Accurate diagnosis. Guitar speakers - purpose, parameters, configuration (Vol.1)

Measurement of the frequency response of acoustic systems at home.

Acoustics for testing:
floor standing Tannoy Turnberry GR LE,
Center channel speakers Tannoy Revolution XT Center,
Shelf speakers Canton Vento 830.2,
Wall speakers Canton Ergo 610.


Microphone placement.






Block diagram of connection for measuring the amplitude-frequency response (AFC).


The following devices were used for the measurement:
1. Measuring microphone Behringer ECM8000
2. External sound card Tascam US-4x4
3. PC Acer V5-572G, DELL INSPIRON 5010
4. Balanced cable XLR-XLR (5m)
5. Two Inakusik Premium cables MiniJack - 2 RCA and MiniJack-MiniJack with 6.3mm adapter (for sound card calibration)
6. Software Room EQ Wizard 5.19(REW).
The Yamaha RX-A3060 AV receiver is set to Pure Direct mode.
All acoustic systems for the initial measurements were connected in turn to the output terminals of the front channels.
Before starting measurements, it is necessary to perform calibration measurements of the sound card. To do this, the output from the PC sound card and the Jack input of the external sound card are connected.
To calibrate the level, you will also need a sound level meter, however, our measurements were made with relative reference to the level, since the entire set of measurements was carried out in order to further adjust the frequency response with the receiver's parametric equalizer and it was necessary to obtain data on its unevenness.
For more accurate measurements, it is also advisable to calibrate the microphone in a special laboratory or use a microphone that is already supplied with a calibration file. For the used models based on the Behringer ECM8000, the frequency response deviations are extremely small, especially in the low and medium frequencies.
Initial measurements(no level binding) .
Pure Direct mode.
Characteristics of the PC sound card Acer Aspire V5-572. Frequency response of the Tannoy Revolution XT Center speaker system.

Frequency response of Tannoy Turnberry GR LE frontal systems in the near field.



Frequency response of Surround Canton Vento 830.2 channels in the near field (smoothing 1/12 and 1/6).



Frequency response of front presence channels and rear presence channels, Canton Ergo 610.

Other applied measurements.
Canton Vento 830.2. Open and closed phase inverter port. Influence of grids in the near field.



Influence of metal meshes in Canton Ergo 610 and massive fabric meshes in Tannoy Turnberry GR LE (at 20cm and 1m).



Frequency response of Tannoy Turnberry GR LE (left and right channel). Change in frequency response at the listening point when switching the HF controller (+3dB) on the speakers.

We continue our tradition and publish another article from the "testing methodology" series. Articles like this serve both as a general theoretical basis to help readers get an introduction to the topic, and as specific guidance for interpreting test results obtained in our laboratory. Today's article on the methodology will be somewhat unusual - we decided to devote a significant part of it to the theory of sound and acoustic systems. Why is this needed? The fact is that sound and acoustics are practically the most difficult of all topics covered by our resource. And, perhaps, the average reader is less savvy in this area than, say, in assessing the overclocking potential of various Core 2 Duo steppings. We hope that the reference materials that formed the basis of the article, as well as a direct description of the measurement and testing methodology, will fill in some gaps in the knowledge of all lovers of good sound. So, let's start with the basic terms and concepts that any novice audiophile must know.

Basic terms and concepts

A little introduction to music

Let's start in an original way: from the beginning. From what sounds through the speakers, and about other headphones. It just so happened that the average human ear distinguishes signals in the range from 20 to 20,000 Hz (or 20 kHz). This fairly solid range, in turn, is usually divided into 10 octaves(you can divide by any other number, but 10 is accepted).

In general octave is the frequency range whose limits are calculated by doubling or halving the frequency. The lower limit of the next octave is obtained by doubling the lower limit of the previous octave. Anyone familiar with Boolean algebra will find this series strangely familiar. Powers of 2 with added zero at the end in pure form. Actually, why do you need knowledge of octaves? It is necessary in order to stop confusion about what should be called lower, middle or some other bass and the like. The generally accepted set of octaves uniquely determines who is who to the nearest hertz.

Octave number	Lower limit, Hz	Upper limit, Hz	Name	Title 2
			deep bass
			Medium bass	Subcounter
			upper bass
			lower middle
			Actually the middle
			Upper middle
			Lower top
			Medium top
			Upper high
			Upper octave

The last line is not numbered. This is due to the fact that it is not included in the standard ten octaves. Pay attention to the column "Name 2". It contains the names of octaves that are distinguished by musicians. These "weird" people have no concept of deep bass, but there is one octave above - from 20480 Hz. Therefore, such a discrepancy in the numbering and names.

Now we can talk more specifically about the frequency range of acoustic systems. We should start with some bad news: there is no deep bass in multimedia acoustics. The vast majority of music lovers at -3 dB have simply never heard 20 Hz. And now the news is pleasant and unexpected. In a real signal, there are no such frequencies either (with some exceptions, of course). An exception is, for example, a recording from the IASCA Competition referee disc. The song is called "The Viking". There, even 10 Hz are recorded with a decent amplitude. This track was recorded in a special room on a huge organ. The system, which will play the Vikings, the judges hung with awards, like a Christmas tree with toys. And with a real signal, everything is simpler: a bass drum - from 40 Hz. Hefty Chinese drums - also from 40 Hz (there is, however, one mega-drum among them. So it starts playing from 30 Hz). Live double bass - generally from 60 Hz. As you can see, 20 Hz is not mentioned here. Therefore, you can not be upset about the absence of such low components. They are not needed to listen to real music.

The figure shows the spectrogram. There are two curves on it: purple DIN and green (from old age) IEC. These curves represent the spectrum distribution of the average musical signal. The IEC characteristic was used until the 60s of the 20th century. In those days, they preferred not to mock the squeaker. And after the 60s, experts drew attention to the fact that the preferences of listeners and music have changed somewhat. This was reflected in the standard of the great and mighty DIN. As you can see, there are much more high frequencies. But the bass did not increase. Conclusion: no need to chase super-bass systems. Moreover, the desired 20 Hz was not put in the box anyway.

