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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.

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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”.

What are Lossless, Lossyless, cue and WAV music formats?

To make it easier to handle bitrates, I’ll give a somewhat simplified understanding of Lossy and Lossless bitrates.

If we imagine the sound in the form of a broken diagram, then in MP3 and OGG formats (these are currently the main Lossy formats, we will not consider the rest here as they are quite rare) from 128 to 256 kbps the ends of the sound are cut off (from this diagram). As for the 320kbps bit rate, the sound is not cut off.

What are bit rates?
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.

Lossless – Lossless, which means that lossless (lossless) audio formats such as FLAC, APE, and WAV, as well as lesser-known ones, convert CDs to digital without loss of quality, that is, you can take a disc from your collection, save it to WAV, re-encode WAV, say to FLAC (or APE), then from FLAC (or APE) to WAV and burn it to disc and you get a disc absolutely identical to your CD. This begs the question: why not just use the WAV format? It’s very simple: lossless formats have the same quality as WAV, but take up less space, this is their advantage. There is a myth that an analog of a CD is an MP3 with a 320 kbps bit rate, but this is not the case, only a lossless image of this CD is an analog of a CD, by the way, and vinyl does not it has analogs at all. The bitrate of the vinyl analog must be equal to infinity, since vinyl records are made from so-called master tapes. A master tape is an analog copy of a piece mixed in a studio.

What are WAV and APE?
It is a lossless compression algorithm for WAV audio files, commonly used to store music extracted from compact discs (CD-DA). First, the original WAV file is removed from the CD-Audio (if a standard disc is fully recorded with music for 80 minutes, then the file will be 700Mb), and then it is archived in APE (standard extension for files compressed by Monkey’s Audio) . Yes, this is comparable to archiving, since APE can later be decompressed and the original WAV obtained, as if it were archived with ZIP or RAR. Compress the APE of the original WAV normally 1.5-2 times.

APE is a format for music connoisseurs, who are often interested in entire albums, not individual compositions. Music databases like freedb also work with albums. Also, a compressed album with one file takes up slightly less space than if each song were separated. But in fact, nobody forbids storing music in APE per track.

Many people don’t like APE because they need to spend more time on it to load it to the site (or from the site) or to the disk grabber. They argue that the size is large and it only causes a lot of problems with APE. The size of APE can be 2 or 4 times larger (depending on the type of music) than MP3. But, for the sound quality you have to pay (and not very, in my opinion, a great price). The extra half hour of horse racing or graberra is well worth it.

The APE bit rate ranges from 700 kb / ps to 1000 and more.

What is FLAC?
FLAC stands for Free Lossless Audio Codec (free lossless audio codec). FLAC is free, open source, and cross-platform. The compression ratios of FLAC are slightly lower than those of Monkey’s Audio, while the encoding (compression) time in FLAC format is approximately the same as that of Monkey’s Audio, however, the decoding (decompression) is much faster. FLAC is very popular on the Oslo network due to its cross-platform nature: it can be used on Windows, Linux, Unix and Mac OS X. There are also portable media players that support playing FLAC files. The Windows version of the codec contains plugins for Winamp (version 2.x / 5.x),

MP3
MP3 is a lossy compression format, that is, lossy. It is based on the assumption that the human ear simply does not perceive some frequencies and consequently they are removed during the compression process, which can significantly reduce the volume occupied by the composition.

The only advantage of MP3 is the size and nothing else. The fact is that when digitizing (encoding, compressing) a musical composition in MP3, frequencies that, according to some experts, cannot be heard by the human ear are discarded, so we obtain a small size (around 70% less than the source, depending on the quality of the bitrate and the codec).

