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Audio. Digital and Analog Audio Part 6


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Audio. Digital and Analog Audio Part 6

Digital Audio

ANALOG AUDIO PROCESSING

digital audio

Any processing of an analog audio signal is accompanied by a certain loss of its quality (frequency, phase, non-linear distortions occur), but it is necessary. The main types of processing are as follows:

amplification of the signal to the level required for transmission, recording or playback through the speaker: having sent the signal from the microphone to the speaker, we will not hear anything: it is necessary to pre-amplify it in terms of level and power, while providing the ability to adjust the volume.

frequency filtering: infrasound, which is harmful to health at certain frequencies, and ultrasounds are cut off from the useful sound range (20 Hz – 20 kHz). In many cases, the range is deliberately reduced (the voice phone channel has a band from 300 Hz to 3400 Hz, the frequency band of metered radio stations is significantly limited). For loudspeaker systems, which usually have 2-3 bands, separation is also necessary, which is usually carried out in the crossover filters already at the level of the amplified (powerful) signal.

frequency correction (equalization): tone control, compensation for uneven recoil due to acoustic properties of the room, compensation for losses in transmission lines, studio processing to achieve the desired “color” of sound, suppression of feedback parasitic acoustics (“whistle”), etc., etc.

Noise suppression: there are special dynamic noise reduction schemes that analyze the signal and reduce the bandwidth in proportion to the level and frequency of the RF components (“denoisers”, “dehissers”). In this case, the noise that is above the bandwidth of the signal is cut off and the remaining noise is more or less masked by the signal itself. Such schemes always lead to a very noticeable degradation of the signal, but in some cases their use is appropriate (for example, when working with a recorded speech or on intercom radio stations). For analog sound recording equipment, compressor / expander-based noise cancellers (“compander” eg Dolby B, dbx systems) are also used, the work of which is less perceptible to the ear.
Impact on dynamic range: In order to make the playback of music programs in ordinary home systems, including car radio, rich and expressive enough, the dynamic range is compressed, making the sound of quiet sounds more strong. Otherwise, in addition to the occasional bursts of fortissimo (in classical music), you will have to listen to the silence from the speakers, especially given the noisy environment. For this, devices called compressors are used. In some cases, on the contrary, it is required to expand the dynamic range, then expanders are used. And to exclude exceeding the maximum level, which will lead to clipping (limiting the signal from above, accompanied by very high non-linear distortions, perceived as wheezing), limiters are used in studies.

special effects for studios, EMP, etc.: available to sound engineers and musicians there is a large number of special equipment to give the sound the desired color or to obtain a specific effect. These are various distorters (the sound of an electric guitar becomes hoarse, grainy), wah-wah prefixes (amplitude modulation that causes a characteristic “croaking” effect), enhancers, and exciters (devices that affect the color of the sound, in In particular, it can give the sound a “tube” tint); flangers, choruses, etc.

sound mixing, echo / reverb: recording in studios is usually done in multi-channel form, then, using mixers, the phonogram is reduced to the required number of channels (usually 2 or 6). In this case, the sound engineer can “push forward” one or another solo instrument recorded on a separate track, changing the loudness ratio of different tracks. Sometimes multiple copies of a lower level are superimposed on the signal with a certain time shift, thus simulating natural reverb (echo). Currently, similar and other effects are mainly achieved using signal processors that process digital signals.


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Audio. Digital and Analog Audio Part 5

Audio. Digital and Analog Audio Part 5

Digital Audio

Any amplification path is non-linear, so harmonic distortion always occurs – new frequency components spaced 3, 5, 7, etc. in frequency. of the tone that generates them (odd harmonics) or in 2, 4, 6, etc. times (even).

Digital Audio

 

The threshold of visibility of harmonic distortions varies widely: from a few tenths or even hundredths of a percentage to 3-7%, depending on the composition of the harmonics. Even the harmonics are less noticeable, since they are in line with the fundamental tone (the difference in frequency is twice corresponding to one octave).

