Weekend reading: materials on setting up audio equipment and designing speaker systems

Speakers in a car, on a hoverboard, portable speakers on a bicycle - music is everywhere today, you can take it with you anywhere and few people bother with setting it up. But the requirements for the sound of home systems are usually higher. Even the coolest model will sound worse if it is connected and configured somehow incorrectly. The article will tell you how to do everything correctly so that the acoustics sound good.

How to create an audio system at home

• Myths and reality: What you need to know about modern audio systems. In this material we answered common questions. You will learn how professional audio equipment differs from “amateur” audio equipment and why in-ear headphones do not harm your ears. We will also tell you about outdoor audio systems that will work in any weather - both rain and frost.

• Five tips: how to properly organize a street audio system. It is more difficult to design an audio system for the street than for a home. Due to the absence of walls, the bass sounds weaker; quiet music is easily drowned out by street noise. This is a translation of material with tips that will help organize an audio system outside the home. We are talking about the inverse square law, which explains the propagation of sound on the street, and the speakers most suitable for outdoor audio systems.

• How to properly place speakers in a room. The first step to creating a home audio system is to think about how to place your speakers and subwoofers in your room. We tell you how to calculate the placement of speakers using an acoustic calculator and how to prepare a room for placing an audio system.

• How the speakers are arranged inside: types of acoustic design. In this article we will tell you how the internal structure of the speaker body—the acoustic design of the speaker—affects its sound. In addition, we will consider the advantages and disadvantages of different types of cases.

• How to adjust room acoustics. In this material we described how you can change the acoustic properties of a room. You will learn more about sound-absorbing materials, acoustic traps and interior items that will help get rid of errors in the sound of your audio system.

• Audio system calibration: why do electronic room correction. Room correctors change the sound of an audio system based on the acoustic properties of the room. Let's talk about exactly how they work and how to use them at home.

Mikhail Olshevsky

It's no secret that the sound quality at concerts sometimes leaves much to be desired. It would seem that great musicians play the best instruments, experienced, qualified sound engineers “sculpt” the sound picture, and the days of the wretched “homemade” have sunk into oblivion, and the result still does not always please listeners. In some cases, the cause of failure is the unsatisfactory performance of the sound system. Let us clarify right away that we will be talking about concert systems, that is, systems for playing music where emotional and artistic perception is important, which does not prevent, however, from extending some of the conclusions to other sound systems. Without pretending to cover the problem completely, we will try to understand the objective and subjective reasons that hinder the achievement of high-quality sound at a concert, and also consider some ways to eliminate them using configuration and setup methods for the sound system. First, let's try to understand what, in fact, are the requirements for a concert sound system; what we have a right to expect from her work; where is the limit of its “satisfactory”? Obviously, ultimately we (that is, listeners, sound engineers and other interested parties) want to satisfy our subjective expectations from listening to musical material. One of the successful options for a subjectively oriented approach to sound reproduction is the creation of Hi-End class systems. However, due to the fact that in this area it is customary to operate with mystical characteristics rather than technical ones, this approach is unsuitable for professional work. Professionalism means repeatable results. Repeatability of results is an integral feature of any professional work. Repeatability of the result (always, regardless of the field of activity) requires an objective approach, and therefore the presence of objective characteristics, standards and measurement tools developed on their basis.

Sound Reproduction Requirements

In the first half of the 20th century, the requirements for high-quality sound reproduction were reduced to the bandwidth of reproduced frequencies, uniformity of the amplitude-frequency response (AFC), low nonlinear distortion and adequate dynamic range. It was believed that the phase relationships of spectral components do not affect the perception of sound. With the deepening of knowledge in the field of psychoacoustics, improving the quality of sound-reproducing devices, the advent of stereophony, and the development of musical technologies, it became clear that transient characteristics (i.e., accurate reproduction of sharp changes in sound pressure) and closely related phase-frequency characteristics ( FCHH). Let's make a short digression about the temporal and spectral representations of the signal. It is known that any signal can be represented both in the time domain using the dependence of the instantaneous value on time, and in the frequency domain using the signal spectrum, that is, in the general case, an infinite number of harmonic oscillations of different frequencies with a certain amplitude and initial phase. It should be noted that the abstract mathematical concept of spectrum differs significantly from the usual “picture” on a spectrum analyzer, which would be more accurately called a spectral density meter. The relationship between the time domain and frequency domain representation of a signal is described by the mathematical Fourier transform. To characterize the properties of the signal transmission path, which makes it possible to calculate the output signal from a known input signal, the following are used: in the frequency domain, amplitude-frequency and phase-frequency characteristics (AFC and PFC); in the time domain, either an impulse or a transient response, which is a mathematical description of the response at the output of the path when a delta function or a Heaviside function is applied to its input, respectively. These two functions are idealized mathematical abstractions. The delta function is an impulse of infinitesimal duration with unit energy. Obviously, the amplitude of such a pulse will be infinite. The Heaviside function represents a single jump in the signal (that is, the signal is zero in the time interval from minus infinity to zero and one in the time interval from zero to infinity). The mathematical Laplace transform allows you to calculate the output signal of a path, knowing its impulse or transient characteristics. This method can be used, for example, when modeling reverberation processes in a room. A good approximation to the delta function in this case is the firing of the starting pistol. The room response is recorded and digitized. The Laplace transform will allow you to “superpose” the properties of the room on any input signal. Thus, we will get a reverb that simulates the properties of a specific hall. In different situations, preference may be given to both the spectral and pulsed research methods, but it should be clearly understood that these are simply two views on the same physical phenomena. For digitized signals, the discrete Fourier transform (DFT) and the fast Fourier transform (FFT) algorithm developed on its basis are used, and instead of the Laplace transform, its analogue for discrete functions is used - the so-called Z-transform.