All Digital Audio Formats

ACC
Advanced audio coding
The format is a further development of the MP3 format.
ALAC
Apple Lossless Audio Codec
Apple Lossless (also known as Apple Lossless Encoder, ALE or Apple Lossless Audio Codec, ALAC) is an audio codec developed by Apple Inc for lossless compression of digital music.
ALS
MPEG-4 audio lossless encoding
MPEG-4 ALS is an efficient and fast codec for a variety of applications.
AMR
Adaptive multiple rate
The AMR compression format was developed specifically for use in cellular systems. Its field of application is voice audio content compression.
MONKEY
Monkey Audio
Monkey’s Audio (Windows only) is considered one of the best lossless audio codecs for storing music due to its effective ratio of output file size to speed.
ATTRAC
Adaptive Transformation Acoustic Coding
ATRAC is a lossy compression system based on psychoacoustic principles. Compresses an audio CD to approximately 1/5 of the original with a slight loss in sound quality.
Asao
Nellymoser audio codec
Nellymoser Asao is a proprietary codec that was designed for low bit rates.
CELTIC
Overlapping energy restricted transformation
The CELT codec is an algorithm for compressing audio data. Like MP3, Vorbis and AAC, it is suitable for high quality music streaming. Unlike these formats, CELT also has a very low latency, lower even than Speex, GSM or G.729.
Dolby
Dolby has developed many audio sound formats. Among them are compression formats.
FLAC
Free Lossless Audio Codec
FLAC is possibly the most popular lossless audio compression format.
LossyWAV
LossyWAV is a free lossy compression format. But, in essence, it is a preprocessor for PCM audio stored in WAV containers.
MP1
MPEG-1/2 Audio Layer I
MPEG-1 Audio Layer I (abbreviated as MP1) is one of the three formats included in the MPEG-1 standard. Even though it is compatible with many media players, the codec is already very outdated and has been superseded by the MP2 and MP3 codecs.
MP2
MPEG-1/2 Audio Layer II
MP2 is still used in the broadcasting industry for satellite transmission of digital video transmission and digital audio transmission.
MP3
Audio Layer III MPEG-1/2
The format is sometimes confused with MPEG-3, but MP3 is designed to compress only audio information and the full name sounds like MPEG Audio Layer-3.
Surround sound MP3
In 2004, Fraunhofer IIS released a backward compatible extension for MP3. MP3 Surround files provide high quality 5.1 sound with new decoders.
MP4
MPEG-4 Part 14
These are file extensions for the MPEG-4 container format, which can include all types of media (video, natural and synthetic audio, 2D and 3D graphics, animated avatars, etc.).
MPC
Musepack
Musepack is a lossy compression scheme invented by German programmer Andree Buschmann.
MT9
A new multi-track waveform data storage format that claims to be MP3.
Ogg Vorbis Audio
The Ogg vorbis format was developed by Xiphophorus. On the same site you can find the source codes of the project. It is part of the Ogg project to create a completely open multimedia system.
OptimFROG
OptimFROG is a lossless compression algorithm whose main goal is to reduce the size of audio files as much as possible. This is somewhat similar to ZIP compression, but is highly specialized for audio data.
Opus
Opus is a highly versatile, royalty-free, open source audio codec.
RealMedia
RealMedia is a proprietary streaming and multimedia file format owned by RealNetworks products and services.
SND
Sound
SND (SouND) is a digital audio file format created by Apple.
Speex
Speex is a patent-free audio compression format developed for voice transmission, as well as for use in open source software (for example, VoIP).
TAK
Tom’s lossless Audio Kompressor
TAK is lossless audio compression that provides APE efficiency and FLAC decoding speed.
VQF
TwinVQ
A proprietary format that was created to replace MP3, but was never fully developed due to its proprietary nature.
Wav
Wave audio file format
The WAV format is perhaps the most common audio storage format. It is the easiest to use to process and is compatible with almost all audio players.
WMA
Windows Media Audio
WMA is a compression format developed by Microsoft.
WavPack
WavPack is a completely open, lossless, high quality, lossy audio compression format with a unique hybrid mode.

Digital audio from A to Z

Confused about the terms used to describe audio devices? We have created a quick guide to help you discover them.

Do you want to immerse yourself in the wonderful (and sometimes overwhelming) world of high definition audio? You have a lot to learn about this world, but the endless abbreviations and terms can be confusing, making the text look like a collection of words.

There is nothing to worry about. At Sony, we make sure you get all the Hi-Res Audio knowledge you need, become a true expert, understand the complexities of terminology, and enjoy the best sound with the best music.

Below is a list of the main terms used by hardcore audiophiles when discussing Hi-Res Audio technology, as well as their definitions.

