Tag: when did digital audio begin

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Digital audio information (Part 1)

The history of recording technology

The creation of sound by computer is a modern stage in the history of the development of sound technology. Let’s take a brief look at this story.

Since the late 19th century, the technical means of storing and transmitting information have developed rapidly. So in the late 1800s, the famous American inventor Thomas Edison made a phonograph.

The principle of operation of the phonograph is as follows. Speech, music, or song create sound vibrations that are transmitted to the recording pen of the phonograph. The needle, acting on the surface of the rotating wax roller, leaves in it a groove with variable depth: a sound track. When a sound is reproduced, the opposite process occurs: the movement of the reading needle along the soundtrack is accompanied by its oscillations with the same frequency. These vibrations are converted by the phonograph into an audible sound. The Edison phonograph is the first sound recording device.

The same idea served as the basis for the production of celluloid gramophone records and mechanisms that reproduce the sound recorded on them: gramophone and gramophone.

In the middle of the 20th century, an electrophone appeared, an electrical analog of a gramophone.
Analog sound representation

The soundtrack of a phonograph record is an example of a continuous form of sound recording.

The electrical signal is transmitted to the speaker of the microphone and converted into sound.

In the 20th century, the tape recorder was invented, a device for recording sound on magnetic tape. It also uses an analog form of audio storage. Only now the soundtrack is not a mechanical “pit groove”, as shown in fig. 1.1, and a line with continuously changing magnetization. With the help of a magnetic reading head, an alternating electrical signal is generated, which is emitted by an acoustic system.

Until recently, all sound transmission technology was analog. This is both telephone communication and radio communication. During a telephone conversation, the sound vibrations from the microphone membrane are converted into an alternating electrical signal that is transmitted through electrical cables. On the receiving phone, they become sound.
Audio encoding and processing

Sound information. Sound is a wave that travels through air, water, or other medium with a continuously varying intensity and frequency.

A person perceives sound waves (air vibrations) with the help of hearing in the form of sound of different volume and pitch. The greater the intensity of the sound wave, the louder the sound, the higher the frequency of the wave, the higher the pitch of the sound.
Dependence of the volume and pitch of the sound on the intensity and frequency of the sound wave.

The human ear perceives sound at a frequency of 20 vibrations per second (low sound) to 20,000 vibrations per second (high sound).

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 “decibel” (dbl) is used (Table 5.1). A decrease or increase in sound volume by 10 dB corresponds to a decrease or increase in sound intensity by 10 times

Sound volume

Sound volume in decibels:
-Lower limit of human ear sensitivity 0
-Rustling leaves 10
-Talk 60
-90 car horn
-120 jet engine
-Pain threshold 140

Sound time sampling. (Part 1)

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

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Parameters that affect the quality of digital audio. (Part 3)

The quality of the digital sound obtained depends on the number of measurements of the sound volume level per unit of 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, if the audio encoding depth is 16-bit, then the number of audio volume levels is:

N = 2 I = 2 16 = 65 536.

During the encoding process, each sound volume level is assigned its own 16-bit binary code, the lowest 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 highest quality of digitized sound, corresponding to the quality of an audio CD, is achieved with a sampling rate of 48,000 times per second, a sampling rate of 16 bits and the recording of two audio tracks (stereo mode) .

It should be remembered that the higher the quality of the digital sound, the greater the volume of information in the audio file.

You can easily estimate the volume of information in a digital stereo sound file with a duration of 1 second with an average sound quality (16 bits, 24,000 measurements per second). To do this, the encoding depth must be multiplied by the number of measurements per second and multiplied by 2 channels (stereo sound):

16 bits × 24,000 × 2 = 768,000 bits = 96,000 bytes = 93.75 KB.

There are three main types of audio digits:

format – no compression;
format (lossy) – lossy compression;
format (lossless): lossless compression.
Lossy compression: technology in which there is a significant reduction of the encoded file compared to the original, due to the removal of information that is not perceived by the human ear.

The downside of this technology is the fact that the compressed file will never be identical to the original.

List of the most common lossy formats:

AAC (.m4a, .mp4, .m4p, .aac): advanced audio encoding (often in MPEG-4 container)
MP2 (MPEG Layer 2)
MP3 (MPEG Layer 3)
MPC (known as Musepack, previously called MPEGplus or MP +)
Ogg Vorbis
WMA (Windows Media Audio)

Lossless – Lossless compressed audio formats, including:

FLAC (Free Lossless Audio Codec)
APE (mono audio)
WV (WavPack)
These formats are capable of converting CD to digital format while maintaining quality. As an example, you can take a CD, convert it to WAV, then WAV to FLAC, then go back from FLAC to WAV, and then burn it to a blank CD and you have an absolutely identical copy of your source.