Studio - an exemplary system

Currently, the closest sound reproduction system to the ideal should be considered a high-quality studio monitoring system. Although it is far from ideal, the de facto studio is still a reference tool for professional control. The entire modern recording industry is built on the basis of this standard, and its universally recognized masterpieces have been created. The fact that all sound engineers conduct a subjective assessment of the quality of a sound system by playing their favorite soundtrack (from that same collection of masterpieces) indirectly confirms the actual acceptance of this standard. The subjective standard sound quality of the studio is enshrined in strict objective criteria of technical parameters: – uniform amplitude-frequency response; – linear phase-frequency characteristic of phase response; – sufficient dynamic range; – minimal nonlinear distortion, especially for high-order harmonics. Much attention is paid to the response to impulse action. Strict requirements are imposed on the premises. This set of requirements allows you to convey the primary signal to the listener in its original form (whether it is a primary audio signal or a synthesized one), and introduce a minimum of distortion during signal transmission. The author is far from thinking that a formal transfer of the sound of studio monitors to a concert hall can be an ideal solution to all problems, but it can become the foundation on which a good sound engineer can adequately realize his artistic ideas. Without such a foundation, a sound engineer is doomed to build in a swamp. The question of the need to ensure equal sound pressure and uniform frequency response in all places in the auditorium remains controversial. Most experts agree that more preferable for aesthetic perception are a general sound pressure level that smoothly decreases away from the stage and a slightly greater decrease in the high-frequency part of the spectrum. Also, a certain boost in the low frequencies is usually required at concerts of modern music. These preferences are still more subjective and depend on the overall concept of the show.

Three factors that distinguish a concert system and prevent the creation of studio conditions on a large stage

