Horn antenna: description, design, properties and use. Exponential horn, its purpose and application. Exponential horn of a cylindrical wave front, its advantage Classification of loudspeakers by design

A loudspeaker is a device that converts an electrical sound signal at the input into an audible acoustic signal at the output. To ensure proper quality, the loudspeaker must operate loudly and efficiently - reproduce the sound signal in the permissible (audible) dynamic (85-120dB) and frequency (200-5000Hz) ranges.

Loudspeakers have the widest application in various spheres of human activity: in industry, transport, sports, culture, and consumer services. For example, in industry, loudspeakers are used to provide public address communication (PAC), in transport - for emergency communications, announcements, in the domestic sphere - for paging alerts, as well as background music broadcasts. In the field of culture and sports, the most widely used are professional acoustic systems designed for high-quality musical accompaniment of events. Sound support systems (SSS) are built on the basis of such systems. Loudspeakers are actively used in a wide range of organizational measures to protect the population: in the field of security - in warning systems and evacuation control (SOEC), in the field of civil defense - in local warning systems (LSS) and are intended for direct (sound) warning of people in case of fire and emergency situations.

2. Transformer loudspeakers

Transformer loudspeakers - loudspeakers with a built-in transformer are the final executive elements in wired broadcast systems, on the basis of which fire warning systems, local warning systems, and public address systems are built. In such systems, the principle of transformer matching is implemented, in which a separate loudspeaker or a line with several loudspeakers is connected to the high-voltage output of the broadcast amplifier. Signal transmission in a high-voltage line allows you to maintain the amount of transmitted power by reducing the current component, thereby minimizing losses on the wires. In a transformer loudspeaker, there are 2 stages of conversion. At the first stage, a transformer is used to reduce the voltage of the high-voltage audio electrical signal; at the second stage, the electrical signal is converted into an audible acoustic sound signal.

The illustration shows the back of a cabinet wall-mount transformer loudspeaker. The transformer loudspeaker consists of the following parts:

The loudspeaker housing, depending on the application, can be made of various materials, the most widespread of which today is ABC plastic. The housing is necessary for ease of installation of the loudspeaker, protection of live parts from dust and moisture, improvement of acoustic characteristics, and formation of the required directivity pattern (NDP).

The step-down transformer is designed to lower the high-voltage voltage of the input line (15/30/60/120V or 25/75/100V) to the operating voltage of the electrodynamic converter (speaker). The primary winding of a transformer can contain multiple taps (e.g., full power, 2/3 power, 1/3 power), allowing the power output to vary. The taps are marked and connected to the terminal blocks. Thus, each such tap has its own impedance (r, Ohm) - reactance (of the primary winding of the transformer) depending on frequency. By choosing (knowing) the impedance value, you can calculate the power (p, W) of the loudspeaker at various voltages (u, V) of the input broadcast line, as:

p = u 2 / r

The terminal block provides convenience for connecting the broadcast line to various taps of the primary winding of the transformer loudspeaker.

Speaker is a device for converting an electrical signal at the input into an audio (audible) acoustic signal at the output. Connects to the secondary winding of the step-down transformer. In a horn loudspeaker, the role of the speaker is played by a driver rigidly attached to the horn.

3. Speaker device

Speaker (electrodynamic transducer) is a loudspeaker that converts an electrical signal at the input into sound waves at the output using a mechanical moving diaphragm or diffuser system (see figure, picture taken from the Internet).

The main working unit of an electrodynamic loudspeaker is a diffuser, which converts mechanical vibrations into acoustic ones. The speaker cone is driven by a force acting on a coil rigidly attached to it and located in a radial magnetic field. An alternating current flows in the coil, corresponding to the audio signal that the loudspeaker must reproduce. The magnetic field in the loudspeaker is created by a ring permanent magnet and a magnetic circuit of two flanges and a core. The coil, under the influence of the Ampere force, moves freely within the annular gap between the core and the upper flange, and its vibrations are transmitted to the diffuser, which in turn creates acoustic vibrations propagating in the air.

4. Horn loudspeaker device

The horn loudspeaker is the (active primary) means of reproducing the audio acoustic signal in the permissible frequency and dynamic ranges. The characteristic features of the horn are the provision of high acoustic sound pressure due to a limited opening angle and a relatively narrow frequency range. Horn loudspeakers are used mainly for voice announcements and are widely used in places with high noise levels - underground parking lots, bus stations. Highly concentrated (narrowly directed) sound allows them to be used on railways. stations, in subways. Most often, horn loudspeakers are used for sounding open areas - parks, stadiums.

A horn loudspeaker (horn) is a matching element between the driver (emitter) and the environment. The driver, rigidly connected to the horn, converts the electrical signal into sound energy, which is received and amplified in the horn. The sound energy inside the horn is amplified due to a special geometric shape that provides a high concentration of sound energy. The use of an additional concentric channel in the design makes it possible to significantly reduce the size of the horn while maintaining quality characteristics.

The horn consists of the following parts (see figure, picture taken from the Internet):

• metal diaphragm (a);
• voice coil or ring (b);
• cylindrical magnet (c);
• compression driver (d);
• concentric channel or projection (e);
• megaphone or bugle (f).

A horn loudspeaker works as follows: an electrical sound signal is fed to the input of a compression driver (d), which converts it into an acoustic signal at the output. The driver is (rigidly) attached to the horn (f) providing high sound pressure. The driver consists of a rigid metal diaphragm (a) driven (excited) by a voice coil (coil or ring b) wound around a cylindrical magnet (c). The sound in this system propagates from the driver, passing through a concentric channel (e), is exponentially amplified in the horn (f), and then goes to the output.

NOTE: In various literature and depending on the context, the following names of the horn may be found - megaphone, bugle, loudspeaker, reflector, trumpet.

5. Connecting transformer speakers

In broadcast systems, the most common option is when several transformer loudspeakers need to be connected to one broadcast amplifier, for example, to increase volume or coverage area.

If you have a large number of speakers, it is most convenient to connect them not directly to the amplifier, but to a line, which in turn is connected to the amplifier or switch (see figure).

The length of such lines can be quite long (up to 1 km). Several such lines can be connected to one amplifier, and the following rules must be observed:

RULE 1: Transformer speakers are connected to the broadcast amplifier (only) in parallel.

RULE 2: The total power of all loudspeakers connected to the broadcast amplifier (including through the relay module) should not exceed the rated power of the broadcast amplifier.

For convenience and reliability of connection, it is necessary to use special terminal blocks.

6. Classification of loudspeakers

A possible classification of loudspeakers is shown in the figure.

Loudspeakers for public address systems can be classified into the following categories:

• By area of ​​application,
• According to the characteristics,
• By design.

7. Application area of ​​loudspeakers

Loudspeakers have a wide range of applications: from loudspeakers used in quiet indoor spaces, to loudspeakers used in noisy open areas, depending on the acoustic characteristics - from voice announcements to background music broadcasts.

Depending on the operating conditions and area of ​​application, loudspeakers can be divided into 3 main groups:

1. Indoor loudspeakers – used for use in enclosed spaces. This group of loudspeakers is characterized by a low degree of protection (IP-41).
2. External loudspeakers – used for use in open areas. Such loudspeakers are sometimes called street speakers. This group of loudspeakers is characterized by a high degree of protection (IP-54).
3. Explosion-proof loudspeakers (explosion-proof) are used for use in explosive areas or in areas with a high content of aggressive (explosive) substances. This group of loudspeakers is characterized by a high degree of protection (IP-67). Such loudspeakers are used in the oil and gas industries, at nuclear power plants, etc.