Speaker Specifications

Now, knowing the ABC of octaves and music, you can begin to understand the frequency response. AFC (frequency response) - dependence of the oscillation amplitude at the output of the device on the frequency of the input harmonic signal. That is, the system is fed a signal at the input, the level of which is taken as 0 dB. From this signal, speakers with an amplifying path do what they can. It turns out they usually do not have a straight line at 0 dB, but in some way a broken line. The most interesting thing, by the way, is that everyone (from audio amateurs to audio manufacturers) strives for a perfectly flat frequency response, but they are afraid to "aspire".

Actually, what is the use of the frequency response and why do the authors of TECHLABS with enviable constancy try to measure this curve? The fact is that it can be used to establish real, and not whispered by the "evil marketing spirit" to the manufacturer, the boundaries of the frequency range. It is customary to indicate at what signal drop the cutoff frequencies are still played. If not specified, it is assumed that the standard -3 dB has been taken. This is where the trick lies. It is enough not to indicate at what drop the border values ​​​​were taken, and you can absolutely honestly indicate at least 20 Hz - 20 kHz, although, indeed, these 20 Hz are achievable at a signal level that is very different from the prescribed -3.

Also, the benefit of the frequency response is expressed in the fact that, although approximately, it is possible to understand what problems the selected system will have. And the system as a whole. The frequency response suffers from all elements of the tract. To understand how the system will sound according to the schedule, you need to know the elements of psychoacoustics. In short, the situation is as follows: a person speaks within medium frequencies. Therefore, he perceives them the best. And on the corresponding octaves, the graph should be the most even, since distortions in this area put a lot of pressure on the ears. It is also undesirable to have high narrow peaks. The general rule here is that peaks are heard better than troughs, and a sharp peak is heard better than a flat one. We will dwell on this parameter in more detail when we consider the process of its measurement.

Phase response (PFC) shows the change in phase of the harmonic signal reproduced by the speaker depending on the frequency. It can be unambiguously calculated from the frequency response using the Hilbert transform. The ideal PFC, which says that the system has no phase-frequency distortion, is a straight line passing through the origin. Acoustics with such a phase response is called phase-linear. For a long time, this characteristic was ignored, since there was an opinion that a person is not susceptible to phase-frequency distortions. Now they measure and indicate in the passports of expensive systems.

Cumulative spectrum attenuation (CCD) - a set of axial frequency response (frequency response measured on the acoustic axis of the system), obtained with a certain time interval during the attenuation of a single pulse and reflected on one three-dimensional graph. Thus, according to the graph of the QLC, it is possible to say exactly which regions of the spectrum will decay at what rate after the pulse, that is, the graph allows you to identify delayed resonances of the speakers.

If the GLC has a lot of resonances after the upper middle, then such acoustics will subjectively sound "dirty", "with sand on the HF", etc.

AC impedance - this is the total electrical resistance of the AC, including the resistance of the filter elements (complex value). This resistance contains not only active resistance, but also reactances of capacitances and inductances. Since the reactance depends on the frequency, the impedance is also completely subordinate to it.

If one speaks of impedance as a numerical quantity completely devoid of complexity, then one speaks about its modulus.

Three-dimensional impedance graph (amplitude-phase-frequency). Usually, its projections on the amplitude-frequency and phase-frequency planes are considered. If you combine these two plots, you get the Bode plot. And the amplitude-phase projection is a Nyquist plot.

Given that the impedance is frequency dependent and not constant, it can be easily used to determine the complexity of the acoustics for the amplifier. Also, according to the schedule, you can tell what kind of acoustics it is (ZYa - a closed box), FI (with a phase inverter), how individual sections of the range will be reproduced.

Sensitivity - see Thiel-Small parameters.

Coherence - coordinated flow of several oscillatory or wave processes in time. It means that the signal from different GG acoustic systems will come to the listener at the same time, that is, it indicates the safety of phase information.

The meaning of the listening room

The listening room (often abbreviated to KdP among audiophiles), and its conditions are extremely important. Some put KDP in first place in importance, and only after it - acoustics, amplifier, source. This is somewhat justified, since the room is capable of doing anything with the graphs and parameters measured by the microphone. There may be peaks or dips in the frequency response that were not present in the measurements in the anechoic room. The PFC will also change (following the frequency response) and the transient characteristics. In order to understand where such changes come from, it is necessary to introduce the concept of room modes.

Room mods are beautifully named room resonances. The sound is emitted by the speaker system in all directions. Sound waves bounce off everything in the room. In general, the behavior of sound in a single listening room (LL) is completely unpredictable. There are, of course, calculations that allow us to evaluate the effect of various modes on sound. But they exist for an empty room with an idealized finish. Therefore, it is not worth bringing them here, they have no practical value in domestic conditions.

However, it is necessary to know that resonances and the reasons for their appearance directly depend on the frequency of the signal. For example, low frequencies excite room modes, which are determined by the size of the CDP. The boominess of the bass (resonance at 35-100 Hz) is a clear representative of the appearance of resonances in response to a low frequency signal in a standard room of 16-20 m 2. High frequencies give rise to slightly different problems: diffraction and interference of sound waves appear, which make the directional characteristic of the speaker frequency dependent. That is, the directivity of the speaker becomes narrower with increasing frequency. It follows from this that the listener will receive maximum comfort at the intersection of the acoustic axes of the speakers. And only he. All other points in space will receive less information or receive it distorted in one way or another.

The influence of the room on the speakers can be significantly reduced by dampening the CDP. To do this, various sound-absorbing materials are used - from thick curtains and carpets to special plates and tricky wall and ceiling configurations. The quieter the room, the more the speaker contributes to the sound, and not the reflections from your favorite computer desk and a pot of geraniums.

Recipes for arranging speakers in a room

Vandersteen recommends placing loudspeakers along the longest wall of the room at points where low frequency modes are least likely to occur. You need to draw a plan of the room. On the plan, divide a long wall into three, five, seven and nine parts in sequence, draw the corresponding lines perpendicular to this wall. Do the same with the side wall. The points of intersection of these lines will indicate those places where the excitation of low frequencies in the room is minimal.

Lack of bass, lack of tight and clear bass:

try moving the speakers closer to the back wall;

check if the speaker stands are stable: if necessary, use spikes or conical legs;

check how solid the wall behind the speakers is. If the wall is flimsy and "sounds", put the speakers in front of a powerful (capital) wall.

The stereo image does not go beyond the space limited by the speakers:

move the speakers closer to each other.

There is no depth of sound space. There is no clear sound image in the center between the speakers:

choose the optimal height of the speakers (use stands) and your listening position.