In order to formulate the requirements for a concert system, it would be logical to turn to the riders of concert artists, where sound engineers formulate the requirements for the system. It would seem that these requirements should guarantee a certain minimum level of quality of system operation. Let's take a critical look at some typical rider points. Acoustic systems of brands or types X or Y, but in no case Z - unfortunately, as practice shows, this is not a guarantee of results, but still... System power, not less than ... kW in the hall (the option W per capita is possible population), not less than...kW in the open air. Sometimes it is specified RMS or Program - a parameter that characterizes little due to the fact that it is calculated arbitrarily depending on honesty, the degree of understanding of the issue, etc.; in fact, with comparable results, the declared power may differ significantly. Sound pressure at the console ...dB (the type of weighting filter is sometimes indicated) is a comparable and easily verified parameter, but is still rarely found among riders. The range of reproduced frequencies is from...Hz to...Hz; Frequency response unevenness ...dB; Uniform distribution of sound across all spectator seats (unevenness is sometimes indicated...dB) - these requirements are reasonable in themselves and form a system of comparable and easily verifiable parameters, but the numerical values ​​given in the riders often go not only beyond the scope of reasonable sufficiency, but also beyond physical feasibility. What are, for example, 138(!) dB at the remote control, a range from 20 to 20000 Hz, unevenness +/- 0 dB, etc.? Needless to say, obviously unrealizable requirements in at least one point of the rider lead to a negative perception of the document as a whole, a disdainful attitude towards other points of the rider, which provokes a lot of conflicts. But that is another topic. And for us it is important that the sound quality at a concert, unfortunately, has little to do with the performance of the riders. Let's return to the studio as a standard and try to figure out what prevents us from achieving “studio quality” at a concert. Let's first consider what the main differences between a concert system and studio monitors are and what problems these differences give rise to. Next, taking these differences into account, we will try to “project” the technical requirements for studio sound onto a concert system. Difference 1 – large sound area
If studio monitors are intended practically for individual listening, then a large system should ideally provide “everything the same”, only for each of the listeners, and their number can reach tens of thousands.
The large distances traveled by a sound wave make tangible physical effects that do not manifest themselves in a small room. This is a decrease in sound pressure with distance due to a non-planar wave front, frequency-dependent absorption in the air, a change in the direction of wave propagation under the influence of wind or a vertical temperature gradient (changes in air temperature depending on height). These phenomena practically make it impossible to provide high-quality sound to a space more than 50 meters long using a concentrated group of loudspeakers. Let's look at this problem in more detail. To obtain a minimum decay of sound pressure with distance, the emitted wave must be plane. The size of the radiating surface in this case must exceed the transverse size of the audience and be commensurate with its length. Since compliance with this condition is obviously impossible, we will have to put up with a certain (depending on the generated wave front) drop in sound pressure. Frequency-dependent sound absorption inevitably leads to a change in the frequency response of the system depending on the distance between the loudspeaker and the listening point and a corresponding change in the timbre coloring of the sound. Distortion of a given wave front, associated with a change in the direction of its propagation, leads to a significant change in the distribution of the sound field compared to the calculated one, a change in the spectral balance, etc. These effects are most pronounced when working outdoors. Probably everyone has heard how the sound at a concert changes after sunset, how the “flangering” effect appears when the wind blows. A possible solution to the problems described in this paragraph is the use of a distributed system in which the distance from each listener to the nearest loudspeaker is minimal and equal. However, such a system is unsuitable for scoring a concert for many reasons, one of which is the lack of correct localization of the sound image. Another problem with “long distances” is the relatively small zone of satisfactory stereo effect. That is, if you try to formally transfer a home or studio stereo system to a large hall, the result for most listeners will not be the best. Difference 2 – a large number of emitters spaced apart in space.
The power of a single loudspeaker is insufficient to sound a large space, so the use of group emitters in the form of stacks, clusters, line arrays, etc. is inevitable.
There are no emitters that satisfactorily reproduce the entire range of sound frequencies, so the division of the range into separate bands is also inevitable. To ensure correct operation of narrowband emitters, filters are used, which, as a rule, introduce phase distortion. Emitters operating simultaneously in one band, as well as emitters of neighboring bands at the crossover frequency, interact (interfer) with each other, creating a complex picture of the distribution of the sound field over the sounded area. This is one of the main problems that arise when implementing large sound systems, so we will dwell on this point in more detail later. Of course, one should not “demonize” the word “interference” itself. It is simply a physical phenomenon that can be either beneficial or harmful in different situations. But we must take into account the influence that it has on sound, firstly, in order to be aware of the limits of the physical feasibility of our requirements for the sound system, and secondly, in order to be able to minimize the undesirable consequences of this phenomenon for us. Difference 3 – the rooms in which the sound system operates have a wide variety of sizes and acoustic properties.
How does the room affect the operation of the sound system? Each enclosing surface of the room reflects the original sound wave. In turn, the newly formed wave is reflected from other surfaces. Multiple reflections form the reverberation process in the room. If in the region of medium and high frequencies the reflections and the primary signal are weakly correlated with each other and the reverberation is diffuse in nature, then in the region of low frequencies a characteristic interference pattern appears in the form of standing waves. Reverberation, as the most important property of auditoriums, has received much attention since the first stage performances appeared. This is a well researched topic. But in the low-frequency region, problems associated with reflected sound have become more acute with the advent of sound reinforcement systems, electric musical instruments and the development of modern music, which is generally characterized by a high level of the low-frequency part of the spectrum. And the short attacks of the sounds of the rhythm section, which are of utmost importance for the perception of modern music, are fundamentally different from the sound of the double bass group of a symphony orchestra.