Each group can be associated with a corresponding class (degree) of IP protection. The degree of protection is understood as a method that limits access to dangerous live and mechanical parts, the ingress of solid objects and (or) water into the shell.

The degree of protection of the enclosure of electrical equipment is marked using the international protection mark (IP) and two numbers, the first of which means protection from the ingress of solid objects, the second - from the ingress of water.

The most common degrees of protection for loudspeakers are:

• IP-41 where: 4 – Protection from foreign objects larger than 1 mm; 1 – Vertically dripping water must not interfere with the operation of the device. Loudspeakers of this class are most often installed in enclosed spaces.
• IP-54 where: 5 – Dust protection, in which a certain amount of dust can penetrate inside, but this should not interfere with the operation of the device; 4 – Splashes. Protection against splashes falling in any direction. Loudspeakers of this class are most often installed in open areas.
• IP-67 where: 6 – Dust tightness, in which dust should not get into the device, complete protection from contact; 7 – During short-term immersion, water should not enter in quantities that interfere with the operation of the device. Loudspeakers of this class are installed in places subject to critical influences. There are also higher degrees of protection.

8. Speaker characteristics

Loudspeakers, depending on the field of application and the class of tasks being solved, can be further classified according to the following criteria:

• by the width of the amplitude-frequency response (AFC);
• by radiation pattern width (WPD);
• by sound pressure level.

8.1 Classification of loudspeakers by frequency response width

Depending on the width of the frequency response, loudspeakers can be divided into narrow-band, the bands of which are sufficient only for reproducing speech information (from 200 Hz to 5 kHz) and wide-band (from 40 Hz to 20 kHz), used for reproducing not only speech, but also music.

The frequency response of a loudspeaker in terms of sound pressure is a graphical or numerical dependence of the sound pressure level on the frequency of the signal developed by the loudspeaker at a certain point in the free field, located at a certain distance from the working center at a constant voltage value at the loudspeaker terminals.

Depending on the width of the frequency response, loudspeakers can be narrowband or wideband.

Narrowband loudspeakers are characterized by a limited frequency response and, as a rule, are used to reproduce speech information in the range from 200...400 Hz - a low male voice, to 5...9 kHz - a high female voice.

Wideband loudspeakers are characterized by a wide frequency response. The sound quality of a loudspeaker is determined by the magnitude of the unevenness of the frequency response - the difference between the maximum and minimum values ​​of sound pressure levels in a given frequency range. To ensure proper quality, this value should not exceed 10%.

8.2 Classification of loudspeakers according to the width of the radiation pattern

The directivity pattern width (DPW) is determined by the type and design of the loudspeaker and significantly depends on the frequency range.

Loudspeakers with a narrow PDP are called highly directional (for example, horn loudspeakers, spotlights). The advantage of such loudspeakers is their high sound pressure.

Loudspeakers with wide NDP are called wide-directional (for example, acoustic systems, sound columns, cabinet loudspeakers).

8.3 Classification of loudspeakers by sound pressure

Loudspeakers can be conventionally distinguished by their sound pressure level.

Sound pressure level SPL (Sound Pressure Level) - sound pressure value measured on a relative scale, referred to a reference pressure of 20 μPa, corresponding to the threshold of audibility of a sinusoidal sound wave with a frequency of 1 kHz. The SPL value called loudspeaker sensitivity (measured in decibels, dB) should be distinguished from the (maximum) sound pressure level, max SPL, which characterizes the loudspeaker's ability to reproduce the upper level of the declared dynamic range without distortion. Thus, the sound pressure of a loudspeaker (in passports denoted as maxSPL) is otherwise called loudspeaker volume and consists of its sensitivity (SPL) and electrical (nameplate) power (P, W), converted into decibels (dB), according to the rule of “ten logarithms":

maxSPL = SPL + 10Lg(P)

From this formula it is clear that a high or low level of sound pressure (loudness) largely depends not on its electrical power, but on the sensitivity determined by the type of loudspeaker.

Indoor loudspeakers, as a rule, have a maxSPL not exceeding 100dB, while the sound pressure, for example, of horn loudspeakers can reach 132dB.

8.4 Classification of loudspeakers by design

Loudspeakers for broadcast systems vary in design. In the most general case, loudspeakers can be divided into cabinet loudspeakers (with an electrodynamic loudspeaker) and horn loudspeakers. Cabinet loudspeakers, in turn, can be divided into ceiling and wall, mortise and overhead. Horn loudspeakers may differ in aperture shape - round, rectangular, material - plastic, aluminum.

An example of classifying loudspeakers by design is given in the article "Design features of ROXTON loudspeakers."

9. Speaker placement

One of the urgent problems is the correct choice of type and quantity. With the correct speaker placement scheme, you can achieve good results - high sound quality, background intelligibility, uniform (comfortable) sound distribution. Let's give a few examples.

For sounding open areas, horn loudspeakers are used due to their characteristics such as a high degree of sound directionality and high efficiency.

It is recommended to install sound floodlights in corridors, galleries and other extended rooms. The spotlight can be installed either at the end of the corridor - a unidirectional spotlight, or in the middle of the corridor - a bidirectional spotlight and can easily penetrate lengths of several tens of meters.

When using ceiling loudspeakers, it is necessary to take into account that the sound wave from the loudspeaker propagates perpendicular to the floor, therefore, the sounded area, determined at the height of the listeners’ ears, is a circle, the radius of which for a 90° radiation pattern is taken equal to the difference between the height of the ceiling (loudspeaker mounting) and the distance to marks 1.5 m from the floor (according to regulatory documents).

In most problems for calculating ceiling acoustics, the (geometric) ray method is used, in which sound waves are identified with geometric rays. In this case, the radiation pattern of the ceiling speaker determines the angle of the top of the right triangle, and half of the base determines the radius of the circle. Thus, to calculate the area voiced by a ceiling loudspeaker, the Pythagorean theorem is sufficient.

To provide even sound throughout a room, speakers should be installed so that the resulting areas slightly overlap each other. The required number of loudspeakers is obtained from the ratio of the area being sounded to the area sounded by one loudspeaker. The placement of loudspeakers is determined by the geometry of the building. The distance between loudspeakers, or spacing, is determined based on the coverage areas. If the placement is incorrect (exceeding the pitch), the sound field will be distributed unevenly, and in some areas there will be dips that worsen perception.

In the case of using loudspeakers with high sound pressure, the level of the reverberation background increases, which leads to such a negative phenomenon as echo. To compensate for this effect, the floor and walls of the room are covered or trimmed with sound-absorbing materials (for example, carpets). Another cause of reverberation is improper speaker placement. In rooms with high ceilings, loudspeakers that are placed closely together can cause a lot of interference between each other. To reduce this influence, it is advisable to place the speakers at a greater distance, but to maintain the characteristics, you will have to increase the power. In such cases, it may be recommended to use suspended audio speakers.

The placement of loudspeakers in rooms is carried out after preliminary calculations. Calculations can both confirm and determine various arrangement patterns, the most effective of which are: arrangement according to the “square lattice”, “triangle”, checkerboard pattern. For the placement of loudspeakers in corridors, the main design parameter is the spacing.

Issues related to electroacoustic calculations and the placement of loudspeakers will be covered in detail in the next article.

The principle of operation of a horn emitter - section Education, Basic principles of the design of concert complexes. Mixing consoles. Equalizers and their applications. Connecting Cables and Connectors The Roughest Explanation of the Operating Principle of a Horn Emitter Can Be Made...