In addition to harmonic distortions, intermodulation distortions occur, which are the differential products of the frequencies of the signal spectrum and its harmonics. For example, at the output of an amplifier, at the input of which two frequencies of 8 and 9 Hz are applied (with a sufficiently non-linear characteristic), a third (1 kHz) will appear, as well as several others: 2 kHz (as the difference of the second harmonics of the fundamental frequencies), etc. … Intermodulation distortion is especially annoying to the ear, as it generates many new sounds, including those that are dissonant to the main ones.

What an audiophile can hear, and not only hear, but also explain to a sound engineer, can be completely invisible to the average listener.

Noise and distortion are largely masked by the signal, but they themselves mask low-level signals that fade or lose clarity. Therefore, the higher the signal-to-noise ratio, the better. Actual sensitivity to noise and distortion will vary based on individual hearing characteristics and training. The level of noise and distortion that does not affect the transmission of speech can be completely unacceptable for music. What an audiophile can hear, and not only hear, but also explain to a sound engineer, can be completely invisible to the average listener.

ANALOGUE AUDIO TRANSFER
Traditionally, audio signals were transmitted over cables and over the air (radio).

Distinguish between unbalanced (classic cable) and balanced transmission line. Unbalanced has two wires: signal (direct) and return (ground). Such a line is very sensitive to external interference, so it is not suitable for transmitting a signal over long distances. Often implemented with a shielded cable, the shield is grounded.

cifrovoe-i-analogovoe-audio-4.jpg
FIG. 4. Unbalanced screened line

The balanced line assumes three wires: two signal wires, through which the same signal flows, but in antiphase, and ground. On the receiving side, the common mode noise (induced in both signal wires) is mutually subtracted and completely disappears, and the useful signal level is doubled.

FIG. 5. Balanced screened line

Unbalanced lines are often used inside devices and for short distances, mainly on user routes. In the professional sphere, balance prevails.

In the figures, the shield connection points are shown conditionally, as they must be selected “in place” each time to achieve the best results. Most of the time, the screen is connected only on the signal receiver side.

Audio. Digital and analog audio Part 3

Audio. Digital and analog audio Part 3

DIGITAL AUDIO

Modern autumn sound sources are diverse and digital media are becoming more and more common: CDs, DVDs, although vinyl records are also preserved. We continue to listen to radio, both terrestrial and via cable (radio hotspots). Sound accompanies television shows and movies, not to mention such a familiar phenomenon as telephony.

Digital Audio

 

A computer receives an increasing share in the world of audio, allowing it to conveniently archive, combine and process sound programs in the form of files. In the digital age, digitized speech and music are transmitted through digital channels, including the Internet, without serious losses in transportation. This is done with digital encoding and the loss is due solely to compression, which is used most often. However, in digital media, either it does not exist at all (CD, SACD), or lossless audio compression algorithms are used (DVD Audio, DVD Video). In other cases, the degree of compression is determined by the required level of quality of the soundtrack (MP3 files, digital telephony, digital television, some types of media).

cifrovoe-i-analogovoe-audio-1.jpg
FIG. 1. Conversion of acoustic sound vibrations into an electrical signal

The reverse conversion of electrical vibrations to acoustic vibrations is carried out using speakers built into radios and televisions, as well as separate acoustic systems, headphones.

Sound is called acoustic vibrations in the frequency range 16 Hz to 20,000 Hz.

Sound is called acoustic vibrations in the frequency range 16 Hz to 20,000 Hz. Below (infrasound) and above (ultrasound), the human ear does not hear, and within the sound range, the sensitivity of hearing is very uneven. , its maximum falls at a frequency of 4 kHz. To hear sounds of all frequencies at the same volume, you must play them at different levels. This technique, called loudness, is often implemented in home computers, although its result cannot be considered unequivocally positive.

cifrovoe-i-analogovoe-audio-2.jpg
FIG. 2. Equal volume curves
(Click on the image to zoom)

The physical properties of sound are generally not presented in linear values, but in relative logarithmic values, decibels (dB), as this is much clearer in numbers and more compact in graphics (otherwise one would have to operate with values ​​that they have many zeros before and after the decimal point, and the second would be easily lost in the context of the first). The ratio of two levels A and B in dB (say voltage or current) is defined as:

With u [dB] = 20 lg A / B. If we talk about powers, then C p [dB] = 10 lg A / B.