System Dynamics and Nonlinear Distortions

The maximum sound pressure created by the system must ensure the achievement of the desired emotional impact on the listener, which is determined by the performer, sound engineer and other interested parties, but may be regulated by sanitary standards and other legislative acts. Regulations vary greatly from country to country. In Russia, the sound pressure level in theatrical and entertainment institutions is regulated, in particular, by the Labor Safety Rules in Theaters and Concert Halls, approved by the Ministry of Culture of the Russian Federation on December 1, 1998. The Rules apply to all operating theaters and concert halls, regardless of their departmental affiliation. According to this document, the sound level in the hall with sound reinforcement should not exceed 96 dB, and with sound design - 100 dB. While studio loudspeaker manufacturers have the luxury of regulating loudspeaker power ratings for playback quality reasons, the acceptable operating modes of drivers used in concert systems define the limit beyond which they will be physically destroyed. There is no need to talk about the quality of sound reproduction when operating at maximum power. The coefficient of nonlinear distortion (THD) of a loudspeaker is very rarely published by manufacturers, obviously because of its value, which frightens a potential buyer. If in the electrical part of the audio path we are talking about SOI of the order of 0.01% or less, then the SOI of a powerful loudspeaker is rarely below 10% at a power of 10% of the maximum permissible. At maximum power, SOI can reach 100% or more. In fact, all this is not as scary as it might seem. The distortions that occur in loudspeakers are “soft”, the main share of them falls on the second and third harmonics. Such distortions are perceived by ear much better than “non-musical” distortions created by amplifiers and other devices in the electrical part of the path. In addition, the maximum power of the system is required, as a rule, in fairly short moments of the “apotheosis” of the concert, and the average level is much lower. Nevertheless, from the above it follows that the desire of everyone interested in good sound to have high system power and, accordingly, a good sound pressure margin and low distortion is quite understandable. Just don’t forget that every 3 dB (some 3 dB) on the remote control indicator requires doubling the number of speakers, amplifiers, energy consumption, cables, etc. In a large touring system, this turns into truckloads of heavy boxes, tons of loads on suspended structures and money...money...money... Therefore, choosing the power of a sound system is always a difficult compromise between the wishes of the sound engineer and the budget. The approach to resolving this issue must be responsible and justified. Given the above, it is extremely important to use all the system's power reserves. Unused reserves often lie in limiter settings and inconsistencies in the level chart. A careless attitude to the setting of limiter response leads to significant underutilization of power or to failure of the drivers. Let us explain with a typical example. Often proprietary controller presets are used without taking into account the specific type of amplifier. In the case when the preset is designed for amplifiers with a sensitivity of 0 dB, and amplifiers with a sensitivity of +6 dB are used, we get a four (!) times less power of the system. If the reverse mismatch occurs, we may end up with quadruple driver overload and/or amplifier overload. Another example is a mismatch between the output level of the mixing console and the input sensitivity of the system controller. As a rule, controllers are designed for the highest possible input levels. The nominal level can be +6, +12, or +18 dB. If the remote control indicators are “targeted”, for example, at 0 dB, then the lack of power will be catastrophic. As a rule, errors in constructing a system level diagram are also associated with the appearance of an overload in the electrical part of the path, producing distortions that should not normally appear. To make maximum use of the system's power reserves, proper sound engineering is also extremely important. Let us give just one typical example. The author repeatedly had the opportunity to observe the following picture at a concert of a famous Russian rock band: hitting a bass drum brought the remote control indicator to a level of 0 dB, while the rest of the time the level did not exceed -10 dB. Moreover, everything was acceptable by ear. The power reserve of the sound system of the club where the concert took place allowed such a free distribution of it, that is, 90% for the “barrel”, and 10% for everything else. It is clear that by correctly tuning the drum (namely, the musical instrument itself) and using dynamic processing, you can achieve the same result by redistributing the percentages the other way around, and thus save 90% of the system power. In a concert on a large stage, such a “luxury” is obviously unacceptable. It is necessary to remember one more type of nonlinear distortion. Often, experienced sound engineers introduce an overall smooth roll-off in the frequency response of the system at frequencies above 10 kHz. This measure can be regarded as a forced compromise. It is associated primarily with distortions that occur at high pressures in horn emitters and phase-equalizing waveguides of linear arrays. The result of these distortions will be the appearance of additional multiples, as well as total and difference frequencies in relation to the original signal. While multiple and sum frequencies fall in the upper part of the audio range, where hearing perception is reduced, or outside the audio range, difference frequencies fall in the lower range, where hearing is most sensitive. For example, the product of distortion of two signals with frequencies of 15 kHz and 18 kHz will, in particular, sound with a frequency of 3 kHz. In the case of playing a complex musical signal with a spectrum overloaded in the 10-20 kHz region, the lower part of the range reproduced by the HF driver may be supplemented by a large number of artifacts that are terrifying in their unmusicality. This type of distortion is easily noticeable when playing choral vocals if the low end frequency of the HF driver is around 1 kHz or lower. This phenomenon can be combated using known methods: dividing the range under consideration into bands and generally increasing the linearity of the emitters. The possibility of radically improving the characteristics of high-frequency range emitters is provided by FIR filters, already used in some digital controllers. Until recently, high requirements for processor performance and RAM capacity hampered the use of this technology in systems working with live sound. The ability to independently influence the amplitude and phase characteristics of the filter, as well as precision step-by-step design of the impulse response, opens up new opportunities for compensating for distortions introduced by a real emitter, significantly bringing its characteristics closer to the ideal. In addition, FIR filters make it possible to significantly increase the steepness of the frequency section in crossovers, reducing the interference of emitters of adjacent bands and without introducing phase distortion. Unfortunately, in the low-frequency region there are restrictions on the use of FIR filters, due to the increase in latency (introduced delay) of the device in which the filter is implemented to unacceptable values. A latency of two to three milliseconds can be considered acceptable for a system controller, which corresponds to a distance of no more than a meter covered by a sound wave during this time. Non-linear distortions that occur in the hearing aid are also useful to take into account when mixing, especially at high sound pressure levels. The nonlinearity of the hearing aid is characterized by the appearance of low-order harmonics, in this it is similar to the nonlinearity of loudspeakers. Low-order harmonics are much less perceived by the ear, so even with a nonlinear distortion coefficient of tens of percent (both in loudspeakers and in the ears), the sound can remain acceptable. The most unpleasant manifestations of these distortions are the appearance of distortion products of other frequencies in the band of greatest hearing sensitivity (2-4 kHz). These can be harmonics of frequencies 1-2 kHz or difference frequencies from a higher range. This manifests itself in a very sharp, “ear-drilling” sound. In this case, it is practically useless to combat the effect by “cutting out” the 3 kHz region; it may be much more effective to reduce frequencies around 1.5 kHz.