What format does the music sound with the best quality?
The most popular is the lossless FLAC format, and one of the most widely used CD to FLAC conversion programs is EAC (Exact Audio Copy).

Of all the parameters of digital audio, it is necessary to pay attention primarily to the following indicators:

sampling rate (precision of digitizing an analog signal in time),
bit rate (amount of information contained in a file in terms of one second).

The sample rate is the frequency at which digital audio is processed. The most common sample rate for quality audio formats is 44.1 kHz.

It is generally accepted that a high bit rate guarantees the best quality; this is true, but only if the source file is of good quality. A high quality MP3 should have a bit rate of 320 kbps, but a high quality FLAC format generally has a bit rate of 900 kbps and higher.

What is the best quality music format?
In addition to the audio formats themselves, for high-quality music sound, high-quality reproduction equipment is also needed: speakers, amplifiers, headphones.

Parameters that affect the quality of digital audio. (Part 2)

The format is also called the number of channels in multichannel sound systems (5.1; 7.1). Initially such a system was developed for cinemas but later spread by Software Codec

 

Software-level audio codec

§ G.723.1 – one of the basic codecs for IP telephony applications

§ G.729 – proprietary narrowband codec used for digital representation of speech

§ Internet Low Bit Rate Codec (iLBC) – a popular free codec for IP telephony (in particular for Skype and Google Talk)

Audio Codec (Audio Codec; Audio Encoder / Decoder) – A computer program or hardware designed to encode or decode audio data.

Software codec

A software-level audio codec is a specialized computer program, a codec that compresses (compresses) or decompresses (decompresses) digital audio data according to an audio file format or streaming audio format. The task of an audio codec as a compressor is to provide an audio signal with a certain quality / precision and the smallest possible size. Compression reduces the amount of space required to store audio data, and it is also possible to reduce the bandwidth of the channel through which the audio data is transmitted. Most audio codecs are implemented as software libraries that interact with one or more audio players such as QuickTime Player, XMMS, Winamp, VLC media player, MPlayer, or Windows Media Player.

Popular software audio codecs by application:

§ MPEG-1 Layer III (MP3) is a proprietary audio recording codec (music, audiobooks, etc.) for computer equipment and digital players

§ Ogg Vorbis (OGG) – the second most popular format, widely used in computer games and file-sharing networks to transfer music

§ GSM-FR is the first digital voice coding standard used in GSM phones

Adaptive Multispeed (AMR): human voice recording on mobile phones and other mobile devices

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.

Sound volume in decibels

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.”

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 fringes are located on the graph, the better it is ultimately possible to recreate the original sound.

Parameters that affect the quality of digital audio. (Part 1)

The best music formats for sound quality Minimum and maximum sound quality

The main parameters that affect the quality of digital audio recording are:

§ The capacity of the ADC and DAC.

§ Sampling frequency of ADC and DAC.

§ Jitter ADC and DAC

§ Resampling

In addition, the parameters of the analog path of digital sound recording and playback devices are still important:

§ Signal to noise ratio

§ Total harmonic distortion

§ Intermodulation distortion

§ Inequality of the amplitude-frequency response

Channel interpenetration

§ Dynamic range

Digital sound recording techniques

Digital sound recording is currently done in recording studios, under the control of high-quality, expensive personal computers and other equipment. In addition, the concept of “home studio” is quite developed, in which professional and semi-professional recording equipment is used, allowing you to create high-quality recordings at home.

Sound cards are used as part of the computers that perform processing on your ADCs and DACs; Most of the time at 24 bit and 96 kHz, a further increase in bit rate and sample rate hardly increases the recording quality.

There is a whole class of computer programs: sound editors that allow you to work with sound:

Record incoming audio stream

§ create (generate) sound

§ modify an existing recording (add samples, change timbre, speed of sound, cut parts, etc.)

§ rewrite from one format to another

Convert convert different audio codecs

Some simple programs only allow converting formats and codecs.

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 audio formats (APE, FLAC)

Lossy compression 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 multi-channel 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.

Why is today’s audio and video called digital?

The current heyday of mobile devices, computers, and the World Wide Web is called the digital age. This is due to the rise of digital information: everything we read and write, see and hear, is translated into a form that a computer can understand. The computer, in turn, opens up a whole universe of possibilities for working with such information: it is becoming easier to copy, transfer and store melons. MTS / Media will help you understand the theory of creating a digital world.