The roughest explanation of the principle of operation of a horn emitter can be done as follows. If you want to be heard from a great distance, then you must turn in the direction from which you can be heard and cup your hands near your mouth. In this case, your phrase in the forward direction will be heard louder than in all others, which is explained by the direction of the sound waves you create.

Without a horn, the energy of the sound waves from the sound source is distributed evenly in all directions, so the volume of the sound in any of these directions is the same.

A horn focuses the energy of sound waves from a source within a certain angle, so the volume of sound in the region of space limited by this angle is higher than in all other directions.

Human hearing has maximum sensitivity in the audio frequency range of the vocal range. The average frequency of this region is approximately 1000 Hz. In a four-band sound reproduction system, the value of this frequency lies on the border between the mid-low and mid-high frequency bands, so any inaccuracy in the tuning of these two frequency channels is very noticeable to the ear and sharply worsens the sound of the entire sound reproduction system. In order to completely eliminate the possibility of inconsistency in the sounds of frequency channels of a multi-band sound reproduction system in this critical area, special acoustic systems are used that reproduce an extended range of mid frequencies. The basis of such an acoustic system is a special mid-frequency dynamic head, which has a slightly smaller diameter than a regular one - about 4-6 inches. This head is installed in a resonator box of a conventional design, but is equipped with a special mid-frequency horn. Thanks to this design, this speaker system combines the advantages of conventional and horn systems, and the upper limit of the mid-frequency band rises to 3 KHz.

The use of dynamic drivers with a titanium diaphragm of a similar design in acoustic systems made it possible to expand the range of the mid-frequency band to the upper limit of the audible range. Such broadband mid-frequency speaker systems make it possible to exclude the high-frequency channel from the multi-band sound reproduction system, but since the power of these systems is low, powerful professional sound reproduction systems still use conventional high-frequency speaker systems to reproduce high frequencies.

Hearing sensitivity in the low-frequency region is exactly as low as it is high in the mid-frequency region. For this reason, very high power is required to achieve a tight, well-felt low-frequency sound. This feature of low-frequency perception is very well illustrated by the human hearing sensitivity curves taken by Fletcher and Munson, which can be found in any good acoustics textbook.

End of work -

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Basic principles of organizing concert complexes. Mixing consoles. Equalizers and their applications. Connection cables and connectors

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All topics in this section:

What is a concert complex
A concert complex is a set of sound systems designed to provide sound in rooms during concert performances. The concert complex includes devices

Concert complexes of medium complexity
With simple systems, everything seems to be clear. Let's now look at a more complex device, for example, one of the concert complexes that are used for scoring clubs, discos or small

Mixing consoles
A mixing console is a device designed to collect electrical signals from all systems of a concert complex - microphones, musical instruments, sound effects and

Sensitivity
This function is sometimes called "input level" or "gain". The sensitivity regulator selects the required gain of the input channel of the mixing console in the range from the output level

Channel equalizer
A channel equalizer is a section of the input channel of a mixing console, designed to adjust the amplitude-frequency response of the channel. The regulators of this section m

Multi-band tone controls
Multiband tone controls, unlike parametric equalizers, do not allow you to change the value of the frequency at which the signal amplitude is adjusted. They only allow you to raise or

Quasi-parametric equalizer
This type of equalizer is a simplified version of the parametric equalizer, from which it differs in the absence of a bandwidth control. Full parametric equalization

Sensitivity switch
The sensitivity switch of the input channel of the mixing console is designed to set the sensitivity of this channel in accordance with the level of the output signal of the source connected to it,

Grouping
Grouping is the grouping of input channels on a mixing console into groups or subgroups. Grouping is only possible on mixing consoles that involve multi-stage

Additional outputs
The system of additional outputs of the mixing console is designed to output a signal from any of its input channels from the console. Through additional outputs these signals, bypassing the main output of the mixing console

Group of controlled additional outputs
The output level of the controlled auxiliary outputs of the mixing console depends on the position of the input channel level controls. By changing the position of the level controls, you can control the balance

Rear panel of the mixing console
On the rear panel of the mixing console there are usually plug connections for connecting the input and output circuits of the console. Each input channel on the rear panel of the console has at least

Graphic equalizer
Graphic equalizer is a multi-band corrector of the amplitude-frequency characteristics of electrical audio signals. The boundaries of the full frequency range in which it operates

Parametric equalizer
The operation of this type of equalizer has already been partially described when describing the principle of operation of a quasi-parametric equalizer for the input channels of mixing consoles. To what has been said it remains to add that

Spectrum Analyzer Applications
As you know, the amplitude-frequency response of a room intended for sound recording must be linear. It should not contain peaks and troughs that could affect the result.

Equalizer settings
The main equalizer of the sound reproduction system is the link between the sound of the sound reproduction system and the sound of the room. Its main function is room sound correction

Practical methods for correcting the amplitude-frequency response of an indoor sound reproduction system
Place the monitoring microphone somewhere in the middle of the room, pointing it towards the stage. Then connect it to one of the channels of the mixing console, set the line x

When setting the main equalizer, place the control microphone slightly away from the axis of symmetry of the hall
The sound characteristics of the main sound reproduction system, taking into account the influence of the room, can be adjusted using a control phonogram. As such a phonogram you can use

Rules to keep in mind when setting up equalizers
1) Make sure the equalizer is turned on and bypass is turned off. 2) Remember that a little more than necessary is already more than necessary. Stop adjusting the strip immediately after the influence

Rolling and laying of connecting cables
Incorrectly folding jumper cables is bound to cause problems sooner or later. According to Murphy's laws, a poorly folded roll at the most inopportune time and at the most inopportune

Laying a multi-wire connecting cable
A multi-wire connecting cable or braid is used to connect external sources and signal receivers with the input and output circuits of the mixing console. The condition of this cable depends

Balanced and unbalanced cables
An unbalanced insulated cable is an ordinary insulated wire placed in a braided shield, also covered with insulation.

Purpose of a symmetrical connection
The main reason why a balanced connection is used is that a balanced line has higher noise immunity than an unbalanced line. Signal amplification, proi

International standards
For three-pin Cannon connectors of the XLR\AXR type, an international standard has been adopted regarding the purpose and numbering of their pins. If the connector is intended for symmetrical connection, then

Rules for handling connecting cables
1) All connections in the concert complex used to transmit sound signals must be symmetrical. An exception can be made only for those circuits whose signals have a high voltage

Crossover
A crossover is a device that divides the input signal spectrum into several frequency ranges. This division corresponds to the frequency bands of acoustic sound reproduction systems. Acoustic

Microphones
Modern microphones well accept all sound components that are necessary to obtain high-quality sound. But at the same time, they also well accept all sound components that are

All these qualities are possessed by most of the dynamic microphones, which do not require additional power sources and have cardioid or supercardioid directional characteristics.

Vocal microphones
When conducting concerts, it is very difficult not to come across a type of microphone such as the Shure SM 58. This microphone, reminiscent in its external shape of ice cream in a waffle cup, represents...

Microphones designed for scoring drum kits
When scoring a drum kit, it is very important to choose microphones for the bass and lead drums, because the sound of these drums determines the character and coherence of the work of the entire rhythm section. Good

Receiving the sound of a piano
In order to accurately convey the sound of a piano, you need to use a large number of microphones, positioning them so that the captured sound most fully corresponds to its purpose in music.

Receiving the sound of brass and saxophone
The sound of brass instruments can be captured using a regular vocal microphone placed directly

Receiving the sound of a flute
Most flutists prefer to use a regular vocal microphone to receive the sound of the flute.