Frequency range and power distribution

Reproducing the full range of frequencies that are considered audio, that is, 20 Hz-20 kHz, is a difficult technical task even in the studio. The large required power of the RF drivers of a concert system leads to an increase in the size and weight of their moving system and, as a result, to a decrease in efficiency at the highest frequencies. Effective reproduction of the lowest frequencies requires the use of subwoofers (or arrays thereof) of very large size and high power. Therefore, in practice, the frequency range is limited to 30-50 Hz at the bottom and 16-18 kHz at the top. The exception is subwoofers for special effects in theater and cinema, where the lower limit of reproduction can be even below 20 Hz. The distribution of the system's output acoustic power over the frequencies of the audio range should approximately correspond to the spectral distribution of the music being reproduced. A common reference signal replacing the music signal in power-related acoustic measurements is pink noise. The spectrum of pink noise is significantly similar to the average spectrum of a musical signal. As a rule, in multi-band full-range cabinets, the ratio of driver powers of different bands corresponds to the distribution of pink noise. Subtle variations are possible in the low frequency region, making different cabinets more or less preferable to a particular style of music. When designing a large system, you should also keep in mind that to match the sound system to the style of music, a few decibels of extra bass may be required (compared to what is needed to reproduce pink noise). Converting these few decibels into power units, we may find that, for example, several times more subwoofers will be required. These factors should also be taken into account when choosing the “caliber” of a line array. For cabinets intended to be combined into clusters, the principle of uniformity of the frequency response of a separate cabinet was traditionally observed, despite the fact that the frequency response of a cluster was not very similar to the frequency response of a separate cabinet. But since there were no real possibilities for calculating the resulting characteristics of the cluster (and without the use of a computer with the appropriate software, this is unlikely to be done in practice), the uniform frequency response at least gave a certain predictability of its “behavior.” The widespread adoption of line array technology coincided with the proliferation of digital speaker controllers and the advent of affordable computers and software. Newly developed speaker systems are usually supplied with presets for digital controllers. And if for a single “cluster” type cabinet the combination of a proprietary preset and the cabinet itself usually gives a uniform frequency response, then for a single element of a linear array the approach can be fundamentally different. Since individual elements of a linear array are not intended for independent use, the requirement for a uniform frequency response for them, essentially, does not make sense. The frequency response of a linear array is highly dependent on its size and curvature, as well as on the distance and direction to the measurement point. Without going into the physical basis of how linear arrays work, we will note their important differences. Firstly, a line array always requires equalization. Secondly, the HF section of the line array element must have many times more power compared to a traditional broadband cabinet. In extreme cases, the required equalization in the HF region (provided that the frequency response of a single array element is uniform) can exceed 20 dB, and in the LF region – 10 dB. Later we will return to the issues of equalizing line arrays.

Phase characteristics and phase distortion

The effect of frequency response distortion on sound is well understood by everyone who has used any equalizer at least once. As for the phase response, its influence on the sound is not so obvious; formal requirements for the system in this area are usually not imposed, but its influence on the aesthetic perception of concert sound is very significant. Therefore, let us dwell on this issue in more detail. Deviation of the phase response from linear leads to different delays of signal components with different frequencies. This leads to subjective blurring of the sound, loss of cohesion, focus, detail, the sound becomes cloudy, opaque, etc. The sound of the bass drum stops “beating the chest”, although it sounds more or less normal. Let's see what it looks like. To explore the adventures, or rather misadventures, of a signal in the audio path, we use a 5 millisecond rectangular pulse, shown in the figure below.

Test pulse

The choice of a signal of real duration (as opposed to mathematical abstractions) does not allow quantitative analysis of the path, but it has a spectrum close to real audio signals and is convenient for observation on the oscilloscope screen. Let us recall that for undistorted signal reproduction, not a uniform, but a linearly decreasing phase response is required (or a zero phase response as a special case of linear). Linear phase response decay means the overall delay of the signal (that is, the same for all its spectral components) without distorting its shape. The following figure shows the zero phase response of the path under test, and below it the result of its influence on the test pulse.

Zero phase response

Visible insignificant distortions are associated with the imperfection of our experimental path, namely with the limitation of the reproduced frequencies from below at a level of about 10 Hz. Below is shown the linearly decreasing phase response of a path introducing a delay of 5 ms.

Linear phase response of a path introducing a delay of 5 ms

The phase response line on the graph undergoes breaks every 360 degrees of phase change. These breaks are conditional. If you draw a graph without breaks, it will not fit on a reasonable size of paper. The curvature of the line is related to the logarithmic scale on the frequency axis. You can understand that the function is linear by estimating the phase changes occurring at equal frequency intervals. For example, in the interval from 0 to 100 Hz the phase changes by 180 degrees, in the interval from 100 to 200 Hz - also by 180 degrees, etc. Below in the figure we see a pulse delayed by 5 ms, which passed through the path without visible distortion.