Harsh annoying sound in the mid and high frequencies:

if the speakers are new, warm them up on a music signal for a few days;

check for strong reflections from the side walls or from the floor in front of the listener.

distortion

It is necessary to move from subjectivism to technical concepts. Let's start with distortion. They are divided into two large groups: linear and non-linear distortions. Linear distortion do not create new spectral components of the signal, only change the amplitude and phase components. (They distort the frequency response and phase response, respectively.) Nonlinear distortion make changes to the spectrum of the signal. Their number in the signal is presented in the form of coefficients of non-linear distortion and intermodulation distortion.

THD (THD, THD - total harmonic distortion) is an indicator that characterizes the degree to which the voltage or current waveform differs from the ideal sinusoidal waveform. In Russian: a sine wave is applied to the input. At the output, it does not look like itself, since the path introduces changes in the form of additional harmonics. The degree of difference between the signal at the input and at the output is reflected by this coefficient.

Intermodulation distortion factor - this is a manifestation of amplitude nonlinearity, expressed in the form of modulation products that appear when a signal is applied, consisting of signals with frequencies f1 and f2(based on the IEC 268-5 recommendation, frequencies are taken for measurements f 1 and f 2, such that f 1 < f 2/8. You can take another ratio between frequencies). Intermodulation distortion is quantified by spectral components with frequencies f2±(n-1) f1, where n=2.3,… At the output of the system, the number of extra harmonics is compared and what percentage of the spectrum they occupy is estimated. The result of the comparison is the coefficient of intermodulation distortion. If measurements are carried out for several n (usually 2 and 3 are enough), then the final intermodulation distortion coefficient is calculated from the intermediate ones (for different n) by taking the square root of the sum of their squares.

Power

You can talk about it for a very long time, since there are many types of measured speaker powers.

A few axioms:

loudness does not depend only on power. It also depends on the sensitivity of the speaker itself. And for an acoustic system, the sensitivity is determined by the sensitivity of the largest speaker, since it is the most sensitive;

the indicated maximum power does not mean that you can apply it to the system and the speakers will play perfectly. Everything is just more annoying. Maximum power for a long time with a high probability of damaging something in the dynamics. Manufacturer's warranty! Power should be understood as an unattainable limit. Only less. Not equal and even more so - more;

little of! At maximum power or close to it, the system will play extremely poorly, because the distortion will grow to completely indecent values.

The power of the speaker system is electrical and acoustic. It is unrealistic to see the acoustic power on the box with acoustics. Apparently, in order not to scare away the client with a small number. The fact is that the efficiency (efficiency) of the GG (loudspeaker head) in a very good case reaches 1%. The usual value is up to 0.5%. Thus, the acoustic power of the system can ideally be one hundredth of its electrical potential. Everything else is dissipated in the form of heat, spent on overcoming the elastic and viscous forces of the speaker.

The main types of power that can be seen on acoustics are: RMS, PMPO. This is electrical power.

RMS(Root Mean Squared - rms value) - the average value of the input electrical power. The power measured in this way has a semantic load. Measured by feeding a sine wave at a frequency of 1000 Hz, limited from above by a given value of THD (THD). It is imperative to study what level of non-linear distortion the manufacturer considered acceptable so as not to be deceived. It may be that the system is claimed to be 20 watts per channel, but the measurements were taken at 10% THD. As a result, it is impossible to listen to acoustics at this power. Also, at RMS power, the speakers can play for a long time.

PMPO(Peak Music Power Output - peak music output power). What is the use of a person knowing that his system can possibly carry a short, less than a second, low frequency sine with great power? However, manufacturers are very fond of this option. Indeed, on plastic speakers the size of a child's fist, there can be a proud figure of 100 watts. Healthy boxes of Soviet S-90s were not lying around! :) Oddly enough, such figures have a very distant relation to real PMPO. Empirically (based on experience and observation), you can get approximately real watts. Take Genius SPG-06 as an example (PMPO-120 Watts). It is necessary to divide the PMPO into 10 (12 watts) and 2 (the number of channels). The output is 6 watts, which is similar to the real figure. Once again: this method is not scientific, but based on the observations of the author. Usually works. In reality, this parameter is not so great, and the huge numbers are based only on the wild imagination of the marketing department.

Thiel-Small parameters

These parameters fully describe the speaker. There are parameters both constructive (area, mass of the moving system) and non-constructive (which follow from the constructive ones). There are only 15 of them. In order to roughly imagine what kind of speaker works in the column, four of them are enough.

Speaker resonant frequency fs(Hz) - the resonance frequency of the speaker, operating without acoustic design. Depends on the mass of the moving system and the rigidity of the suspension. It is important to know, since the speaker practically does not sound below the resonant frequency (the sound pressure level drops strongly and sharply).

Equivalent volume Vas(liters) - the useful volume of the case required for the operation of the speaker. Depends only on the area of ​​the diffuser (Sd) and the flexibility of the suspension. It is important because, while working, the speaker relies not only on the suspension, but also on the air inside the box. If the pressure is not what you need, then you will not see the ideal operation of the speaker.

Full quality factor Qts- the ratio of elastic and viscous forces in the moving system of the speaker near the resonance frequency. The higher the quality factor, the higher the elasticity in the dynamics and the more willingly it sounds at the resonant frequency. It consists of mechanical and electrical quality factors. Mechanical - this is the elasticity of the suspension and the corrugation of the centering washer. As usual, but it is the corrugation that provides greater elasticity, and not external suspensions. Mechanical quality factor - 10-15% of full quality factor. Everything else is an electrical quality factor formed by a magnet and a speaker coil.

DC resistance Re(Ohm). There is nothing special to explain here. Head winding resistance to direct current.

Mechanical quality factor Qms- the ratio of elastic and viscous forces of the speaker, elasticity is considered only the mechanical elements of the speaker. It consists of the elasticity of the suspension and the corrugation of the centering washer.

Electrical quality factor Qes- the ratio of elastic and viscous forces of the speaker, elastic forces arise in the electrical part of the speaker (magnet and coil).

diffuser area SD(m 2) - measured, roughly speaking, with a ruler. It has no secret meaning.

Sensitivity SPL(dB) - the sound pressure level developed by the loudspeaker. Measured at a distance of 1 meter with an input power of 1 watt and a frequency of 1 kHz (typically). The higher the sensitivity, the louder the system plays. In a two-way or more-way system, the sensitivity is equal to the SPL of the most sensitive speaker (usually a bass mug).