Radio microphones
Radio microphones have a number of positive properties. For example, they do not require a connecting cable, which reduces interference levels. However, they also have peculiar disadvantages.

Matching devices
Direct connection matching devices are designed to match the output and input of two connected devices. Most often, the matching parameters are the input and output resistance of the connection

By simultaneously switching on multiple delay lines, you can create extraordinary volume of sound.
Some tape reverb models have a special input for connecting a remote control pedal. This pedal is designed to stop the movement of the reverb strip during

Tape reverb device
A typical example of a tape reverb is the model of the Japanese company Roland RE - 201. This model can be found quite often, so we will give a fragment from the technical description for this reverb

Rules for working with a digitally controlled digital delay line
The D 1500 digital delay line has 16 banks for storing data - from 0 to 9 and from A to F. Before working with this delay line, you must enter the input and output level controls

Reverberation
The effect of artificial reverb has a very significant difference from the effect produced by a delay line, because reverberation is the sum of a large number of delayed decays

Spring reverb
Spring reverbs are still used in various studios today. Most of them were produced by AKG and Roland, but they were also produced by other companies. Now spring reverbs you

Digital reverb
Nowadays, a wide variety of digital reverb models are produced. They have a wide range of different capabilities, have many specialized sound effects programs,

Digital reverbs with analog control
One of the first analog-controlled digital reverbs was the Yamaha R 1000, which only had four reverb programs. However, it was very convenient to use, which

Special digital reverbs
At the time of its introduction, the Alises Midiverb digital reverb was the cheapest digital reverb that had multi-bank hardware programming. This reverb was produced in a small

Sound effects obtained by using a delay line
Audio delay can create several different audio effects. Delay the signal for a period of time from 1 to 16 milliseconds, produced with a small modulation depth

Reverb sound effects
Reverberation sound effects programs typically reflect conditions in which similar reverberation occurs. For example, “small room”, “large hall”, “soft sheet”, etc. Nevertheless,

Compensation of signal delay in a concert complex
The speed at which sound waves travel in air is approximately 330 m/sec. Therefore, when placing additional sub-sounding acoustic systems in the middle part of a large hall

Simple rules to make working with sound effects easier
1. Before starting work, check that the inputs and outputs of audio processing devices are correctly connected to the additional outputs and inputs of the mixing console. Make sure all audio processing devices are

Compressors and limiters
First, some technical definitions. A compressor is an amplifier with a variable transmission ratio, the value of which decreases with increasing amplitude of the input signal.

Application of compressors and limiters
Compressors and limiters can be used both to process the input signals of a mixing console and to process its various output signals. The composition of the mobile concert complex usually includes

Setting the Noise Limiter
One of the most common applications of noise limiters is in processing the sound of percussion instruments. The noise limiter is connected to the channel of the selected instrument, for example, through connectors

External control input
Many models of noise limiters have an external control input. This input is designed to control the operation of the noise limiter using external audio signals. When connected

Application of exciters
The principles of application and construction of exciters were first defined by the electronic equipment manufacturer Afex. The exciter works by using certain types of hormones.

Control and measuring devices
The most common measuring devices for concert complexes are all kinds of level meters. Most of these meters are designed to control and set relative

Amplifiers
Of all the electronic systems of the concert complex, the maximum load falls on the power amplifier system, the main purpose of which is to convert electrical voltages

Turning power amplifiers on and off. Power amplifiers are always the last to be turned on and the first to be turned off.
When turning on the power to the power amplifiers, you must adhere to the following order: 1. Make sure that all power amplifiers of the audio system are turned off and the volume controls are turned off.

The procedure for eliminating simple faults in power amplifiers
1) Turn off the amplifier and disconnect it from the power supply. Do not touch any parts when the amplifier is turned on, as power supply of electrical circuits and power amplifier blocks is high

Maximum amplification power
In order for an amplifier to produce amplification with a minimum amount of distortion, it must have the largest possible output signal power reserve. This power reserve is usually limited to

Amplifier power and load resistance
The ability of an amplifier to create a signal of a certain power is characterized by the amount of current that the amplifier can create in the load connected to it. In order not to get attached to numbers

Crossovers
A crossover is designed to divide the full spectrum of an audio signal into several frequency bands in a multi-band sound reproduction system. Multi-band sound reproduction system

Passive crossovers
A passive crossover is a set of passive crossover filters whose crossover frequencies are fixedly matched to each other. Most often, passive crossovers are built inside a lot

Advantages created by the use of crossovers
All acoustic systems of a multi-band sound reproduction system are specialized to one degree or another. They reproduce some frequencies well and reproduce much worse or not at all

Cut-off frequency and slope
When setting up a crossover, it is necessary to take into account that the cutoff frequency of any of its bands is not a cutoff in the exact meaning of the word, but only some extreme frequency at which crossover begins.

Additional Crossover Features
Sometimes special horn low-frequency acoustic systems are used to reproduce the lowest frequencies of sound signals. The length of these horns can exceed 2.5 meters. In such a loudspeaker

Sound reproduction system control processors
Control processors for sound reproduction systems are quite complex devices, representing a combination of various crossover systems, equalizers, limiters, delay lines and devices.

Design and principle of operation of dynamic loudspeaker heads
Regardless of the type of driver design, all drivers operate on the same principle. All dynamic heads have a fixed magnet in their design,

The process of burning out the dynamic head coils
The dynamic head coils are wound from thin wire coated with varnish insulation. From prolonged heating, this insulation gradually becomes brittle, crumbles and burns. Because of uh

Bass Horn Speaker Systems
The horns of bass speaker systems are impressive in size. For example, because the length of the sound wave at a frequency of 60 Hz is 5.5 meters, the length of the horn that can influence the direction of this

Multi-way speaker systems
Recently, multi-band acoustic systems have become increasingly used in the practice of operating concert complexes. These systems can reproduce the full or almost full range of frequencies

If the system can only be installed and connected in one single way, it is almost impossible to make a mistake when assembling it
The signal connection in most multi-way speaker systems is made using unbalanced multi-pin connectors, which eliminates the possibility of incorrect connection.

Phasing of dynamic heads of acoustic systems
The dynamic heads in all acoustic systems of the sound reproduction system must be turned on in phase with respect to each other, i.e. the positive terminals of the dynamic heads must be connected

Relationship between electrical power of loudspeaker systems and sound pressure level
The volume of sound emitted by a speaker system is characterized by the sound pressure level, and not by the amount of electrical power of the speaker system. In order to be able to compare

Coordination of acoustic sound reproduction systems
In the simplest case, a high-power acoustic reproduction system can be composed of similar multi-band acoustic systems, each of which has dynamically balanced

Dependence of sound pressure level of a sound reproduction system on distance
When moving away from the sound source, the sound pressure created by it decreases by 4 times, which corresponds to a decrease in the sound pressure level by 6 dB. That. sound reproduction system

Monitor systems
The monitor system is the supporting sound reproducing system of the concert complex. This system is designed to create additional sound in some part of the sounded room.