Pulse delayed by 5 ms

Now let's look at the nonlinear phase response of the path and the result of its influence on our impulse.

"> An example of a nonlinear phase response of a path

"> Output pulse. Nonlinear phase response

We see that the impulse has undergone significant distortion. The short rise and fall of the pulse, containing most of the energy of the high-frequency components, remained in place, and the low-frequency component of the pulse was delayed. Let us recall that the frequency response of the tract under study is uniform. To simulate phase distortions, we used one second-order phase-correcting (or so-called allpass) filter. In reality, phase distortions are introduced to one degree or another by all parts of the sound system. Taking into account the fact that any filters with which we purposefully influence the frequency response have a side effect on the phase response, the resulting phase distortions of the signal can change the signal beyond recognition (an exception may be digital filters implemented on the basis of finite impulse response or FIR filters, but This is a topic for a separate conversation). Let's now see how the signal is distorted by various parts of the electrical part of the path.

"> Distortion introduced by the low-cut filter (24 dB, 30 Hz)

Distortion introduced by a two-way crossover and a low-cut filter

Comments, I think, are unnecessary. Let us now feed the sampled sound of a bass drum, which is shown in the following figure, to the input of the path.

The sound of a big drum at the input of the tract

Let's look at the effect of a two-way crossover and low-cut filters on the sound of a bass drum.

"> The sound of a bass drum at the output of the tract

The shape distortions are not as noticeable as on the test signal, but it is clearly visible that the amplitude of the oscillations in the initial part of the pulse has become smaller, and the main share of slow low-frequency oscillations has begun to lag behind the area of ​​fast oscillations responsible for the “click.” Thus, the sound of the bass drum becomes "blurred". Loudspeakers, and especially groups of them, introduce even more significant phase distortions. The next two figures show the typical distortion introduced by an inexpensive one-third octave graphic equalizer with the 100 Hz frequency boosted (top) and cut (bottom) by 15 dB. We use a test pulse with a duration of 5 ms.

Pulse with raised frequency 100 Hz

Pulse with cut frequency 100 Hz

We also see a noticeably distorted pulse, which has also acquired a long resonant tail. Let us note once again that all classical filters used in electrical circuits to adjust the frequency response inevitably affect the phase response. Moreover, the general rules are as follows: the steeper the filters we use, that is, the larger the amplitude differences over a smaller frequency interval we create, the larger the phase distortions we get. An exception may be appropriately designed FIR filters, an important property of which is the ability to independently synthesize the frequency response and phase response. This leads to important conclusions. 1. All filters included in the signal path are potential sources of phase distortion. High slope filters should be used with caution. 2. It is necessary to take into account the influence of filters on the phase shift of the signal at the crossover frequencies. Even the inclusion of a “harmless” low-cut filter can be significant. In combination with the phase shift introduced by the cabinet, you can get a complete “antiphase” at the crossover frequency and a corresponding dip in the frequency response. In the next part of the article we will look at the effects of interference of emitters, the influence of the room on the operation of the sound system, we will touch on the issues of equalization of line arrays, we will talk about methods of forming polar patterns, paying attention to “cardioid” subwoofers and directional sub-bass arrays.
December 26 at 00:06 Master class 2 2011 (65)

How to choose audio devices

• How to choose headphones. We tell you how to make the right choice and protect yourself from buying counterfeit goods. Let's look at the entire purchasing process: from setting a budget to starting to use it.

• “Sound is everything”: Review of microphones for creating audio/video content. In a vlog or podcast, clear audio is just as important as the content. Based on recommendations from famous podcasters and streamers, we have compiled a list of affordable and worthy speech microphones.

• 10 tips when buying a portable audio player. Smartphones help you accomplish many tasks, but the ability to listen to high-definition music isn't available on every device. The solution for music lovers may be to purchase a portable audio player, advice on choosing which you will find in this article.

• How to choose speakers: some tips. We understand the variety of types of columns. Let's talk about the features of different manufacturing countries, sizes and price categories. In the second part of the article we will talk about the structure and operating principles of speakers.

Photo: PxHere/PD

• Which wireless technology is right for you? Although wireless speakers are most commonly associated with Bluetooth, there are competing wireless standards. In this article, we look at what other wireless technologies underlie audio systems, and what are the pros and cons of each of them. Let's talk about protocols that work with both mobile devices and personal computers.

Special Settings

The following functions will be useful for some car enthusiasts:

• turning off Demo mode;
• setting date and time;
• setting up the display of traffic jams via TMC on the radio.

Demo mode is needed to demonstrate the capabilities of the device in a store environment. If you leave it, then when the device is turned off, the backlight and luminous inscriptions on the display will work.

