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Audio. Digital and Analog Audio Part 6

ANALOG AUDIO PROCESSING

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

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).

 

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.

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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

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.

 

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).

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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.

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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.

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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. .