Hi-Res Audio / Hi-Res Audio

Hi-Res Audio generally means digital recordings with a higher sample rate than audio CDs and the MP3 format. This technology offers much higher sound quality while retaining more data than converting the original studio recording to MP3 files. Some of the high resolution audio formats are WAV, DSD, ALAC, FLAC, and AIFF.

DSD and PCM

What is the difference? There are two main ways to process / encode audio in digital formats: PCM and DSD. In short, editing is easier with PCM. However, the DSD file format is used in recording studios and this digital format is believed to be as close as possible to the original analog source. Below is a more detailed description of each format:

DSD

Direct Stream Digital is a digital recording method in which the audio signal is encoded using pulse density modulation like digital media. The sample rate of this audio format is 2.8224 MHz or 5.6448 MHz, which is 64-128 higher than the sample rate of audio CDs.

PCM

Pulse Code Modulation (PCM) is the basis for digital audio recording whereby the standard analog audio signal is converted to digital. This is the standard form of digital sound on computers and CDs. The analog signal is sampled at regular intervals and its amplitude is recorded as a point on a digital scale.

With data loss

The lossy format removes some of the information from the original digital recording in an attempt to preserve the quality of the original sound as much as possible when played back. This is the case for MP3 and AAC audio formats. The compressed file takes up much less space than the original file, but the quality suffers.

No data loss

The lossless encoding format allows you to store digital audio without losing the original data or allows you to reconstruct it when played back. Lossless audio files are generally larger than lossless files. However, it achieves significantly better sound quality. Examples of audio recordings of this type are files with the extensions FLAC and Apple Lossless.

No compression

The definition of the concept is derived from the name: uncompressed raw data. In general, uncompressed audio files like WAV and AIFF are of the best quality. The downsides of uncompressed audio are that they take up a lot of space and require a lot of bandwidth to open and play.

kHz / bit

This is a standard notation for the relationship between sample rate and bit depth.

Number of kilohertz (kHz)

It is a unit of sampling frequency which is the number of times the audio signal is quantized per second. Therefore, the higher the kHz number, the better the sound quality.

Bit depth

The bit depth of a digital recording determines how many bits (that is, data) are used to store each sample of the analog signal. Bit depth is directly related to the resolution of each sample. The higher the bit depth, the better the sound quality.

Now that you understand the complexities of Hi-Res Audio terminology, try to find examples for each concept.

What to expect from digital audio

A few years ago, the word “multimedia” entered the computer lexicon, and more recently, the PC is increasingly used as a home entertainment center. In both cases, the computer must reproduce the sound, which, as you might guess, exists on it only in digital form. And if with the advent of the first transistor technology, the phenomenon of “transistor sound” was vigorously discussed and covered with myths and legends; However, it is often believed that computer signal processing, on the other hand, is obviously better. So what is digital audio and how is it inferior to or superior to analog?

From a human point of view, sound is air vibrations with a frequency of approximately 16 Hz to 20 kHz. A person perceives the lower frequencies (with sufficient amplitude) not as sound, but as vibration. Superiors are not captured at all. The upper limit of the frequency range depends on age: in young children it reaches 22-24 kHz, and gradually decreases to 8-12 kHz over time. Therefore, the human ear can hear signals of a very wide bandwidth. For comparison: the eye can perceive color only in the range that covers the change in frequency of electromagnetic oscillations by less than 2 times. Of course, not all frequencies are equally important. For example, a range of 500 to 3500 Hz is sufficient for speech intelligibility. But to listen to music or the soundtrack of a movie, this is not enough. Ideally, the sound field in the listening area should be indistinguishable from the sound field in the recording area. That is, the entire audio path, from a studio microphone to a home speaker, must not introduce distortions that are within the resolution of the human auditory analyzer.

The sound that our ears perceive when playing a digital recording has previously undergone a series of transformations:

1) electromechanical conversion of air vibrations into an electrical signal;

2) amplification and processing of an analog electrical signal (frequency equalization, addition of reverb, etc.), mixing;

3) analog to digital conversion;

4) digital signal processing: frequency correction, mixing, mastering, etc .;

5) storage or transmission of digitized sound;

6) digital signal processing: frequency correction, volume control, oversampling;

7) digital to analog conversion;

8) Analog signal processing (frequency equalization, mixing, adding reverb, etc.);

9) amplification of the analog signal;

10) electromechanical transformation of electrical current oscillations into sound oscillations.