In addition to the frequency range, which determines the human ear’s sensitivity to tone, there is also the concept of loudness range, which shows the ear’s sensitivity to loudness level and covers the range from the lowest audible sound to the ear (threshold sensitivity) to the strongest, beyond which is the pain threshold. The sensitivity threshold is taken as a sound pressure of 2 x 10-5Pa (Pascal), and the pain threshold is pressure, 10 million times higher. In other words, the audibility range, or the pressure ratio between the loudest and the lowest sound, is 140 dB, which is markedly higher than the capabilities of any audio equipment due to its own noise. Only high definition digital formats (SACD, DVD Audio) match the theoretical limit of dynamic range (the ratio of the loudest sound reproduced by the equipment to the noise level) 120 dB, CD provides 90 dB, vinyl record – approximately 60 dB.

cifrovoe-i-analogovoe-audio-3.jpg
FIG. 3. Hearing sensitivity range

Only high definition digital formats (SACD, DVD Audio) match the theoretical dynamic range limit

Noise is always present in the audio path. This is both the intrinsic noise of the amplifying elements and the external interference. Signal distortions are divided into linear (amplitude, phase) and non-linear or harmonic. In the case of linear distortion, the signal spectrum is not enriched with new components (harmonics), only the level or phase of the existing ones changes. Amplitude distortions that violate the original level relationships at different frequencies result in audible timbre distortions. For a long time it was believed that phase distortions were not critical to hearing, but today the opposite has been shown: both timbre and sound localization are highly dependent on the phase relationships of the signal’s frequency components. .

Audio. Digital and analog audio

Audio. Digital and analog audio

Digital Audio

Although we assimilate most of the external information with the help of our eyes, sound images are no less important to us and often even more.

Digital Audio

Try watching a movie with the sound turned off; in 2-3 minutes you will lose the thread of the plot and the interest in what is happening, no matter how big the screen and the high quality image. Therefore, a pianist played off-screen in silent movies. If you remove the picture and leave the sound, the movie can be “heard” like a fascinating radio show.

Hearing gives us information about what we do not see, since the sector of visual perception is limited, and the ear captures the sounds that come from everywhere, complementing the visual images.

Hearing gives us information about what we do not see, since the visual perception sector is limited, and the ear captures sounds from all directions, complementing visual images. At the same time, our hearing with great precision can locate an invisible sound source in direction, distance, speed of movement.

They learned to convert sound into electrical vibrations long before images. This was preceded by a mechanical recording of sound vibrations, whose history dates back to the 19th century.

Accelerated progress, including the ability to transmit sound at a distance, was made possible by electricity, with the advent of amplification, acoustic and electroacoustic technology and transducers – microphones, pickups, dynamic heads, and other emitters. Today, audio signals are transmitted not only over cables and over the air, but also over fiber optic communication lines, primarily in digital form.

Acoustic vibrations are converted into an electrical signal, usually by microphones. Any microphone contains a moving element whose vibrations generate a current or voltage in a certain way. The most common type of microphone is the dynamic one, which is a reverse speaker. The vibrations of the air set in motion a membrane that is rigidly connected to a moving coil in a magnetic field. A condenser microphone is, in fact, a condenser, one of whose plates vibrates in time with the sound, and with it the capacitance between the plates changes. Ribbon microphones use the same principle, only one of the plates is freely suspended. Similar to a condenser electret microphone, whose plates, in the process of oscillation, generate by themselves an electric charge proportional to the amplitude of the oscillations. Many models of microphones have a built-in amplifier (the level of the signal directly from the acoustic-electric transducer is very low). Unlike a microphone, the pickup of an electric musical instrument registers vibrations not from air, but from a solid body: a string or the soundboard of an instrument. The cartridge reads the disc slot using a stylus mechanically connected to moving coils in a magnetic field, or magnets if the coils are stationary. Or the vibrations of the needle are transmitted to the piezoelectric element which, under mechanical stress, generates an electrical charge. In magnetic recording, an audio signal is recorded on a magnetic tape and then read with a special head. Finally, in cinematography, optical recording was traditionally adopted: an opaque soundtrack was applied from the edge of the film,

In synthesizers, sound is born directly in the form of electrical vibrations, there is no primary transformation of acoustic waves into an electrical signal.