Binary and decimal number systems

Before understanding how an image obtained by the lens of a smartphone, or a book, or is converted into a file on a computer, you must understand at least a first approximation of how this same computer works.

At the most basic level, the computer, despite the buzz attributed to intelligence, operates with absolutely primitive categories: yes, no, no, no. In the jungle of microcircuits, this dualism is expressed in the presence or absence of an electrical signal. Everything that a computer has to digest must first be “chewed”, decomposed into simple elements, reduced to a set of two opposing concepts.

“No” in computer language replaces the number 0, “yes” – 1. That is why computer information is called digital. Everything your computer or smartphone stores, all the complex algorithms built into the most complex programs, and a masterpiece frame from the last party, and your favorite song, and an unfinished letter to your boss with the title “let’s go. … “, this is all just a long string of zeros and ones.

The base number in our daily life is 10; We use numbers from 0 to 9, that is, the decimal number system is familiar to us. In the world of computers, the base number is 2 (just two digits, 0 and 1), and the number system is called binary or binary. In the decimal system, to go from single-digit numbers to two-digit numbers, you must first count to nine, to go to three-digit numbers, up to 99. The principle of digit formation in systems is the same: appears a new digit in a number after all available digits in the current one have been used up.

Now we understand how any number can be converted to a digital form, understandable to a computer. Also, we can see what it is, the minimum information is 1 or 0. This minimum piece is called a bit. To write the number 2 in the binary system, you need 2 bits of information (10), to write the number 4 – 3 bits (100), for 15 – 4 bits (1111).

Letters in numbers

In fact, most of the time we are not dealing with bits of information, but with bytes. A byte has 8 bits. If you see that we are talking about the amount of information, say 10 MB, then the letter “B” is exactly one byte, not one bit. In cases where bits are indicated, the word “bit” is written in its entirety.

A byte is an analog of a word in machine language. At the dawn of the computer age, 8 bits of information corresponded to a memory cell of machines, the 8 bits were transmitted together as a whole. Then the “words” from the machine started to get longer, but they were still multiples of eight times the number of bits.

Why exactly 8? It happened like that. 8 bits were needed to represent 1 character of text in one of the earliest computer encodings. An encoding is a table of correspondence between text characters and binary numbers. If you try to type all variants of eight-digit numbers consisting of zeros and ones, from 00000000 to 11111111, there will be 256 such variants, that is, how many characters are in many existing encodings, and they are all called 8-bit.

A coding table is a kind of instruction for a computer with which it translates the letters of a text into binary numbers and vice versa. However, not all characters from all languages ​​fit in one encoding, and each language needs its own instructions. For this reason, national encodings have become widespread in the world. So, in the Cyrillic encodings (KOI-8, Windows-1251, MacCyrillic) there are large and small letters of the Russian and Latin alphabets, numbers, punctuation marks and auxiliary symbols. If support for Cyrillic encoding is not installed on a computer somewhere in China, you will not be able to type Russian characters and the operating system will not be able to display them.

Later, along with 8-bit encodings, 16-bit encodings became widespread, in which almost every imaginable character of every language can be found. However, each letter in this encoding already has two bytes.

So 1 letter is 1 or 2 bytes.

Things You Should Know About Digital Music Quality (Part 2)

4. The search for an ideal is harmful.
Each of us wants the world and its components to be ideal; this is the axiom. Any DJ wants to have speakers in clubs connected and in tune, every track in the collection shines with quality mastering, and so on. But only the results of the work done are taken into account, each of us is forced to make commitments every day.

So interestingly, this also applies to the quality of the music. We already noted at the beginning that this is an important point, but not enough to deny the space of options and the possibilities of making decisions, perfectionism is completely out of place here. For example, an original underground producer puts out a new track at 128 KBPS, and it will definitely break the crowd. A dilemma arises: to play it or not?
Purists will answer negatively. But you have to be honest with yourself and judge by the emotions you want to convey through music. If the cumulative mass of factors exceeds five minutes of not-so-high-quality sound on your computer, the doubt can be dismissed. Don’t let dogma and the false pursuit of perfection damage your mission as an artist. You can buy the version in the best quality later. For now, do your thing.

5.Music is created with the playing environment in mind.
Good sound producers listen to the tracks as if they are making them in every possible system: in earplugs, cheap plastic speakers for a computer, etc., with the idea of ​​how other people will eventually hear it.
This brings us back to the first point: the work of the producer and the mastering engineer decides much more than the minor aspects. Club tracks with abundant bass sub-registers sound bad on the radio, and loud, howling radio mixes with tight dynamic range sound bad in the club. And the file format is irrelevant here. Producers are forced to compromise – this is an integral part of their workflow, and no expensive equipment or ghost software can affect this like you can.