Inductance Le(Henry) is the inductance of the speaker coil.

Impedance Z(Ohm) - a complex characteristic that appears not on direct current, but on alternating current. The fact is that in this case, the reactive elements suddenly begin to resist the current. The resistance depends on the frequency. Thus, impedance is the ratio of the complex voltage amplitude and the complex current strength at a certain frequency. (Complex impedance depending on frequency, in other words).

Peak power Pe(Watt) is the PMPO discussed above.

Mass of the moving system mms(d) is the effective mass of the moving system, which includes the mass of the diffuser and the air oscillating with it.

Relative stiffness cms(meters/newton) - the flexibility of the movable system of the loudspeaker head, displacement under the influence of a mechanical load (for example, a finger that aims to poke the speaker). The higher the setting, the softer the suspension.

Mechanical resistance rms(kg/s) - active mechanical resistance of the head. Everything that can provide mechanical resistance in the head is included here.

Motor power BL- the value of the magnetic flux density multiplied by the length of the wire in the coil. Also, this parameter is called the force factor of the speaker. We can say that this is the power that will act on the diffuser from the side of the magnet.

All of these parameters are closely related. This is pretty obvious from the definitions. Here are the main dependencies:

fs increases with an increase in the rigidity of the suspension and decreases with an increase in the mass of the moving system;

Vas decreases with increasing suspension stiffness and increases with increasing diffuser area;

Qts increases with an increase in the stiffness of the suspension and the mass of the moving system and decreases with an increase in power BL.

So, now you are familiar with the basic theoretical apparatus necessary to understand articles on acoustic systems. Let's go directly to the testing methodology used by the authors of our portal.

Test Methodology

AFC. Measurement technique and interpretation

At the beginning of this section, we will deviate a little from the main topic and explain why all this is being done. First, we want to describe our own frequency response measurement method so that the reader does not have additional questions. Secondly, we will describe in detail how to perceive the obtained graphs and what can be said from the given dependencies, and also what should not be said. To start the methodology.

Measuring microphone Nady CM-100

Our frequency response measurement technique is quite traditional and differs little from the generally accepted principles for conducting detailed experiments. Actually the complex itself consists of two parts: hardware and software. Let's start with a description of the real devices that are used in our work. As a measurement microphone, we use a Behringer ECM-8000 high-precision condenser microphone with a circular pattern (omnidirectional), at a relatively low price, it has quite good parameters. So to speak, this is the "heart" of our system. This tool is designed specifically for use with modern technology as part of budget measuring laboratories. We also have a similar Nady CM-100 microphone at our disposal. The characteristics of both microphones will almost repeat each other, however, we always indicate which microphone was used to measure one or another frequency response. For example, here are the declared technical characteristics of the Nady CM-100 microphone:

impedance: 600 ohms;

sensitivity: -40dB (0dB=1V/Pa);

frequency range: 20-20000 Hz;

maximum sound pressure: 120 dB SPL;

power supply: phantom 15 ... 48 V.

Frequency response of the measuring microphone

M-Audio AudioBuddy Mic Preamp

As a microphone preamplifier, we use an external compact solution M-Audio AudioBuddy. The AudioBuddy preamp is designed specifically for digital audio applications and is optimized for microphones that require phantom power. Plus, the user has independent outputs: balanced or unbalanced TRS. The main parameters of the preamplifier are as follows:

frequency range: 5-50,000 Hz;

microphone gain: 60 dB;

input impedance of the microphone input: 1 kOhm;

instrument gain: 40 dB;

instrument input impedance: 100 kOhm;

power supply: 9 V AC, 300 mA.

Sound card ESI [email protected]

For further analysis, the signal from the output of the amplifier is fed to the input of a computer audio interface, which is an ESI PCI card. [email protected] This solution can be safely attributed to the class of semi-professional devices or even entry-level professional ones. Main parameters:

number of I/O: 4 inputs (2 analog, 2 digital), 6 outputs (2 analog, 4 digital);

ADC/DAC: 24-bit/192 kHz;

frequency response: 20 Hz - 21 kHz, +/- 0.5 dB;

dynamic range: ADC 114 dB, DAC 112 dB;

inputs: 2 analog, 2 digital (S/PDIF Coaxial);

outputs: 2 analog, 2 digital (S/PDIF Coaxial or Optical);

MIDI: 1 MIDI in and 1 MIDI out

interface: PCI;

sync: MTC, S/PDIF;

Drivers: EWDM driver support for Windows 98SE/ME/2000 and XP, MAC OS 10.2 or older.

In general, the unevenness of the path of the entire system in the frequency range of 20-20000 Hz lies within +/- 1 ... 2 dB, so our measurements can be considered quite accurate. The main negative factor is that all measurements are taken in an average living room with standard reverberation. The area of ​​the room is 34 m 2 , the volume is 102 m 3 . The use of an anechoic chamber, of course, increases the accuracy of the result, but the cost of such a chamber is at least several tens of thousands of dollars, so only large manufacturers of acoustic systems or other very wealthy organizations can afford such a "luxury". However, there are tangible advantages in this: for example, the frequency response in a real room will always be far from the frequency response, which was obtained by the manufacturer in the test chamber. Therefore, based on our results, we can draw some conclusions about the interaction of specific acoustics with the average room. This information is also very valuable, because any system will be operated in real conditions.

Popular utility rightmark Audio Analyzer

The second important point is the software part. We have several professional software packages at our disposal, such as RightMark Audio Analyzer ver. 5.5 (RMAA), TrueRTA ver. 3.3.2, LSPCad ver. 5.25, etc. As a rule, we use the convenient RMAA utility, provided that it is distributed free of charge and constantly updated, it is very practical and provides high measurement accuracy. In fact, it has already become the standard among test packages throughout the RuNet.

Program TrueRTA

Measuring module JustMLS programs LSPCad

It would seem that any measurement should be carried out according to strictly established rules, but in the field of acoustics there are too many of these rules, and often they differ somewhat from each other. For example, the basic norms and methods of measurement are given in several very weighty documents at once: outdated GOST USSR (GOST 16122-87 and GOST 23262-88), IEC recommendations (publications 268-5, 581-5 and 581-7), German standard DIN 45500, as well as US regulations AES and EIA.