You can disable Demo mode in the hidden menu by pressing the SRC key on the radio that is turned off. By turning the joystick handle, find the DEMO item and switch the sensor from the ON to OFF position. The BAND button is used to exit the menu.

Time and date are also selected in a hidden menu, but in the System section, where you can select 12 or 24-hour mode and change the desired indicators by turning the wheel.

Traffic displays via TMC are supported by Garmin and iGo programs. By installing one of them, you can receive information about traffic on the roads of large cities (about traffic jams, accidents, road repairs, etc.). iGo has the ability to save a list of radio stations that broadcast data of interest to the car owner. Using these programs, the display of the radio can display the route between specified points, the time of possible delays due to traffic jams, etc. To operate these applications, you need a set of iGo Primo licenses and special TMC licenses.

In order to set sound reproduction parameters in the car radio that will allow the owner to enjoy good quality music, it is not necessary to contact specialists; you can do it yourself. High sound quality can be achieved from an inexpensive audio system by correctly defining the sound reproduction parameters. Some models have the ability to install additional applications that will increase the level of comfort when traveling by car.

How to set up audio equipment

• How to set up a vinyl player. The sound of a vinyl record depends on how accurately the player's needle conveys the topography of the track. Just buying a good model is not enough - an incorrectly installed player will distort the sound. In this article we talk about the stages of setting up vinyl players and the main components of turntables.

• How to get the best sound from your CD player. Although setting up a CD player doesn't have as much of an impact on the sound as setting up a vinyl turntable, a few simple steps can help make your CDs sound cleaner. In this article we have collected “tricks” that will reduce sound distortion when playing a disc.

• Bluetooth audio: wireless audio characteristics. Many Bluetooth devices use the standard SBC codec to compress the audio stream, which can affect the sound of the material. We'll tell you about different ways to transmit Bluetooth audio and recommend devices that support improved codecs.

• What is a balanced headphone connection and why is it needed? By connecting headphones balanced to an amplifier, music lovers increase the volume and detail of the sound. Moreover, “regular” headphones can also be made balanced - just connect a special cable to them. We'll tell you how a balanced connection works and understand the principles of connecting headphones to an amplifier.

• Sound is not true to size: minimizing the disadvantages of bookshelf speakers. Bookshelf speakers are used in acoustic systems for small rooms, the size of which does not exceed 20 square meters. In this article we tell you how the location of the speaker affects its sound and give practical advice on setting it up.

Microphone and speaker device

The operation of all modern audio equipment is based on the use of a process of processing, transmission and amplification by converting audio frequencies into an electrical signal and vice versa. At the same time, speakers and microphones become the most important components of such equipment.

What is acoustics

The concept of “acoustics” has many meanings, each of which is associated with sound. But first of all, it is the science of sound, its physical nature, the principles of origin, perception, and distribution. One of its sections is electroacoustics, which allows you to study issues of reception, reproduction, and recording of sound information using technology.

It is within the framework of such scientific research that issues of the formation and development of broadcasting systems, television, radiotelephone communications, and sound amplification systems are studied. When it comes to electrical equipment, acoustics (or acoustic system) is a device that is used to convert current signals into sound vibrations.

How to listen to music from your computer

• Computer for Hi-Fi: what hardware is needed for comfortable sound reproduction. To play Hi-Fi audio, you don’t have to buy a separate player - you can listen to music from your computer. We'll tell you what "stuffing" a "music lover's PC" can have - let's talk about choosing a hard drive, video card, processor, power supply and software.

• Only music. Making the PC system unit quiet. In this article we will tell you in more detail about silent computer components that will not interfere with listening to music from your computer. Let's talk about the main sources of noise in a PC, soundproofing the case, silently cooling the processor, and choosing a storage device for your music collection.

• We are building a computer desk for a music lover. The sound of any speakers will be distorted if the speaker system is placed on a tabletop made of insufficiently dense material - in this case, the table acts as a resonator. “A music lover’s computer desk” can be bought in a store, but it will be cheaper to make it yourself. From the article you will learn what and how to build a table for a computer and speakers.

Photo: PxHere/PD

• Network music: computer specs for Hi-Fi and how to store your music collection. A few more tips for those who want to listen to Hi-Fi audio from a computer. In the first article, we’ll take a closer look at choosing an external DAC and tell you how a music playback program can affect the sound. In the second article we will discuss the differences between different data storage devices.

• How to digitize vinyl. Digitizing analog media such as vinyl records is a great way to expand your digital media library. In this article, we'll tell you what equipment and software you need to digitize vinyl, how to test your player before recording, and what format to save your files in.

Our Telegram channel is about audio gadgets in microformat:
How to make listening to an audio system at home more comfortable A guide for a beginner: what is important to know about headphone ear pads How do you use old audio equipment? A few more words about using audio gadgets Where to start when choosing home acoustics?