Inclined monitor speaker systems
Inclined, oblique-shaped monitor speakers are located at the front of the stage opposite the performers whose sound they reproduce. These speakers should

Communication between the main and monitor sound reproduction systems
All possible details of the relationship between the main and monitor systems are discussed in the chapter concerning the layout and assembly of the concert complex. To find out the basic principle of this mutual

Independent monitoring system
The central part of an independent monitoring system is the monitor mixing console. This mixing console is located in close proximity to the main mixing console and is connected to

Monitor system sound mixing
Mixing sound from a monitor system is very different from mixing sound in a room. When mixing sound in the hall, it is necessary to build only one balance, and the monitor system may require up to 16

When moving large weights, try to make the most efficient use of their inertia
When unloading speaker systems from a truck, they should be lifted by hand with the front panel facing down. To prevent a heavy box from slipping out of your hands, it must be supported from below with your fingers. This is pr

System assembly
When assembling the system, you will make fewer mistakes and spend less time if you adhere to a certain sequence of its assembly. For example, assembling a concert complex is better

Procedure for handling damaged and spare connecting cables
All questionable connecting cables should be stored separately in one place for later inspection. For example, you can wind them into one skein by joining or tying their ends together. Behind

Basics of the layout of a concert complex when holding a group concert
To hold a combined concert with the participation of several groups, it is necessary to prepare in advance, taking into account the specifics of the lineups participating in the concert. However, it will be easier to work with different groups

If all microphones and junction box input jacks are labeled, connecting instruments takes less time and attention
To avoid confusion that may arise when you are forced to use the inputs of the stage distribution box inappropriately, it is useful to create a table of correspondence between the numbers of input channels and

Microchannel mixing console
It is extremely difficult to flexibly control a band's sound using an 8-channel mixing console. It can be used successfully if the output signals of some instruments are previously

Ti-channel mixing console
The 12-channel mixing console allows you to more precisely control the drum sound, because... the working space occupied by a drum kit on such a console can be larger than on an 8-channel microphone

Ti-channel mixing console
The 20-channel mixing console provides the widest possibilities for building the sound of a small group, because... the number of its channels exceeds the number of individual instruments in the group. Will distribute

Grouping rules
A minimum of 4 group channels are required to control the monophonic balance of groups of instruments. To carry out the simplest stereo mixing, it is necessary to distribute pairs of

Assembly procedure for the concert complex
In principle, there is no strictly defined order for assembling the concert complex. The only assembly principle that should not be violated is the following. No need to unpack and install additional

Final tuning of the sound of the concert complex
First of all, the final adjustment of the sound of the concert complex should in no case develop into a rehearsal. The purpose of this important operation is to obtain the final sound

Adjusting the sound of percussion instruments
Having placed the microphones of the drum kit in accordance with the intended scheme for obtaining its sound, listen to the signals of each of them separately. Select the required channel sensitivity value,

Setting the Bass Guitar Sound
Before you begin adjusting the sound of the bass guitar channel, you must set the bass guitar channel level control to the position corresponding to 0 dB, and set the bass guitar channel sensitivity control to

Adjusting the Sound of Electronic Keyboards
The native sound of electronic keyboard instruments is designed to be directly connected to a sound reproduction system. However, connecting them directly is not as simple as m

The power supply phase of all electronic devices installed on stage must match the power supply phase of the concert complex equipment
Setting up the channels of keyboard instruments must be done at the maximum level of their output signal, because in this case you will be guaranteed against accidental overload of the mixer input channels

Adjusting the sound of an electric guitar
If the level of noise in the electric guitar channel is not too high, then adjusting its sound is quite simple. Select the channel sensitivity so that its signal is equally strong

Adjusting the Vocal Sound
The correct sound settings of the vocal channels largely determine the sound quality of the entire balance of the sound reproduction system. Vocals should be heard extremely clearly, loudly and cleanly, and be in perfect

Setting up audio processing device channels
Before you begin, make sure that all audio processing devices you will be using are working properly. Check the connections of their outputs and inputs. Jack connectors that

Power supply for the concert complex
The power supply phases of all devices and systems of the concert complex must match. The neutral power supply wires of all devices must be connected to the neutral phase of the power supply network. Completely under

Creating Sound Balance
Once all the equipment is set up and the performers are on stage and ready to play, you can begin mixing the sound. However, in order to carry out this reduction, it is necessary

The relationship between vocals and music
The ratio in which vocals should be present in the overall balance of the work is determined by the function it performs. For example, in simple songs, the vocals should somewhat dominate the music. Ste

Rhythm section balance
The sound of the rhythm section should be smooth and tight. To achieve maximum saturation of the bass drum sound, you need to make sure that it does not hum or sound too dull. If its sound

Checking the balance quality
With prolonged, painstaking listening to the sounds of individual instruments, attention becomes tired, and the ear gradually loses the ability to reliably assess the balance of the overall sound. Therefore it is necessary to

Recording of a concert performance
It’s a good idea to record all concerts with your participation on magnetic tape. Listening to these recordings, you can find many common mistakes that are repeated every concert. Having analyzed

Basic principles of mixing the sound of concerts of independent performers
A sound engineer performing sound mixing at a concert of an independent performer must take into account the specific distribution of the performance load in such a concert. An independent performer is not

Recommendations for mixing sound at a concert
1. when adjusting the sound at a concert, listen carefully to the sound and feel free to make the necessary retuning 2. while pre-setting the balance at the very beginning of the concert, the floor

Insufficient sound volume in the monitor system
Low sound volume from a monitor system is a very serious problem. In the process of work, all sound engineers will inevitably encounter it sooner or later, and sometimes they have to fight with it.

Drum monitor sound volume is insufficient
The drum monitor volume is rarely loud enough. It's very difficult to get a drummer to get into balance with his own monitor system, because that's what the monitor system does.

A special problem for drums
Do you know what words are particularly unpleasant for a sound engineer to hear? No, it's not "no money." It's much more unpleasant to know that the drummer is singing. These words terrify even the most steadfast sound engineers.

Psychoacoustic effect of perception of sound volume of a monitor system
In the process of adjusting the sound of a monitor system, as well as during long musical rehearsals, the auditory attention of people on stage becomes tired, so a constant increase in volume is required.

Troubleshooting technical problems
When the power amplifier's mains fuse blows, all of its electrical units are completely de-energized. The output signal disappears completely, the power indicator does not light up, and the fans turn off.

Reconfiguring equipment for the next concert
If the equipment has retained its settings from previous concerts, setting it up for a new concert is not difficult. In such cases, the sound of the sound reproduction systems is usually

Accelerated sound setup
It is incredibly difficult to immediately adjust the sound of a completely untuned system, especially if you were seated at the console 15 minutes before the start of the performance. The hall is full of noisy people listening

Simple rules for dealing with unexpected situations
- no matter what happens, try to remain calm. Determine the reason, think over a course of action and act boldly and decisively. -- when checking the operation of a complex system, operate the system

Hearing protection
Protect your hearing. The life of a sound engineer depends entirely on his condition. If you're going to be stuck in a noisy truck for six hours, wear headphones for the entire trip. If you

Rules of conduct on stage for vocalists
Do not point the microphone towards monitor speakers.

Final word
In order to work successfully in music production, you must really love your job. You need to have a considerable sense of humor and be able to instantly analyze a lot of details, you need to be able to

As you know, a loudspeaker can be horn-loaded. There are two known modifications of the horn head device. In the first of them, the so-called wide-neck, the throat of the horn is directly adjacent to the diffuser of the head. Due to the fact that the mouth has a diameter larger than the diameter of the head diffuser, the directionality of such a horn is sharper than the directionality of the head. Therefore, sound energy is concentrated on the horn axis and the sound pressure increases here.

In the second modification (narrow-neck), the horn is connected to the diaphragm (diffuser) of the head through a pre-horn chamber, which plays a role similar to that of an electrical matching transformer. Here the mechanical resistance of the moving system of the head and the throat of the horn is consistent, which increases the load on the diaphragm and, as it were, increases its radiation resistance, due to which the efficiency greatly increases. Thus, this makes it possible to obtain high sound pressure.