History of Digital Audio Part 2

History of Digital Audio Part 2

Digital Audio

Different formats use different methods of audio compression, but bit rate still plays a role as a measure of audio quality. The sample rate also plays an important role and the number of hertz shows how many parts per second the file is divided into. The lower limit of the sample rate for audio files is 44.1 kHz (44100 Hz), if it is lower, it is not sufficient.

digital audio

VBR vs CBR

Constant Bit Rate (CBR) and Variable Bit Rate (VBR) are two methods of obtaining Bit Rate. Constant bitrate means that you set a certain bitrate for the entire file, and with a variable bitrate, its value changes throughout the entire music file as needed.

CBR is like packing something in a larger box than necessary, and VBR packs in a box that matches the outline of its contents. People often use an overestimated bit rate of 320 kbps, when this is not necessary, often a VBR of 192 kbps is sufficient. By ear, you are unlikely to feel a difference.

DRM

DRM (Digital Rights Management) is the most terrible invention since the nuclear bomb and is best left untouched. Music stores primarily use DRM protection to protect it from illegal copying and use.

DRM files are not compatible with all players and you may forget to transfer files in MSC / UMS mode with them. DRM-protected music is usually in WMA or AAC formats. In short, the use of DRM only creates additional problems for people.

History of digital audio

History of digital audio

digital audio

By its nature, sound is an oscillatory movement of particles in an elastic medium that propagates in the form of waves. After it became clear that sound represents such vibrations, the idea came up of recording them by repeating the shape on solid material.

DIGITAL AUDIO

So, in 1877, Thomas Edison created a phonograph, a device for the mechanical recording and reproduction of sound. And in 1888, the German E. Berliner invented the gramophone – the era of gramophone records began, which became the first massive carriers of audio information.

Thomas Edison and his phonograph

FIG. Inventor Thomas Edison and His Record Making: The Phonograph

Having studied the laws of electromagnetism, man made successful experiments to convert sound waves into electromagnetic waves and preserve them. This is how magnetic tape appeared, which became widespread in the middle of the 20th century.

For digital technology to store, process, and reproduce sound, it is converted to digital format by an analog-to-digital converter (ADC), which converts an analog signal into a sequence of numbers. This is called Pulse Code Modulation (PCM).

It happens like this: the ADC measures the amplitude of an analog signal many times per second and outputs the results in the form of numbers. However, the measurement result does not exactly match a continuous electrical signal: it depends on the number of measurements and their precision.

The frequency at which the measurements are taken is called the sample rate, and the precision of the amplitude measurements indicates the number of bits used to indicate the result of the measurement. This parameter is called the bit depth. For example, if the sampling frequency is 44.1 kHz, this means that the signal is measured 44 100 times in one second.

For the analog signal to be accurately reconstructed from its samples, the sample rate must be twice the maximum audio frequency. That is, if the analog signal contains frequency components from 0 Hz to 20 Hz, then the frequency of its sampling must be at least 40 kHz.

Digital audio formats

Of course, for digitized sound to be stored, transmitted, and converted, there must be certain digital sound standards – audio formats. Today, there are many such formats, each of which uses its own sound processing algorithm. They also differ in the information carriers.

The most popular and widespread in the field of home use today are ordinary music CDs – CDs. There are also relatively new recording formats, Super Audio Compact Disk (SACD) and DVD-Audio (or simply DVD-A). In addition, formats that use digital data compression have become widespread.

The most popular among them is MPEG-1/2 / 2.5 Layer 3 (MP3). Microsoft also did not stay away from the sound industry, as it developed its own compression algorithm, WMA, which is also actively promoted in the market.

New audio file formats appear every year, but no player on the market supports the playback of all formats.

In fact, the term MP3 player is only correct for players that support the MP3 format. Let’s see what’s what in audio formats.