We make our measurements in the following way. The acoustic system (AC) is installed in the center of the room at the maximum distance from walls and bulky objects, a high-quality stand 1 m high is used for installation. The microphone is installed at a distance of about a meter on a straight axis. The height is chosen so that the microphone "looks" approximately at the center point between the midrange and tweeters. The resulting frequency response is called a characteristic taken on a straight axis, and is considered one of the most important parameters in classical electroacoustics. It is believed that the fidelity of reproduction directly depends on the uneven frequency response. However, read about it below. We also always measure the angular characteristics of the system. In the ideal case, it is necessary to obtain a whole set of dependencies in the vertical and horizontal planes with a step of 10 ... 15 degrees. Then it is quite reasonable to draw conclusions about the speaker pattern, give advice on the correct arrangement in space. In fact, the angular frequency response is no less important than the straight-axis frequency response, since they determine the nature of the sound reaching the listener after reflection from the walls of the room. According to some reports, the proportion of reflections at the listening point reaches 80% or more. We also record all possible characteristics of the path with all available frequency adjustments, 3D-type modes, etc.

Simplified block diagram of the measurement process

You can tell a lot from these charts...

Subjective Listening

So, the frequency response graphs are received. What can be said by studying them in detail? In fact, much can be said, but it is impossible to unambiguously evaluate the system according to these dependencies. Not only is the frequency response not a very informative characteristic, and a number of additional measurements are required, for example, impulse response, transient response, cumulative attenuation of the spectrum, etc., it is rather difficult to give an unambiguous assessment of acoustics even using these exhaustive dependencies. Strong evidence of this is the official statement by AES (Journal of AES, 1994) that a subjective assessment is simply necessary to get a complete picture of the loudspeaker in addition to objective measurements. In other words, a person can hear a certain artifact, and it is possible to understand where it comes from only after a series of accurate measurements. Sometimes measurements help to identify an insignificant flaw that can easily slip past your ears when listening, and you can “catch” it only by focusing your attention on this particular range.

To begin with, it is necessary to break the entire frequency range into characteristic sections so that it is clear what is at stake. Agree, when we say "mid frequencies", it's not clear how much it is: 300 Hz or 1 kHz? Therefore, we suggest using the convenient breakdown of the entire sound range into 10 octaves, described in the previous section.

Finally, we pass directly to the moment of the subjective description of the sound. There are thousands of terms for evaluating what is heard. The best option is to use some documented system. And there is such a system, it is offered by the most authoritative publication with a half-century history of Stereophile. Relatively recently (in the early 90s of the last century), the Audio Glossary was published under the editorship of Gordon Holt. The dictionary contains an interpretation of more than 2000 concepts that in one way or another relate to sound. We suggest that you familiarize yourself with only a small part of them, which refers to the subjective description of sound in the translation of Alexander Belkanov (Magazine "Salon AV"):

ah-ax (rhymes with "rah" - Hurrah). Vowel coloring caused by a peak in the frequency response in the region of 1000 Hz.

Airy - airiness. Refers to high frequencies, sounding light, gentle, open, with a feeling of unlimited top. The property of a system that has a very flat response at high frequencies.

aw - (rhymes with "paw" [po:] - paw). Vowel coloring caused by a peak in the frequency response around 450 Hz. Seeks to emphasize, embellish the sound of large brass (trombone, trumpet).

Boomy - Read the word "boom" with a long "m". Characterizes an excess of middle bass, often with a predominance of a narrow bass band (very close to "one-note-bass" - bass on one note).

Boxy (literally - "box"): 1) characterized by "oh" - the color of vowels, as if the head is talking inside the box; 2) is used to describe the upper bass/lower mids of speakers with excessive cabinet wall resonances.

Bright, brilliant - bright, shiny, sparkling. An often misused term in audio, it describes the degree of hardness of the edge of the reproduced sound. Luminance refers to the energy contained in the 4-8 kHz band. This does not apply to the highest frequencies. All living sounds have brightness, the problem arises only when it is redundant.

Buzz - a buzzing low-frequency sound that has a fluffy or spiky character due to some uncertainty.

Chesty - from chest (chest). A pronounced density or heaviness in the reproduction of a male voice due to excessive energy in the upper bass / lower midrange.

Closed-in (literally - hidden, closed). Needs openness, air and good detail. Closed sounding is usually caused by high-frequency roll-off above 10 kHz.

Cold - cold, stronger than cool - cool. It has some excess highs and weakened lows.

Coloration - coloring. The audible "signature" with which a reproducing system colors all signals passing through it.

Cool - cool. Moderately devoid of density and warmth due to monotonous decay starting at 150 Hz.

Crisp - crisp, well-defined. Accurately localized and detailed, sometimes excessive due to a peak in the middle of the high range.

Cupped-hands - a mouthpiece from the palms. Coloring with nasal overtones or in extreme manifestation - sound through a megaphone.

Dark - dark, gloomy (literally). Warm, soft, overly rich sound. Perceived by ear as a clockwise slope of the frequency response over the entire range, so that the output level is attenuated with increasing frequency.

Dip (literally - immersion, failure). A narrow dip in the middle of a flat frequency response.

Discontinuity (literally - gap). Change in timbre or color when a signal passes from one head to another in multi-band acoustic systems.

Dished, dished-down - in the form of a saucer, an inverted saucer. Describes the frequency response with a failed mid. There is a lot of bass and treble in the sound, the depth is exaggerated. Perception is usually lifeless.

Dry (literally - dry). Describes the quality of the bass: lean, lean, usually overdamped.

Dull (literally - dull, dull, boring, lethargic, depressed). Describes a lifeless, veiled sound. Same as "soft" - soft, but to a greater extent. Audible high-frequency roll-off effect after 5 kHz.

her - rhymes with we. Vowel coloring caused by a peak in the frequency response around 3.5 kHz.

eh - as in "bed". Vowel coloration caused by a short rise in frequency response around 2 kHz.

Extreme highs - ultra high. The range of audible frequencies is above 10 kHz.

Fat (literally - plentiful, rich, fatty, oily). An audible effect of moderate redundancy in the middle and upper bass. Too warm, more "warm".

Forward, forwardness (literally - brought to the fore, forwardness). Playback quality that gives the impression that sound sources are closer than they were when recorded. As a rule, this is the result of a "hump" in the middle range plus a narrow directionality of the speakers.