There are many different types of horns, but practically the most often used in household equipment is an exponential horn, the cross section of which varies according to the law:

S = S 0 ∙ e βx ,

Where S 0 – area of ​​the horn inlet,

β – exponent index.

In Fig. 1 shows various horn profiles:

As can be deduced from the formula above, the cross-section of such a horn increases by the same percentage for each unit of its axial length. The value of this percentage increment determines the lower limit frequency of the horn. In Fig. Figure 2 shows the dependence of the percentage increment of the cross section per 1 cm of axial length on the lower limit frequency. So, for example, to ensure that the horn reproduces the lower limit frequency of 60 Hz, the cross-sectional area must increase by 2% for every 1 cm of its axial length. This dependence can also be represented in the form of the following expression:

f UAH = 6,25 ∙ 10 3 ∙ lg (0,01 k + 1)

Where k – increment of cross-sectional area, %.

For low frequencies (up to 500 Hz), this expression is simplified and takes the form: f UAH = 27k

If the horn is made of a square or circular cross-section, then the side of the square or the diameter of the circle should increase for every 1 cm of the length of the horn by √ k percent. If it is made of a rectangular cross-section with a constant height, then the width of the horn section should increase byk percent for every 1 cm of its length.

However, maintaining the required percentage increase in the cross-section is not yet sufficient for good reproduction of low frequencies. It is necessary to have a sufficient area of ​​its outlet - the mouth. Its diameter (or the diameter of an equal circle) should be:

D ≥ λ UAH / ∏ ≈ 110/f gr.n

Thus, for a lower cutoff frequency of 60 Hz, the diameter of the mouth will be about 1.8 m. For lower cutoff frequencies, the size of the mouth will be even larger. In addition, the horn head, while reproducing low frequencies well (abovef UAH ), does not reproduce a wide frequency range well enough. Given this, it is advisable to have two horn heads: one for reproducing low frequencies and the other for high frequencies. In Fig. Figure 3 shows the appearance and cross-section of such a speaker with two horn heads and a bass reflex for reproducing frequencies lowerf UAH mouthpiece

The use of low-frequency horn designs in residential premises is limited by the size of the room. However, if such a possibility exists, then the calculation of the horn should begin by specifying the area of ​​the mouth at the selected lower limit frequency, reducing the cross-section by percent for every 1 cm of axial length until a cross-sectional area equal to the area of ​​the head diffuser is reached. At the same time, in order to mate the head with a wide-necked horn, the horn must have a cross-section of the same shape, i.e. round or elliptical. For narrow-neck horns, the identity of the cross-sectional shape and the diaphragm of the head is not necessary, since the throat and diaphragm are articulated through the pre-horn chamber. Note that the height of the chamber must be significantly greater than the amplitude of oscillations of the moving system of the head in order to avoid the occurrence of strong nonlinear distortions due to the asymmetry of the deformation of the air volume in the chamber. However, if the pre-horn height is too high, high-frequency reproduction is impaired.

Sometimes, in order to reduce the overall dimensions of speakers, rolled horns are used, the various designs of which are shown in Fig. 4. Rolled horns are calculated in almost the same way as regular ones. When calculating the profile, it is necessary to ensure that at the transition points (knee bends) there are no sudden changes in sections that cause irregularities in the frequency response.

A horn of limited length has resonant properties. As a result, the active component of the horn's input impedance depends in a complex way on frequency, creating uneven sensitivity of the loudspeaker. The unevenness of the frequency response of the horn impedance decreases if the diameter of the horn mouth is approximately Let us recall the basic relationships between the parameters of an exponential horn:

If it is necessary to emit sound with a frequency of 100 Hz, then the critical frequency should be selected below 100 Hz, for example, 60 Hz. Then

For transmitting high frequencies and the ability to create a sufficiently large transformation ratio of the pre-horn chamber

Rice. 4.40. Loudspeaker with folded horn

a throat diameter of no more than 2 cm will be required. Then: Thus, to transmit low frequencies with a horn loudspeaker, starting from 100 Hz, a horn with a diameter of about a meter and a length of more than one and a half meters is required. If transmission of even lower frequencies is necessary, then the dimensions must be even larger. Therefore, they resort to “folding” the horn in order to at least reduce its length. Such labyrinth horns are used quite widely, for various frequency ranges. The horn diagram is shown in Fig. 4.40.

8.3. Horn loudspeakers.

One of the most common types of audio equipment widely used nowadays is horn loudspeakers.According to GOST 16122-87, a horn loudspeaker is defined as “a loudspeaker whose acoustic design is a rigid horn.” Thus, a horn can be considered a full-fledged acoustic design along with those discussed earlier in section 8.2.3. The ability of horns to amplify and direct sound in the desired direction (long used in the creation of musical instruments) led to the fact that horn loudspeakers began to be used from the very beginning of the development of electrical engineering, they appeared even earlier than diffuser loudspeakers.

However, the creation of a real horn loudspeaker with a design very close to the modern one begins in 1927, when famous engineers from Bell laboratories (USA) A.Thuras and D.Wente developed and patented a “compression horn emitter” the next year. An electromagnetic transducer with a frameless coil made of aluminum tape wound on an edge was used as a loudspeaker (driver). The driver diaphragm was made from a downward facing aluminum dome. Even then, both the pre-horn camera and the so-called Wente body were used (we will talk about them in more detail later). The first commercially produced model 555/55W (form. "Western Electric") was widely used in cinemas in the 30s.

A significant step towards expanding the range towards low frequencies was the invention of P. Voigt (England), where it was first proposed to use “folded” horns, which are widely used today. The first complex designs of curled low-frequency horns for high-quality acoustic systems were developed by Paul Klipsh in 1941 and were called Klipschhorn. Based on this design with a horn design, the company still produces high-quality acoustic systems.

It should be noted that in Russia the first samples of horn loudspeakers were created in 1929 (engineers A.A. Kharkevich and K.A. Lomagin). Already in 1930-31, powerful horn loudspeakers up to 100 W were developed for sounding Red and Palace Squares.

Currently, the scope of application of horn loudspeakers is extremely wide, including sound systems for streets, stadiums, squares, sound reinforcement systems in various rooms, studio monitors, portal systems, high-quality household systems, public address systems, etc.

Causes The spread of horn loudspeakers is due primarily to the fact that they are more efficient, their efficiency is 10%-20% or more (in conventional loudspeakers the efficiency is less than 1-2%); In addition, the use of rigid horns allows the formation of a given directivity characteristic, which is very important when designing sound reinforcement systems.

How they work First of all, the horn loudspeaker (RG) is an acoustic impedance transformer. One of the reasons for the low efficiency of direct radiation GG is the large difference in density between the diaphragm material and the air, and therefore the low resistance (impedance) of the air medium to vibrations of the loudspeaker. A horn loudspeaker (through the use of a horn and a pre-horn chamber) creates additional load on the diaphragm, which provides better impedance matching conditions and thereby increases the radiated acoustic power. This makes it possible to obtain a large dynamic range, lower nonlinear distortion, better transient distortion and provide less load on the amplifier. However, when using horn loudspeakers, specific problems arise: to emit low frequencies, it is necessary to significantly increase the size of the horn; in addition, high sound pressure levels in a small pre-horn chamber create additional nonlinear distortions, etc.