Before looking at the various audio file formats (codecs), let’s take a look at a few terms.

Bitrate

Bit rate is the space required for 1 second of music. With a bit rate of 128 kbps (kilobits per second) = 16 kbps (kilobytes per second), approximately 5 megabytes are needed for 5 minutes of music.

The higher the bit rate, the higher the quality of the music. But this as long as the bit rate of the original format is higher than the bit rate of the encoded format. By compressing a CD to MP3 at 320 kbps, you get better sound quality than 128 kbps, but converting from 128 kbps to 320 kbps will not improve the quality and may even degrade it.

Often times a 128kbps bit rate masquerades as CD quality, but this is not actually the case. If you have enough high-quality equipment, you will hear it immediately. Manufacturers like to give an estimate of the number of songs that go into a player at a very low bit rate, and many consumers are unaware that audio files vary in size. Therefore, you should not rely on the numbers in the advertisements, in fact, much less the songs in your collection can fit in the player.

Compression

Uncompressed audio takes up a lot of space. To reduce the size of audio files in formats such as MP3, programs cut off the part of the frequency range that the human ear cannot hear.

Hardware for processing digital audio – Part 4

Hardware for processing digital audio – Part 4

digital audio processing

As a practical example of a MIDI device, consider a conventional MIDI keyboard.

DIGITAL AUDIO PROCESSING MAC

Simplified, a MIDI keyboard is a shortened grand piano keyboard in a housing that contains a MIDI interface that allows you to connect it to other MIDI devices, such as a MIDI synthesizer, that is installed on your computer’s sound card. With special software (for example, a MIDI sequencer), you can turn a MIDI synthesizer into play mode, for example, on a grand piano, and by pressing the keys on a MIDI keyboard, you can hear the sounds of a piano from line. Naturally, the matter is not limited to the grand piano: in the GM standard there are 128 melodic instruments and 46 percussion instruments. Additionally, using a MIDI sequencer, you can record notes played on a MIDI keyboard on a computer for further editing and arrangement, or simply to print notes.

It should be noted that since MIDI data is a set of commands, music written using MIDI is also recorded using synthesizer commands. In other words, a MIDI score is a sequence of commands: what note to play, what instrument to use, how much and how much it will sound, etc. Familiar MIDI files (.MID) are more than just a collection of such commands. Naturally, since there are a large number of MIDI synthesizer manufacturers, the same file can sound differently on different synthesizers (because the instruments themselves are not stored in the file, there are only instructions to the synthesizer on which instruments to play. while the differences synths may sound different).

Let’s go back to the consideration of sound cards. As we have already clarified what MIDI is, we cannot ignore the characteristics of the hardware synthesizer built into the sound card. A modern synthesizer, most often, is based on the so-called “wave table” – WaveTable (in short, the principle of operation of such a synthesizer is that the sound in it is synthesized from a set of recorded sounds dynamically superimposing them and change of sound parameters), before the main synthesis type was FM (Frequency modulation: sound synthesis generating simple sinusoidal oscillations and mixing them). The main characteristics of a WT synthesizer are: the number of instruments in the ROM and their volume, the presence of RAM and its maximum volume, the number of possible signal processing effects, as well as the possibility of channel-by-channel effect processing. . (of course, in the case of an effects processor), the number of oscillators that determines the maximum number of voices in polyphonic mode (polyphonic) and, perhaps most importantly, the standard by which the synthesizer is manufactured (GM , GS or XG). By the way, the amount of memory of the synthesizer is not always a fixed value. The thing is that recently synthesizers have stopped having their own ROM, but use the main RAM of the computer: in this case, all the sounds used by the synthesizer are stored in a file on disk and, if necessary, are they read into RAM.

Hardware for processing digital audio – Part 3

Hardware for processing digital audio – Part 3

digital audio processing

A MIDI synthesizer is a synthesizer that meets the requirements of the standard that we will now talk about. MIDI is a generally accepted specification related to the organization of a digital interface for musical devices, which includes a standard for hardware and software.