6. The “golden age of audio” is fiction.
People ooze feelings or chant mantras too often, as was good in the past. That, in general, does not stand up to criticism: Stereo as such did not exist until the late 1960s, and the golden age of declining pop music gave rise to formats as unhealthy as eight-track cassettes.
Amplifiers and monitors have changed dramatically for the better, keeping pace with advances in technology. Yes, in the 70s and 80s it was possible to achieve good sound from high quality printed records, but in proportion to them there were many terrible circulations and publications that just sounded disgusting, ask older DJs and music lovers.

7. Technology comes first.
Thanks to technological progress, we can listen to as much music as ever, good or bad, until we can tell. The most suitable music fans are happy to listen to a variety of genres and styles in different formats on different devices and have fun. Because the main thing is the music, if it is good in itself, you can abstract from background noise and interference from shortwave radio, a joke club stereo system, and excessive volume.
So, gentlemen, intellectual audiophiles and expensive equipment manufacturers, we perfectly feel the difference. A hamburger eaten at the race on Wednesday does not prohibit a gourmet restaurant on Saturday. Everything must have its place and its time
Wireless audio systems, streaming, portable players … all have contributed to making music available to more people than ever. But even such a positive dynamic meets fierce resistance from fans of luxury sound at any price and sacrifice.
You have the option to choose between two completely contradictory situations. In the first, you find yourself in a sound-dampened listening room, where a stereo system is playing for thousands of dollars, and your friends stroking their beards and curling their mustaches, praising the “delicious” sound of the hi-hat and noticing “Texture” of the percussion nuances in the bass player’s performance. And in the second, you’re tearing up a crowded little bar, playing your set on lousy gear at full volume, where the girls start turning the tables because you’ve just started a crazy 128 kilobits-per-second remix.

Things to know about the quality of digital music

One of the key aspects of a positive music experience is the quality of the recordings and the quality of the sound that we enjoy. This is a very speculative topic, clashing technologies, devices and, first of all, the listeners themselves. The mass of the common people oppose audiophiles of all kinds with views of varying degrees of radicalism, but with an equally high level of rejection of the habits of their opponents.

This crowd of connoisseurs of $ 500 cables, tube amps, and high-end stereos are joined by respected artists and producers who explain that music should sound great, that it sounded like that at the time of recording, but with the advent of digital technology (so there is a mastery of audio file compression and the general portability of playback devices), the quality of music inevitably deteriorates, and generally we need to do something about it. Stop the loudness race or buy expensive CDs, get a player, amplifier and speakers, for example, at a decent price.
They think we are fools that we buy MP3s from online retailers like iTunes. Who listen to satellite radio and the internet. Who get fresh music every day on popular digital audio platforms. Who are happy with DJ sets playing from flash drives.
But these “nuances” not only prevent us from listening to a large amount of music using the above methods, but also from enjoying it.

Without a doubt, the quality of the music plays an important role. For example, DJs know this very well, working with musical material much closer and closer than the public. There is a difference between a specially compressed MP3 file and its source on a CD; it is a fact. However, the authoritarian tone of audiophiles and high-end music equipment manufacturers should soften, and the rhetoric should become more mundane and closer to the average consumer of music products.

We decided to collect 7 data on sound quality that will dissipate the clouds a bit over digital formats and portable audio.

1. The file format is not critical.
What the producer of a track does with it in the studio is a thousand times more important than in what format the result of this work will be encoded. You can’t make candy out of shit – a decent track with an artistic message, properly produced, mixed, and mastered in an acceptable dynamic range (where you didn’t go overboard with compression in the first place), even on unimportant speakers, will sound better than a dull, gray and poorly mastered track. even if you hear it in lossless format on a stylish stereo system. Always. This should be obvious to everyone.

2. Compressing the file size by 80% does not reduce the audio quality proportionally.
When you compress digital audio, you get rid of the main ballast without affecting the quality of music the human ear can hear. This process is called lossless compression (very similar to RAR or ZIP files). If you want to reduce the size of the audio file even more radically, you will have to shred the source and its sound forever; this is already a case of the notorious “quality loss”. Yes, as a result, the track undergoes irreparable changes, but people too often create darkness, claiming that this happens indiscriminately.
It’s time to admit that most people can’t hear some of the details on the album. It’s just that our ears are not comparable to the hearing of a dog and other animals. You can get rid of a lot of secondary information in the audio and no one will know the difference. This is psychoacoustics in action, this is how lossy audio compression works. There is a certain threshold below which the difference begins to be heard (MP3 with a bitrate of 96 kilobits per second cannot be compared with an analog of 320), but this does not mean that the myth about the relationship between the percentage of compression and the end result is true. It is a myth.