Glare (literally - dazzling, sparkling). Unpleasant quality of hardness or brightness due to excessive energy in the lower or middle top.

Golden (literally - golden). A euphonious color characterized by roundness, richness, melodiousness.

Hard (literally - hard, hard). Aspiring to steel, but not so piercing. This is often the result of a moderate "hump" around 6 kHz, sometimes caused by slight distortion.

Horn sound - a horn sound made through a horn. The "aw" coloring found in many loudspeakers that have a midrange horn driver.

Hot (literally - hot). Sharp resonant surge at high frequencies.

Hum (literally - buzzing). Continuous "itching" at frequencies that are multiples of 50 Hz. Caused by the penetration of the main power frequency or its harmonics into the playback path.

Humped (literally - hunched). Characterizes the sound pushed forward (according to the spatial characteristic). The overall sound is sluggish, poor. Caused by a wide rise in the mids and a fairly early roll-off of lows and highs.

ih - as in the word "bit". Vowel coloring caused by a peak in the frequency response around 3.5 kHz.

Laid-back (literally - pushed back, pushed back). Suppressed, distant sound, with exaggerated depth, usually due to a dip in the middle range in the form of a saucer.

Lean - thin, skinny, frail. The effect of a weak decline in the frequency response down, starting from 500 Hz. It is less pronounced than "cool" - cool.

Light - light. The audible effect of tilting the frequency response counterclockwise from the middle. Compare with "dark" - dark.

Loose - loose, dangling, unstable. Refers to poorly defined/blurred and poorly controlled bass. Amplifier damping or driver/loudspeaker styling issues.

Lumpy (literally - lumpy). A sound characterized by some discontinuity in the frequency response in the lower part, starting from 1 kHz. Some areas appear to be bulging, others seem to be weakened.

Muffled - muted. Sounding very sluggish, dull, not having high frequencies at all. The result of high frequency rolloff above 2 kHz.

Nasal (literally - nasal, nasal). The sound is similar to speaking with a stuffy or blocked nose. Similar to the coloration of the vowel "eh". In loudspeakers, this is often caused by a measurable pressure peak in the upper midrange followed by a subsequent dip.

oh - pronunciation as in the word "toe". Vowel coloration caused by a wide frequency response peak around 250 Hz.

One-note-bass - bass on one note. The predominance of one low note is a consequence of a sharp peak in the lower range. Usually caused by poor damping of the woofer, room resonances can also appear.

oo - pronunciation as in the word "gloom". The color of the vowel is caused by a wide peak in the frequency response around 120 Hz.

Power range - maximum energy range. The frequency range of approximately 200-500 Hz corresponds to the range of powerful orchestra instruments - brass.

Presence range (literally - presence range). The lower part of the upper range is approximately 1-3 kHz, creating a sense of presence.

Reticent (literally - restrained). Moderately pushed back. Describes the sound of a system whose frequency response is saucer-shaped in the midrange. The opposite of forward.

Ringing (literally - ringing). Audible resonance effect: coloration, smeared/blurred sound, shrillness, buzz. It has the nature of a narrow peak in the frequency response.

Seamless (literally - without a seam, from a single / solid piece). It has no perceptible breaks in the entire audible range.

Seismic - seismic. Describes bass reproduction that makes the floor appear to be shaking.

Sibilance (literally - whistle, hiss). A coloring that emphasizes the vocal "s" sound. It can be associated with a monotonous rise in the frequency response from 4-5 kHz or with a wide overshoot in the 4-8 kHz band.

Silvery - silvery. Somewhat harsh, but clear sounding. Flute, clarinet, alto gives definition, but the gong, bells, triangle can communicate obsession, excessive harshness.

Sizzly - hissing, whistling. Raising the frequency response around 8 kHz, adding hiss (whistling) to all sounds, especially to the sound of cymbals and hissing in vocal parts.

Sodden, soggy (literally - wet, swollen with water). Describes loose and poorly defined bass. Creates a feeling of ambiguity, illegibility in the lower range.

Solid-state sound - transistor sound, semiconductor sound. A combination of sonic qualities common to most transistorized amplifiers: deep, tight bass, slightly pushed back bright stage character, and crisp, detailed highs.

Spitty (literally - spitting, snorting, hissing). The sharp "ts" is a coloring that unnecessarily emphasizes musical overtones and sizzles. It's like the sound of a vinyl record. Usually, the result is a sharp peak in the frequency response in the extreme high frequencies.

Steely - steel, steely. Describes shrillness, sharpness, importunity. Like "hard", but more so.

Thick - fat, thick, dull. Describes wet/dull or bulky, heavy bass.

Thin - liquid, frail, thin. Very lacking in bass. The result of a strong, monotonous downward decay starting at 500 Hz.

Tizzy (literally - excitement, anxiety), "zz" and "ff" - the coloring of the sound of cymbals and vocal hissing, caused by an increase in frequency response above 10 kHz. Similar to "wiry", but at higher frequencies.

Tonal quality - tonal quality. The precision/correctness with which the reproduced sound reproduces the timbres of the original instruments. (It seems to me that this term will be a good substitute for timbre resolution - A.B.).

Tube sound, tubey - sound due to the presence of tubes in the recording / playback path. The combination of sound qualities: richness (richness, liveliness, brightness of colors) and warmth, an excess of medium and a lack of deep bass. A bulging image of the scene. Tops are smooth and thin.

Wiry - hard, tense. Causes irritation with distorted high frequencies. Similar to brushes striking cymbals, but capable of coloring all the sounds produced by the system.

Wooly - sluggish, vague, shaggy. Refers to a dangling, loose, ill-defined bass.

Zippy - lively, fast, energetic. Slight emphasis on upper octaves.

So, now, looking at the given frequency response, you can characterize the sound with one or more terms from this list. The main thing is that the terms are systemic, and even an inexperienced reader can, by looking at their meaning, understand what the author wanted to say.

On what material are the acoustics tested? When choosing a test material, we were guided by the principle of diversity (after all, everyone uses acoustics in completely different applications - cinema, music, games, not to mention different tastes in music) and the quality of the material. In this regard, a set of test disks traditionally includes:

DVDs with movies and concert recordings in DTS and DD 5.1 formats;

discs with games for PC and Xbox 360 with high-quality soundtracks;

high-quality recorded CDs with music of various genres and directions;

MP3 discs with compressed music, material that is mainly heard on MM speakers;

special audiophile-quality test CDs and HDCDs.