Classification: horn loudspeakers can be divided into two large classes - wide-necked and narrow-necked. Narrow-neck RGs consist of a specially designed dome loudspeaker called a driver, a horn, and a pre-horn chamber (often with an additional insert called a phase shifter or Wente body). Wide-neck RGs use conventional high-power dynamic direct-radiation loudspeaker heads and a horn whose throat diameter is equal to the diameter of the head.

In addition, they can be classified according to the shape of the horn: exponential, convoluted, multi-cell, bipolar, radial, etc. Finally, they can be divided into frequency domain playback: low-frequency (usually collapsed), mid- and high-frequency, as well as Areas of use in official communications (for example, megaphones), in concert and theater equipment (for example, in portal systems), in sound systems, etc.

Device Basics: The main elements of a narrow-neck horn loudspeaker, shown in Fig. 8.32, include: a horn, a pre-horn chamber and a driver.

Horn - is a pipe of variable cross-section on which the driver is loaded. As noted above, it is one of the types of acoustic design. Without decoration, the loudspeaker cannot emit low frequencies due to the short circuit effect. When installing a loudspeaker in an infinity screen or other type of design, the acoustic power emitted by it depends on the active component of the radiation resistance Cancer=1/2v 2 Rizl. The reactive component of the radiation resistance determines only the added mass of air. At low frequencies, when the wavelength is greater than the size of the emitter, a spherical wave propagates around it, while at low frequencies the radiation is small, the reactance predominates, as the frequency increases, the active resistance increases, which in the spherical wave equals Rizl=  cS(ka) 2 /2 (in a plane wave it is greater and equal Rizl=  WithS),S is the area of ​​the emitter, a is its radius, k is the wave number. A special feature of a spherical wave is that the pressure in it drops quite quickly in proportion to the distance p~1/r. It is possible to provide radiation at low frequencies (i.e. eliminate the short circuit effect) and bring the waveform closer to a flat one if the emitter is placed in a pipe whose cross-section increases gradually. This pipe is called mouthpiece

The entrance hole of the horn in which the emitter is located is called throat, and the output hole emitting sound into the environment is mouth. Since the horn must increase the load on the diaphragm, the throat must have a small radius (area) for effective energy transformation to occur. But at the same time it must have a sufficiently large diameter of the mouth, because in narrow pipes, where the wavelength is greater than the radius of the outlet -a-, (i.e. the condition >8a is met), most of the energy is reflected back, creating standing waves, this phenomenon is used in musical wind instruments. If the pipe opening becomes larger (<a/3),то Rизл приближается к сопротивлению воздушной среды и волна беспрепятственно излучается в окружающее пространство устьем рупора.

Generator shape the horn must be chosen in such a way as to reduce the "spreading" of energy, i.e. rapid decrease in sound pressure, therefore, transform the spherical shape of the wave front so that it approaches a plane wave, which increases the radiation resistance (in a plane wave it is higher than in a spherical wave) and reduces the rate of pressure decrease; in addition, the choice of the shape of the generatrix makes it possible to concentrate sound energy in a given angle, i.e., it forms the directivity characteristic.

Thus, the horn should have a small throat size, and the cross-section at the throat should slowly increase, while the size of the mouth should be increased. In order for large mouth sizes to be achieved with an acceptable axial length of the horn, the rate of increase in the horn cross-section must increase as the cross-sectional area increases (Fig. 8.33). This requirement is met, for example, by the exponential shape of the horn:

Sx=S 0 e  x , (8.2)

where So is the cross-section of the horn throat; Sx is the cross section of the horn at an arbitrary distance x from the throat;  is an indicator of horn expansion. The unit of  is 1/m. The horn expansion index is a value measured by the change in the horn cross-section per unit of its axial length. An exponential horn is shown in Fig. 2, where it is shown that the axial length of the horn dL corresponds to a constant relative change in the cross section. Analysis of the wave processes occurring in an exponential horn shows that the radiation resistance to which the radiator is loaded depends on frequency (Fig. 8.34). It follows from the graph that in an exponential horn the wave process is possible only if the oscillation frequency of the emitter exceeds a certain frequency called critical(fcr). Below the critical frequency, the active component of the radiation resistance of the horn is zero, the resistance is purely reactive and equal to the inertial resistance of the air mass in the horn. Starting from a certain frequency, which is approximately 40% higher than the critical one, the active resistance of radiation exceeds the reactive resistance, so the radiation becomes quite effective. As follows from the graph in Fig. 8.34, at frequencies more than four times higher than the critical frequency, the radiation resistance remains constant. The critical frequency depends on the horn expansion ratio as follows:  cr= s/2, Where With - sound speed. (8.3)

If the speed of sound in air at a temperature of 20 degrees is 340 m/sec, you can obtain the following relationship between the horn expansion indicator  and critical frequency f cr (Hz):  ~0.037f cr.

Not only the value of the critical frequency of the horn, and therefore the frequency response of the radiation resistance, but also the dimensions of the horn depend on the horn expansion index. The axial length of the horn can be determined from formula (1) at x=L as:

L=1/  ln S l /S 0 (8.4)

From expression (3) we can draw the following conclusion: since in order to reduce the critical frequency of the horn, the horn expansion index (2) should be reduced, the axial length of the horn L should thereby increase. This dependence is the main problem with the use of horn loudspeakers in high-quality loudspeaker systems and is the reason for the use of "rolled" horns. It should be pointed out that when constructing a graph of the radiation resistance of an exponential horn (Fig. 8.36), the reflection of waves from the mouth into the horn, which always partially occurs for horns of finite length, is not taken into account. The resulting standing waves create some fluctuations in the radiation resistance values. Reflection of sound from the mouth of the horn occurs only in the low-frequency region. As the frequency increases, the acoustic properties of the media (in the horn and outside the horn) are leveled out, sound is not reflected into the horn, and the input acoustic impedance of the horn remains almost constant.

Pre-horn camera: Since the radiated acoustic power of a loudspeaker depends on the active resistance of radiation and the oscillatory speed of the emitter, to increase it in narrow-necked horn loudspeakers, the principle of acoustic transformation of forces and velocities is used, for which the dimensions of the throat of horn 2 are reduced several times in comparison with the dimensions of emitter 1 (Fig. 8.35). The resulting volume between the diaphragm and the throat of horn 3 is called the pre-horn chamber. We can conditionally imagine the situation in the pre-horn chamber as oscillations of a piston loaded on a wide pipe with area S 1, which turns into a narrow pipe S 0 (Fig. 8.35). If the piston diaphragm were loaded only on a wide pipe with an area equal to the area of ​​the diaphragm (wide-neck horn) , then its radiation resistance would be equal Rizl= WithS 1 , and the acoustic power emitted by it would be approximately equal to Ra= 1/2R izl v 1 2 =1/2  WithS 1 v 1 2 (these relations are strictly satisfied only for a plane wave, but can be applied in this case under certain assumptions.) When installing the diaphragm in the pre-horn chamber, i.e. when it is loaded onto the second pipe with a narrow inlet, additional resistance (impedance) to the vibrations of the diaphragm arises (due to the reflected wave arising at the junction of the two pipes). The value of this impedance is Z L (referred to the point of entry into the second pipe, i.e. at x = L ) can be determined from the following considerations: if we assume that the air in the pre-horn chamber is incompressible, then the pressure p created in the chamber under the action of force F 1 on a piston (diaphragm) with area S 1, is transmitted to the air in the throat of the horn and determines the force F 0 , acting in the throat of a mouthpiece with an area S 0 :

p=F 1 /S 1 , F 0 =pS 0 (8.5).

