converter (DAC for short), where there will be 2 stages:

Digital Audio

1) Conversion from analog to digital (a / d);

2) Conversion from digital to analog (d / a).

This processor is sometimes called an ad / da converter. Here, the analog audio signal is processed into digital, then redirected to the central processor and memory, and then to the storage medium. Stored digital recordings (often in .WAV format) are sent back to memory and the CPU, and then converted back to analog by the DAC.

The digital audio / MIDI sequencer allows you to record the sound of synthesizers, guitars, and microphones to files with the .wav extension. No matter how sound is transferred to the computer, it will still go to the DAC, computer memory, and hard drive. The resulting data type is called digital audio data. If you record in “CD quality” (among other things one of the lowest possible), every second of the sound is divided into 44,100 pieces. What is this data? Only numbers. But unlike the MIDI format that encodes the notes played, digital audio data is a digital representation of the actual sound wave. This is the same sound described in numbers. Can you guess that this format takes up thousands of times more space than midi data? This is true.

It is a graphical representation of digital audio data. For a computer, this is a sequence of numbers. With this data, you can perform various operations to change and improve. Outwardly, the signals appear to undergo a series of effects, but in reality what happens is a mathematical process.

How MIDI is converted to sound
You may be wondering how to convert MIDI to audio, is there a “convert” utility for that? Connect the output jacks of your synthesizer to your sound card (or audio interface, or mixer with firewire, etc.) and start recording. Analog waves go through a digital converter (DAC), are converted into numbers, and voila! you will receive digital audio data. The nice thing about a sequencer is that you first record a MIDI track and then refine it. in editors and translate it to digital audio for a perfect recording (well maybe not perfect, there is nothing perfect in the world). Yes; you are using synthesizer software, the process will be called slightly differently, but the gist is the same. The computer creates an audio track based on MIDI data and records it in audio format.

Time to process the resulting files perfectly in sync with plugins or effects. You can also save the finished tracks in MIDI format (then you can edit them at any time) and add the sound of vocals, guitars, or whatever else you want. The sequencer can work simultaneously with MIDI files and digital audio.

Effects types
One of the main and most used effects is VIBRATO.
Distinguish amplitude vibrato, when the amplitude of the signal changes periodically. The frequency of change should be small, from a few fractions of a hertz to 10-12 Hz. Tremolo is a type of amplitude vibrato. The frequency of vibration in the case of a tremolo is not less than 10-12 Hz, and the resulting signal is output in portions.

Frequency vibrato. In a non-electronic way, it was done with electric guitars. By changing the tension of the strings with a special lever, the musician changes the pitch (understand – frequency) and achieves the effect of frequency vibrato. The same can be done with synthesizers and midi keyboards using a special wheel or lever. In music editors, you can also adjust the frequency of the sound, change it within the specified or desired limits.

Ring vibrato. The signal passes through a filter, the settings of which are periodically changed. An interesting and beautiful sound is obtained due to periodic changes in the coloration of the timbre.

Effects: Reverb, Chorus, Flanger, Phaser, Delay: effects based on the delay of the signal.

Reverberation: the effect is created by mixing the main signal with copies lagged for different periods of time, obtained as a result of the reflection of various obstacles (walls, objects, etc.) The number of copies can be infinite, the reflected signal can return to reflected from another obstacle (the delay increases naturally) and again summarized with the main one. With a short delay, the effect results in an immersive and booming sound experience. .

What is digital audio?

What is digital audio?

DIGITAL AUDIO

In fact, there can be several types of “digital sound”, more precisely, the types of its representation on a computer.

Digital Audio

The now familiar “digitized sound” is an analog of a photograph, an exact digital copy of sounds input from outside. It can be a microphone recording of your voice, a copy of audio tracks from a CD, or other sources. Like photography, this sound takes up a lot of space … however, the appetite for photography compared to sound is simply negligible! One minute of digital audio recorded at the highest quality requires approximately 10 megabytes. It is true that there are special compression methods that reduce the volume of computer sound ten times. But more on that later.

Besides “digital”, there is also “synthesized” sound – more precisely, music in MIDI format. Well, you are probably familiar with synthesizers. Briefly, the essence of MIDI technology can be summed up as follows: the computer not only plays the melody you need, but synthesizes it using a sound card. MIDI melodies are just command systems that control a sound card, note codes that it should “display” (indicating instruments, duration and some other parameters of this note). This technology is ideal for computer composers, as it allows you to easily change any parameter of the melody created on the computer: replace instruments, add or remove them, change the tempo and even the style of the song. And files with MIDI music are small, only a few tens of kilobytes. But MIDI has drawbacks too: you can’t record a voice to a MIDI file, and music sounds good only on a very high-quality sound card. Transfer the file you created to a neighbor’s computer equipped with a $ 10 card, and you will long think where all the charm and beauty of the melody has evaporated. It is true that MIDI can be relatively easily converted to digital sound format; reverse conversion, unfortunately, is impossible at the current level of computer technology development.

Finally, there is a third type of sound you can work with at home: “tracker” or “sampler” technology, a kind of love that comes from digital and synthesized sound. When you work with programs of this type, you will “build” a musical composition from small “pieces” of digital or synthesized sound that are repeated periodically: loops or samples. It is on this principle that compositions are created in the current popular style of “house”, “trance”, “techno” …

In short, all simple dance (not to say grosser, primitive), rhythmic music. This type of music, a cross between digital and synthesized, is called “tracker” and has a limited but loyal audience of fans.