Let's take a closer look at the test disks. Their purpose is to identify the shortcomings of acoustic systems. Allocate test disks with a test signal and with musical material. Test signals are generated reference frequencies (allow you to determine the boundary values ​​of the reproducible range by ear), white and pink noise, a signal in phase and antiphase, and so on. The most interesting to us seem to be the popular test disk FSQ (Fast Sound Quality) and Prime Test CD . Both of these discs, in addition to artificial signals, contain fragments of musical compositions.

The second category includes audiophile discs containing entire compositions recorded in studios of the highest quality and precision mixed. We use two licensed HDCDs (recorded at 24 bits and 88 kHz) - Audiophile Reference II (First Impression Music) and HDCD Sampler (Reference Recordings), as well as the Reference Classic CD sampler of classical music from the same label Reference Recordings .

Audiophilereference II(the disc allows you to evaluate such subjective characteristics as musical resolution, involvement, emotionality and presence effect, the depth of the nuances of the sound of various instruments. The musical material of the disc is classical, jazz and folk works recorded with the highest quality and produced by the famous sound wizard Winston Ma. On the recording you can meet magnificent vocals, powerful Chinese drums, deep string bass and get real listening pleasure on a really high-quality system.

HDCDsampler from Reference Recordings contains symphonic, chamber and jazz music. Using the example of his compositions, one can track the ability of acoustic systems to build a musical stage, to convey macro- and micro-dynamics, the naturalness of the timbres of various instruments.

referenceClassic shows us the real forte of Reference Recordings - chamber music recordings. The main purpose of the disk is to test the system for the correct reproduction of various timbres and the ability to create the correct stereo effect.

Z-characteristic. Measurement technique and interpretation

Surely even the most inexperienced reader knows that any dynamic head, and, consequently, the speaker system as a whole, has a constant resistance. This resistance can be regarded as resistance to direct current. For household equipment, the most familiar numbers are 4 and 8 ohms. In automotive technology, speakers with a resistance of 2 ohms are often found. The impedance of good monitor headphones can reach hundreds of ohms. From the point of view of physics, this resistance is due to the properties of the conductor from which the coil is wound. However, speakers, like headphones, are designed to work with audio-frequency alternating current. It is clear that with a change in frequency, the complex resistance also changes. The dependence characterizing this change is called the Z-characteristic. The Z-characteristic is quite important to study, because it is with the help of it that one can draw unambiguous conclusions about the correct matching of the speaker and amplifier, the correct calculation of the filter, etc. To remove this dependence, we use the LSPCad 5.25 software package, or rather, the JustMLS measuring module. Its capabilities are:

MLS size (Maximum-Length Sequence): 32764,16384,8192 and 4096

FFT (Fast Fourier Transform) size: 8192, 1024 and 256 points, used in different frequency bands

Sample rate: 96000, 88200, 64000, 48000, 44100, 32000, 22050, 16000, 1025, 8000 Hz and user selectable Custom (Select).

Window: Half Offset

Internal representation: 5 Hz to 50000 Hz, 1000 frequency points with logarithmic frequency.

To measure, you need to assemble a simple circuit: a reference resistor (in our case C2-29V-1) is connected in series from the speakers, and the signal from this divider is fed to the input of the sound card. The whole system (speaker/AC+resistor) is connected through the AF power amplifier to the output of the same sound card. We use the ESI interface for this purpose. [email protected] The program is very convenient because it does not require careful and lengthy settings. It is enough to calibrate the sound levels and press the "Measure" button. In a fraction of a second, we see the finished chart. Further, it is analyzed, in each case we pursue different goals. So, when studying a low-frequency speaker, we are interested in the resonant frequency to check the correct choice of acoustic design. Knowing the resonant frequency of the high-frequency head allows you to analyze the correctness of the crossover filter solution. In the case of passive acoustics, we are interested in the characteristic as a whole: it should be as linear as possible, without sharp peaks and dips. So, for example, acoustics, the impedance of which sags below 2 ohms, will be "not to the taste" of almost any amplifier. Such things should be known and considered.

Nonlinear distortion. Measurement technique and interpretation

Nonlinear distortions (Total Harmonic Distortion, THD) are the most important factor when evaluating loudspeakers, amplifiers, etc. This factor is due to the nonlinearity of the path, as a result of which additional harmonics appear in the signal spectrum. The harmonic distortion factor (THD) is calculated as the ratio of the square of the fundamental harmonic to the square root of the sum of the squares of the additional harmonics. As a rule, only the second and third harmonics are taken into account in calculations, although accuracy can be improved by taking into account all additional harmonics. For modern acoustic systems, the coefficient of non-linear distortion is normalized in several frequency bands. For example, for the zero complexity group according to GOST 23262-88, the requirements of which significantly exceed the minimum requirements of the IEC Hi-Fi class, the coefficient should not exceed 1.5% in the frequency band 250-2000 Hz and 1% in the band 2-6.3 kHz. Dry figures, of course, characterize the system as a whole, but the phrase "SOI = 1%" still says little. A vivid example: a tube amplifier with a THD of about 10% can sound much better than a transistor amplifier with the same coefficient of less than 1%. The fact is that the distortion of the lamp is mainly due to those harmonics that are screened by the auditory adaptation thresholds. Therefore, it is very important to analyze the spectrum of the signal as a whole, describing the values ​​of certain harmonics.

This is how the signal spectrum of a particular acoustics looks like at a control frequency of 5 kHz

In principle, you can see the distribution of harmonics over the spectrum with any analyzer, both hard and soft. The same RMAA or TrueRTA programs do this without any problems. As a rule, we use the first one. The test signal is generated using a simple generator, several control points are used. So, for example, non-linear distortions increased at high frequencies significantly reduce the microdynamics of the musical image, and a system with high distortions as a whole can simply greatly distort the timbre balance, wheeze, have extraneous overtones, etc. Also, these measurements make it possible to evaluate the acoustics in more detail in combination with other measurements, to check the correctness of the calculation of crossover filters, because the nonlinear distortions of the speaker increase greatly outside its operating range.

Article structure

Here we describe the structure of the article on acoustic systems. Although we try to make reading as pleasant as possible and do not squeeze ourselves into a certain framework, the articles are written with this plan in mind, so that the structure is clear and understandable.