When processing an analog signal in a studio, devices with an analog interface and digital “fill” are often used, so the chain of analog-to-digital and digital-to-analog conversions can be much longer.

The first four stages are most often carried out on studio equipment, which has incomparably higher performance than home equipment. Therefore, although the distortions are unavoidable, we will assume that they are insignificant compared to the distortions of a similar nature introduced by the household equipment in the last five stages. In amateur audio recording, additional distortion should be considered in the early stages, which will be described below.

Electromechanical conversion is usually done with a studio microphone. This device generates a very weak signal that needs amplification and is also extremely susceptible to mechanical stress. Even under ideal conditions, for example in a concert hall, acoustic noise can cause the dynamic range of the music being played to be less than the maximum dynamic range of a 16-bit sound presentation.

A signal recorded from several microphones is inevitably processed: the required volume levels of the different channels are selected, the noise is cut with filters, etc. Also, the dynamic range of the signal is generally compressed. The last operation leads to a significant increase in the noise level, but without it, the recording would sound unsatisfactory on middle-class consumer equipment, first of all, too quiet.

The distortions introduced by the sound path have a varied physical nature and very different manifestations, but nevertheless they can be divided into three large groups.

Digital sound. Digital audio encoding

What determines the quality of an audio signal?

The purity and timbre of the sound are mainly determined by the audio codec, or rather, by its bit depth and sample rate (the higher they are, the better the sound). This processing can be done in hardware with a special chip, an audio processor, or in software that uses controllers, which consumes CPU resources.

What is AC’97, HDA?
AC’97 and HDA (High Definition Audio) are Intel’s proposed standards for audio codecs. AC’97 was introduced in 1997 and then improved several times, but eventually became obsolete and is now replaced by HDA. HDA is fully AC’97 compliant with improved performance and enhanced capabilities.
What is the difference between AC’97 and HDA?
AC’97 defines the maximum bit depth of a 16-bit audio codec at a sampling rate of 48 kHz, HDA – 32-bit / 192 kHz. Additionally, HDA devices support 8-channel (7.1) audio, DVD-Audio, Dolby surround sound technologies, and other advanced features.
What is the sample rate and bit depth of the codec?
Sampling is the acquisition of instantaneous values ​​(samples) of an analog signal with a certain time step in the digitization process. The frequency of this step is called the sample rate (it is also the sample or sample rate). The larger it is, the better the sound recorded and reproduced. In studio equipment, the frequency is 48 kHz, in home systems – 44.1 kHz.
Bit depth determines the quality of the recorded audio. Higher is better. The bit value, for example 32, denotes the number of bits that are allocated to record the amplitude of the signal at the time of its measurement.
Consequently, the more often (sample rate) and more accurately (bit depth) the audio signal is measured, the higher quality audio file is obtained.
What is the signal-to-noise ratio?
The ratio of the pure audio signal to the noise generated by the device itself. The higher the value (in dB), the better. The Sound Blaster X-Fi sound card has a signal-to-noise ratio of 118 dB. Most audio codecs are 80-95 dB.
What is DAC and ADC?

The DAC (digital to analog converter) and ADC (analog to digital converter) are part of the codec and directly perform sampling: during playback, the DAC converts the digital code to an analog signal, while recording, the ADC performs the reverse conversion. The better the ADC, the clearer and more detailed the sound that will flow from the speakers. The better the DAC, the more accurately the analog signal will be converted to digital.
Codecs for multi-channel audio support include various DACs and ADCs.

What is the bit rate?
The bit rate (literally, the information bit rate) determines the maximum amount of information that can be transmitted through the audio channel per unit of time. A high bit rate is needed to transmit a rich sound image and is not required when encoding speech. Audio recordings with a 128 Kbps bit rate are suitable for inexpensive speakers, but when accessing expensive equipment, it makes sense to get music at a 192-256 Kbps bit rate.
Convenient solution: variable bit rate encoding, change the bandwidth of the audio channel according to the quality and saturation of the musical fragment.