3. People make the most of life when music isn’t of the best quality.
Life story. In the 90s, the article’s conditional hero came to an illegal rave, spent the whole night, and decided he would make DJing the profession of his life. A brave step and a fateful decision. But what happened to the sound at that party? Everything was wrong, remember. The needle flew, the EQ not tuned, and the amps periodically cut out. Has anyone fired on this? Barely.
Have you ever been to a nasty sounding party that changed your life? Danced all night by bad announcers in a strange club and left in the morning with your future life partner?

The benefits of digital audio

The basics of “numbers”

Each of the multimedia devices on sale today, be it a CD player, a voice recorder or a flash memory player, uses many different types of presentation of data streams, which are then converted into sound. And even more sound formats used for professional purposes have been invented. An inexperienced buyer is forced to gather information about designations on boxes and devices from a variety of sources, often receiving incorrect information or even more confusion.

Almost all devices in the “Portable Audio” section of the ZOOM.CNews.ru catalog support multiple sound formats at the same time, and many devices that do not belong in this category are also tagged with support for playing sound files. To help our reader, we decided to create a short glossary of abbreviations and talk about the most common formats. We plan to leave it open for updates and modifications, adding new formats and describing in more detail the advantages and disadvantages of the already common or forgotten ones.

A little theory

To begin with, remember that digital sound is nothing more than a collection of numbers. The determining factor is the system by which sound as air pressure is converted into data streams and encoded for further processing and reproduction. Consequently, digital sound is usually included in computer files with various extensions, which more often (but not always) can determine their format. And the same concept of format can have, paradoxically, two meanings. First, the format may exist as a general characteristic, including both the type and the physical characteristics of the medium (disk or cassette), recording method, encoding principles, and protection against errors. Second, the format can only be understood as the method of encoding and compressing sound, as standard media are used for transfer, for example a computer.

Analog sound, unlike digital, is reproduced on analog devices and has a number of significant differences. While not a data stream, analog audio is represented as a continuous electrical signal that represents a change in the sound wave. To translate it into digital format, the sound is “digitized”, that is, it is divided into certain segments, in which the numerical value of the amplitude is fixed at that moment. We will not delve into the principles of digital sound creation, but it is absolutely necessary to note that the more often a sound segment is divided and its characteristics described, the clearer and more complete the sound image itself is created.

This process generates an enormous flow of data that describes the sound, and it is clear that each digital audio format is nothing more than a compromise between the need to present the sound as loud as possible and the limitations of the memory of the computer or device. Of reproduction.

A little more theory. In most cases, the human ear perceives sound with a frequency no higher than 22,000 Hz and, to describe it fully in digital form, a sampling frequency of at least 44.1 kHz is required. Since it is absolutely impossible to determine the value of the signal at any given moment, during digitization quantization occurs, that is, the replacement of the actual values ​​of the signal by approximate values. The more levels of audio quantization, the more accurately the signal level is described. As a result, each standard CD carries an audio signal with a sampling frequency of the same 44.1 kHz and a 16-bit quantization level,

The benefits of digital audio

And now, briefly on why this algorithm was developed. Digital sound has enormous advantages over analog, although we must not forget its certain disadvantages. The main value of digital sound is the possibility of infinitely long storage and endless playback of material without losing the original quality, while analog sound loses quality with each re-recording. In addition, the transmission of sound and its processing by modern digital means is facilitated, first of all, by specialized computers. Furthermore, the digital signal on transmission lines is more resistant to interference than the analog signal. It is also important that digital technology,

Historically, digital sound was undoubtedly the initiative of company engineers who adopted Philips-developed Audio-CDs, also called CDDA – Compact Disk Digital Audio.

Digital audio encoding

In fact, this or that 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.

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” (for example, in the form of one of the PCM variations discussed above) 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. Thus, 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 depth 16-bit quantization bits. One hour of music in this format takes approximately 600 MB (60 minutes * 60 seconds * 2 channels * 44100 samples per second * 2 bytes per sample = approximately 605 MB). Considering that, for example, an ordinary music lover’s music collection may have 5000 tracks with an average length of about 3 minutes each, the amount of memory required to store it in its original digital form turns out to be very impressive. . 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 stream (the term “data Original “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 form of data compaction. 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 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. There, where only intelligibility is important in sound (for example, in telephony, where the presence of frequencies above 4 kHz is not necessary), the audio information in the encoding process is seriously “simplified”.