From this we obtain the following relations: F 1 /S 1 =F 0 /S 0 , F 1 /F 0 =S 1 /S 0 . The ratio of the emitter area to the horn throat area S 1 / S 0 is called acoustic transformation coefficient and is designated P. Therefore, the relationship of forces can be represented as: F 1 =nF 0 . From the condition of equality of the volumetric velocities of the diaphragm and air at the mouth of the horn (i.e., from the condition of maintaining the volume of air displaced by the diaphragm during displacements from the pre-horn chamber), the following relations are obtained: S 1 v 1 = S 0 v 0 or: v 0 /v 1 =S 1 /S 0 =n. (8.6).

The obtained relationships allow us to draw the following conclusion: the diaphragm, under the influence of a greater force (F 1 > F 0), oscillates at a lower speed (V 1<. V 0), значит, она испытывает большее сопротивление среды при колебаниях. Значение Z L в таком случае (учитывая, что импеданс по определению есть отношение силы к скорости колебаний Z L =F 1 /v 1) будут равны с учетом соотношений (8.5)и (8.6): Z L =F 1 /v 1 =S 1 p/v 1 =S 1 p/{v 0 S 0 /S 1 }=(S 1 2 /S 0 2)S 0 p/v 0 . (8.7)

If the piston stood at the inlet of a narrow pipe, then its resistance would be equal to Rizl=cS 0, while by definition Rizl=F 0 /v 0 =S 0 p/v 0, i.e. S 0 p/v 0 =сS 0 , substituting this expression into formula (8.7) we get:

Z L =(S 1 2 /S 0 2 )S 0  With=(S 1 /S 0 ) S 1  With. (8.8)

This multiplication of impedance сS 0 by a coefficient (S 1 2 /S 0 2 ) is equivalent to the use of some kind of step-down transformer, as can be seen in the corresponding equivalent electrical circuit (Fig. 8.37)

Therefore, if, in the presence of additional resistance, the radiated acoustic power increases and is equal to:

Ra=1/2 cZ L =1/2  WithS 1 v 1 2 (S 1 /S 0 ). (8.9)

Thus, the use of acoustic transformation due to the pre-horn chamber makes it possible to increase the acoustic power by (S 1 / S 0) times, which significantly increases the operating efficiency of the horn loudspeaker. The value of the acoustic transformation coefficient is limited, since it depends on the area of ​​the emitter (S 1) and the area of ​​the horn throat (So). An increase in the emitter area is associated with an increase in its mass. A large mass emitter has a high inertial resistance at high frequencies, which becomes comparable to the radiation resistance. As a result, at high frequencies the oscillatory speed decreases, and therefore the acoustic power. The acoustic transformation coefficient increases as the horn throat area decreases, but this is also acceptable within certain limits, because leads to an increase in nonlinear distortions. Typically, the acoustic transformation coefficient is chosen to be about 15-20.

The efficiency of a horn loudspeaker can be approximately estimated using the formula: Efficiency=2R E R ET /(R E +R ET ) 2 x100%, (8.10)

where R E is the active resistance of the voice coil, R ET =S 0 (BL) 2 /cS 1 2, where B is the induction in the gap, L is the length of the conductor. The maximum efficiency of 50% is achieved when R E = R ET, which cannot be achieved in practice.

Nonlinear distortions in horn GGs are determined both by ordinary reasons that arise in loudspeaker heads: nonlinear interaction of the voice coil with the magnetic field, nonlinear flexibility of the suspension, etc., and by special reasons, namely high pressure in the throat of the horn, and thermodynamic effects begin to affect, as well as nonlinear air compression in the pre-horn chamber.

Emitter, which is used for horn loudspeakers is a conventional electrodynamic loudspeaker. For wide-neck horns (without a pre-horn chamber) this is a powerful low-frequency loudspeaker. Wide-neck horns are now used as low-frequency design in a number of designs of acoustic units, for example Genelek (this technology is called waveguide TL), portal sound systems, etc.

Narrow throat horn loudspeakers use special types of electrodynamic loudspeakers (commonly called drivers An example of the design is shown in Fig. 8.32. As a rule, they have a dome diaphragm made of hard materials (titanium, beryllium, aluminum foil, impregnated fiberglass, etc.), made together with a suspension (sinusoidal or tangential corrugation). A voice coil is attached to the outer edge of the diaphragm (frame made of aluminum foil or rigid types of paper with two or four layers of winding). The suspension is secured with a special ring on the upper flange of the magnetic circuit. An anti-interference liner (Wente body) is installed above the diaphragm - acoustic lens to align the phase shifts of acoustic waves emitted by different parts of the diaphragm. Some high-frequency models use special annular diaphragms.

To analyze the operation of horn loudspeakers in the low frequency region, the method of electromechanical analogies is used. Calculation methods mainly use the Thiele-Small theory, on which the calculation methods for conventional cone loudspeakers are based. In particular, measuring the Thiele-Small parameters for the driver allows one to evaluate the shape of the frequency response for low-frequency horn loudspeakers. Figure 8.37 shows the shape of the frequency response, where the inflection frequencies of the curve are determined as follows:f LC =(Q ts)f s /2; f HM = 2f s / Q ts ; f HVC =R e / L e ; f HC =(2Q ts)f s V as /V fs ;where Q ts is the overall quality factor; f s \resonant frequency of the emitter; R e , L e – resistance and inductance of the voice coil, V fs – equivalent volume, V as – volume of the pre-horn chamber.

A complete calculation of the structure of the sound field emitted by horn loudspeakers, including taking into account nonlinear processes, is carried out using numerical methods (FEM or BEM), for example, using software packages: http://www.sonicdesign.se/ ;http://www.users.bigpond.com/dmcbean/ ;http:/melhuish.org/audio/horn.htm.

Since one of the main tasks of horn loudspeakers is the formation of a given directivity characteristic, which is of fundamental importance for sound systems for various purposes, a wide variety of horn shapes, the main ones being the following:

= exponential horn, most horn loudspeakers for sounding open spaces are made with it, for example, domestic models 50GRD9, 100GRD-1, etc.;

=sectional horns that were designed to combat the exacerbation of directivity characteristics at high frequencies (Fig. 8.38). A sectional horn consists of a number of small horns connected together by throats and mouths. In this case, their axes turn out to be fanned out in space, although the directionality of each cell becomes sharper with frequency, the overall directionality of the group emitter remains wide.

=radial the horn has different curvature along different axes (Fig. 8.39a, b). The width of the radiation pattern is shown in Fig. 8.43b, from which it can be seen that in the horizontal plane it is almost constant, in the vertical area it decreases. These types of horns are used in modern studio monitors, in addition, they are used in cinema systems.

To expand the directivity characteristics in horn loudspeakers, they are also used acoustic dissipative lenses (Fig. 8.40).

=diffraction the horn (Fig. 8.41a, b) has a narrow opening in one plane and a wide opening in the other. In a narrow plane it has a wide and almost constant radiation pattern, in a vertical plane it is narrower. Variants of such horns are widely used in modern sound reinforcement technology.

Horn uniform coverage(after a number of years of research were created at JBL), they allow you to control the directivity characteristics in both planes (Fig. 8.42a, c).

Special shape folded horns used to create low-frequency emitters Fig. 8.43. The first cinema systems with a folded horn for cinema were created back in the 30s. Rolled horns in both narrow-neck and wide-neck loudspeakers are now widely used for high-quality control units, for powerful acoustic systems in concert and theater equipment, etc.

There are currently other types of horns in production, both for sound reinforcement equipment and for household audio equipment. In the practice of scoring large concert halls, discos, stadiums, etc., suspended sets of horn loudspeakers called clusters.