Step-by-step setup

Setting up the radio is described in detail in the 2022 Pioneer DEH-1900UB instruction manual. It will be carried out in this way also when changing the factory settings of other similar devices. The exception will be processor devices, the configuration of which is more difficult.

When changing sound settings, it is important to pay attention to the following indicators:

• low frequency sounds;
• balance of front and rear speakers;
• distribution of the sound signal between the speakers on the right and left sides.

Equalizer

Setting the equalizer on the radio helps improve sound quality, even when using a low-quality speaker system. This device helps regulate the frequency of sound.

The equalizer (EQ) can be found in the menu section called "Audio". At this point, it is not the entire sound range that is regulated, but the required frequency bands (in Pioneer radios there are 5 of them: 8 kHz, 2.5 kHz, 800 Hz, 250 Hz, 80 Hz). The frequency at which the filter gain changes is called the speaker cutoff threshold.

Pioneer car radios provide several options for standard equalizer settings and 2 sets of custom settings that the owner can create himself. Switching between these versions is done from the menu or with the EQ key.

To configure the radio yourself, you need to confirm the selection of the “Equalizer” item in the “Audio” section by pressing the joystick. By turning it, you can select the desired frequency and press the center of the knob again, setting the position in the range from -6 to +6. This will change the volume of the selected frequencies.

The consumer himself determines which sound parameters to set. There are several recommendations that will allow you to make the right choice of characteristics:

• to play rock music, it is better to increase the bass volume (80 Hz) to +2;
• percussion instruments sound good at a frequency of 250 Hz;
• voice transmission parameters are adjustable to 250-800 Hz;
• frequencies for electronic music - 2.5-5 kHz.

After completing the parameter changes, swing the knob to the left to access the main menu.

High pass filter

After this, the music is configured in the HPF menu item, the full name of which is High Pass Filter. It stops sound from the speakers that is higher than the set value to reduce the amount of melody distortion.

If the system does not have a subwoofer, it is better to set the HPF threshold at frequencies of 50-63 Hz. It is recommended to check the result at a volume of 30.

If you have a subwoofer, you can raise the lower threshold to 80-120 Hz and higher without degrading the sound quality.

This will cause the music playback volume to increase.

When adjusting the frequency attenuation rate, it is recommended to select 24 dB per 1 octave.

Low pass filter

Using a low-pass filter, you can configure the car radio for a subwoofer in the corresponding menu section.

It provides 3 subwoofer mode values:

• The cutoff frequency of the subwoofer, which can be set at the owner’s request, in the range of 63-100 Hz.
• Speaker volume on a scale from -6 to +6. It is recommended to set the parameter to the same value as when setting the equalizer.
• Slope of frequency attenuation. It is better to choose the value selected in the HPF item (12 or 24 dB).

If the settings are consistent, the high and low frequencies in the melody will be balanced, and the person using the speaker system will not experience discomfort when listening to it.

Radio setup

To set up a radio in a Pioneer radio, you need to select a band, find and save the desired radio stations. This can be done in several ways:

• Automatic search for radio stations. In the main menu you should find the BSM section; in it you can start a search for stations with the highest frequency in the radio range. After this, you need to save it, assigning the button value from 1 to 6 to it. Then the search for stations will continue. In the hidden menu, you can change the search step from 100 kHz to 50 kHz to expand the range of the operation.
• Semi-automatic search for radio stations. If you press the Right key in radio mode, a search for stations will start.
• Manual radio tuning. Pressing the “Right” button on the control panel several times while in radio mode will switch to a frequency. The found station is saved in the device memory.

How to turn on and adjust the sound in speakers on a PC and laptop

If everything is physically connected, but there is still no sound, you need to click on the speaker icon in the lower right corner of the screen and check if it is disabled. Here you can adjust the volume.

More extensive adjustments are made in the “Speaker Properties” window, in which we go through the “Sound” option in the control panel. In this window you can adjust the sound and adjust the volume.

When using a desktop computer, speakers are often built into the monitor. It also contains audio controls.

Why is there noise coming from the speakers when the computer is turned on?

Another common problem that users encounter is noise coming from the speakers. It appears even if everything worked fine yesterday.

The most common causes of noise are the following:

• defective or broken speakers;
• incorrect choice of connectors when connecting;
• damage or squeezing of wires;
• conductors of poor quality;
• driver problem;
• bad contact.

You can connect built-in and external speakers to your PC and laptop. The latter are connected via a regular audio jack, USB or bluetoth. If the sound quality is poor, we check the drivers, cables, connectors, and operation of the sound card.

Why is there no sound when the speakers on the computer and laptop are turned on?

It would seem that everything was connected according to the instructions, but the sound that you have been waiting for for so long does not come from the speakers.

• Are the speakers connected to the network and PC?
• Is the sound on the computer turned on? Click on the speaker icon in the lower control panel and make the necessary settings;
• whether codecs are installed. This problem often occurs when playing video files;
• Is the sound card disabled? You can check this through the control panel;
• whether the cables are connected.