Digital Audio Processing

This specification is intended to organize a local area network of electronic instruments (Fig. 7). MIDI devices include a variety of musical instruments and hardware that meet MIDI requirements. Therefore, a MIDI synthesizer is a musical instrument, generally intended to synthesize sound and music, and also conforming to the MIDI specification. Let’s briefly see why there is a separate class of devices called MIDI.

The fact is that the implementation of sound processing software is often associated with drawbacks due to various technical characteristics of this process. Even leaving sound processing operations on a sound card or any other equipment, many different problems remain. First, it is often desirable to use hardware synthesis of musical instrument sounds (at least because a computer is too general an instrument, often only a hardware sound and music synthesizer is needed, nothing more). Second, software sound processing is often accompanied by time delays, while concerted work requires instantaneous reception of the processed signal. For these and other reasons, they resort to the use of special equipment for processing, and not computers with special programs. However, when using equipment, there is a need for a single standard that allows devices to connect to each other and combine. It was these prerequisites that led several leading companies in the musical equipment field to approve the first MIDI standard in 1982, which was subsequently continued and continues to this day. What, ultimately, is a MIDI interface and the devices included in it from a personal computer’s point of view?

Hardware: These are installed on the sound card: a synthesizer of various sounds and musical instruments, a microprocessor that controls and controls the operation of MIDI devices, as well as several standardized connectors and cables for connecting additional devices.
Programmatics is a MIDI protocol, which is a set of messages (commands) that describe various functions of the MIDI system and with which communication (information exchange) between MIDI devices takes place. The messages can be considered as a means of remote control.
The scope of this article does not allow us to delve into the description of MIDI in particular, it should be noted, however, that with respect to sound synthesizers, MIDI sets strict requirements for their capabilities, the sound synthesis methods used in them. , as well as for the synthesis control parameters. Furthermore, in order for music created on one synthesizer to be easily transferred and played successfully on another, several standards have been established for the matching of instruments (voices) and their parameters on various synthesizers: the General MIDI (GM) standard, General Synth (GS) and eXtended General (XG). The basic standard is GM, the other two are its logical extensions and extensions.

Hardware for processing digital audio – Part 2

Hardware for processing digital audio – Part 2

Digital Audio Processing

4. Mixing unit. On sound cards, the mixing unit provides adjustment of:

DIGITAL AUDIO PROCESSING

signal levels of the line inputs;
MIDI input and digital audio input levels;
the level of the general signal;
panorama
doorbell.
Let us consider the most important parameters that characterize sound boards and sound-music. The most important characteristics are: maximum sample rate in record mode and in playback mode, maximum sample rate or bit depth (maximum quantization level) in record and playback mode. Furthermore, since sound cards also have a synthesizer, the parameters of the installed synthesizer also refer to its characteristics. Naturally, the higher the quantization level that the card is capable of encoding the signals, the better the signal quality. All modern sound card models are capable of encoding a signal with a 16-bit level. One of the important features is the ability to simultaneously play and record audio streams. Function cards play and record simultaneously is called full duplex (full duplex). There is another characteristic that often plays a decisive role when buying a sound card: the signal-to-noise ratio (Signal-to-noise ratio, S / N). This indicator affects the purity of the signal recording and playback. The signal-to-noise ratio is the ratio between the signal power and the noise power at the output of the device; this indicator is generally measured in dB. A good ratio is 80 to 85 dB; ideal – 95-100 dB. However, it should be noted that the quality of playback and recording is strongly influenced by interference (interference) from other components of the computer (power supply, etc.). As a result, the signal-to-noise ratio may deteriorate. In practice, there are many methods to solve this problem. Some suggest grounding the computer. Others, to protect the sound card from interference as much as possible, “pull” it out of the computer case. However, it is very difficult to completely protect yourself from interference, as even the map elements themselves are created by floating above each other. They are also trying to fight this by filtering every item on the board. But no matter how much effort is made to solve this problem, it is impossible to completely eliminate the influence of external interference.

Another equally important characteristic is the non-linear distortion coefficient, or total harmonic distortion, THD. This figure also critically affects the clarity of the sound. The non-linear distortion coefficient is measured in percentage: 1% – “dirty” sound; 0.1% – normal sound; 0.01%: pure Hi-Fi sound; 0.002% – High Fidelity Sound – Hi-End .. Non-linear distortion is the result of inaccuracy in restoring the signal from digital to analog. Simplified, the process of measuring this coefficient is carried out as follows. A pure sine signal is supplied to the input of the sound card. At the output of the device, a signal is taken, the spectrum of which is the sum of the sinusoidal signals (the sum of the original sinusoid and its harmonics). Then, using a special formula, the quantitative ratio of the original signal and its harmonics obtained at the output of the device is calculated.

What is a MIDI synthesizer? The term “synthesizer” is commonly used to refer to an electronic musical instrument in which sound is created and processed, changing its color and characteristics. Naturally, the name of this device comes from its main purpose – sound synthesis. There are only two main methods of sound synthesis: FM (frequency modulation) and WT (wave table). Since we cannot dwell on them in detail here, we will describe only the main idea of ​​the methods. FM synthesis is based on the idea that any oscillation, even the most complex, is essentially the sum of the simplest sinusoids. Thus, it is possible to superimpose signals from a finite number of sinusoid generators and, by changing the frequencies of the sinusoids, obtain sounds similar to the real ones. Wavetable synthesis is based on a different principle. Sound synthesis using this method is achieved by manipulating the prerecorded (digitized) sounds of real musical instruments. These sounds (called samples) are stored in the permanent memory of the synthesizer.

Hardware for processing digital audio

Hardware for processing digital audio

Digital Audio Processing

An important part of the conversation about sound has to do with hardware.

Digital Recording

There are many different devices for audio processing and input / output. With regard to an ordinary personal computer, one should dwell on sound cards in more detail. Sound cards can be divided into sound, music and zvukomuzykalnye. By design, all sound cards can be divided into two groups: main (installed on the computer motherboard and providing audio data input and output) and daughter (they have a fundamental structural difference from main boards ; most of the time they are connected to a special connector located on the main board). Daughter cards are most often used to provide or extend the capabilities of a MIDI synthesizer.

Sound, music and sound cards are created in the form of devices inserted into the motherboard slot (or already integrated from scratch). Visually, they usually have two analog inputs: line and microphone, and several analog outputs: line outputs and a headphone output. Recently, the cards have also been equipped with a digital input and output, which provides audio transmission between digital devices. The analog inputs and outputs usually have connectors similar to the headphone jacks (1/8 ”). Generally, the sound card has a little more than two inputs: analog CD, MIDI, and other inputs. Unlike the mic and line inputs, they are not located on the back panel of the sound card, but on the card itself; there may be other inputs, for example to connect a voice modem. The digital inputs and outputs are usually S / PDIF (digital signal transfer interface) with a corresponding connector (S / PDIF stands for Sony / Panasonic Digital Interface – Sony / Panasonic digital interface). S / PDIF is a “home” version of the more complex professional standard AES / EBU (Audio Engineering Society / European Broadcast Union). The S / PDIF signal is used to digitally transmit (encode) 16-bit stereo data at any sample rate. In addition to the above, sound-music cards have a MIDI interface with connectors for connecting MIDI devices and joysticks, as well as for connecting a daughter music card (although recently the ability to connect the latter has become a rarity). Some sound card models are equipped with a front panel for user convenience,

Let’s define several basic blocks that make up the sound and sound-music boards.

1. Digital signal processing block (codec). This block is used for analog-to-digital and digital-to-analog conversions (ADC and DAC). This block determines the characteristics of the card, such as the maximum sample rate for recording and playback of a signal, the maximum quantization level, and the maximum number of processed channels (mono or stereo). To a large extent, the characteristics of noise also depend on the quality and complexity of the components of this block.

2. Synth Block. Present on musical cards. Made on the basis of FM or WT synthesis, or both at the same time. It can work both under the control of its own processor, and under the control of a special controller.

3. Interface block. Provides data transfer over various interfaces (eg S / PDIF). A purely sound card often lacks this block.

4. Mixing unit. On sound cards, the u