1. Introduction

General information about the company is written here (if we get to know it for the first time), general information about the product line (if we take it for the first time), we give an outline of the current market situation. If the previous options do not fit, we write about trends in the acoustics market, in design, etc. - so that 2-3 thousand characters are written (hereinafter - k). The type of acoustics is indicated (stereo, surround sound, triphonic, 5.1, etc.) and positioning on the market - as a multimedia game for a computer, universal, for listening to music for an entry-level home theater, passive for a home theater, etc.

Tactical and technical characteristics, summarized in the table. Before the table with TTX, we make a small introduction (for example, "we have the right to expect serious YYY parameters from acoustics costing XXX"). The table view and the set of parameters are as follows:

For systems2.0

Parameter	Meaning
Output power, W (RMS)
Speaker external dimensions, WxDxH, mm
Gross weight, kg
Net weight, kg
Speaker diameter, mm
Speaker impedance, Ohm
Supply voltage, V
Frequency range, Hz
Frequency response unevenness in the operating range, +/- dB
Bass control, dB
Crosstalk, dB
Signal-to-noise ratio, dB
Completeness
Average retail price, $

For systems2.1

Parameter	Meaning
Satellite output power, W (RMS)
SOI at rated power, %
External dimensions of satellites, WxDxH, mm
Gross weight, kg
Net weight of satellites, kg
Subwoofer net weight, kg
Speaker diameter, mm
Speaker impedance, Ohm
Magnetic shielding, availability
Supply voltage, V
Adjustment of high frequencies, dB
Bass control, dB
Crosstalk, dB
Signal-to-noise ratio, dB
Completeness
Average retail price, $

For 5.1 systems

Parameter	Meaning
Output power of front satellites, W (RMS)
Output power of rear satellites, W (RMS)
Output power of the central channel, W (RMS)
Subwoofer output power, W (RMS)
Total output power, W (RMS)
SOI at rated power, %
External dimensions of the front satellites, WxDxH, mm
External dimensions of the rear satellites, WxDxH, mm
External dimensions of the central channel, WxDxH, mm
External dimensions of the subwoofer, WxDxH, mm
Gross weight, kg
Net weight of front satellites, kg
Net weight of rear satellites, kg
Net weight of the central channel, kg
Subwoofer net weight, kg
Speaker diameter, mm
Speaker impedance, Ohm
Magnetic shielding, availability
Supply voltage, V
Frequency range of satellites, Hz
Subwoofer frequency range, Hz
Frequency response unevenness in the full operating range, +/- dB
Adjustment of high frequencies, dB
Bass control, dB
Crosstalk, dB
Signal-to-noise ratio, dB
Completeness
Average retail price, $

We take the given tables as a basis, if there is additional data, we make more columns, the columns for which there is no data, we simply remove them. After the table with performance characteristics, small preliminary conclusions.

3. Packaging and equipment

We describe the delivery set and the box, at least two photos. Here we evaluate the completeness of the kit, describe the nature of the cables included in the kit, if possible, evaluate their cross section / diameter. We make a conclusion about the compliance of the set with the price category, convenience and design of the package. We note the presence of a Russian-language instruction manual, its completeness.

4. Design, ergonomics and functionality

We describe the first impression of the design. We note the nature of the materials, their thickness, quality factor. We evaluate design decisions in terms of potential impact on sound (don't forget to add the word "presumably"). We evaluate the workmanship, the presence of legs / spikes, grill / acoustic fabric in front of the diffusers. We are looking for fasteners, the ability to install on a rack / shelf / wall.

Ergonomics and impressions of working with acoustics (excluding listening) are described. There is a click when turned on, whether the length of the wires is sufficient, whether it is convenient to use all controls. Implementation of controls (analogue sliders or "knobs", digital knobs, toggle switches, etc.) Several photos of controls, remote control if available, photos of speakers in an environment or in comparison with ordinary objects. Convenience and speed of switching, the need to check the phasing, whether the instruction helps, etc. We note the effectiveness of magnetic shielding (on a CRT monitor or TV). We pay attention to additional inputs, operating modes (pseudo-surround sound, built-in FM tuner, etc.), service capabilities.

5. Construction

We disassemble the speakers, if there is a subwoofer, then also it. We note the following design features:

Type of acoustic design (open, closed box, phase inverter, passive radiate, transmission line, etc.) + general photo of the internal structure;

The dimensions and internal volume of the case, suggest the compatibility of AO with GG;

The location of the loudspeaker heads (GG), the method of attachment to the acoustic design;

Quality of internal installation, assembly, fastening + 1-2 photos with internal installation details;

The presence of mechanical damping, the quality of its execution and the materials used + photo;

The shape and dimensions of the phase inverter (if any), its location (probable effect on sound) and the manufacturer's likely devices to eliminate jet noise + photo;

The quality of internal wiring, the presence of overload protection, suggestions for modernization;

Used GG - type, material of manufacture (paper, impregnated silk, aluminum, plastic, etc.), nature of the diffuser surface (conical, exponential surface, corrugated, with "stiffeners", etc.) and protective cap (flat , "acoustic bullet", etc.), suspension (rubber, paper, etc.), degree of suspension rigidity), coil diameter, tweeter cooling, marking, resistance + photo of each GG;

Type of fastening of the wire to the speakers (separless, screw clamps, spring clamps, under the "banana", etc.) + photo;

Signal cable connectors - types, quantity, workmanship.

With diagrams and graphs, we illustrate the following things:

Amplifying microcircuit (s) - a table with key characteristics, their analysis for compliance with performance characteristics and speakers, if possible - give a graph of the dependence of power on SOI and a photo, you can have a photo of the radiator;

Power transformer - a table with currents, the type of transformer (torus, on W-shaped plates, etc.) indicating the total power in VA, conclusions about the presence of a supply power reserve, the presence of a power filter, etc. + photo;

Separation filter - we sketch the circuit, indicate the order of the filter (and, accordingly, the signal attenuation), we conclude that it is justified; applications (in the presence of appropriate measurements), we calculate the cutoff frequency in the event that in the future we measure the resonance and / or Z-characteristic;

We make a calculation of the resonant frequency of the phase inverter, give the formula and justify its use.

6. Measurements

We make the following measurements and provide an analysis for each of them, make assumptions about the nature of the sound.

Axial frequency response of the column with detailed analysis;

Frequency response of speakers at angles of 30 and 45 degrees, analysis of the nature of the dispersion of the speaker;

Subwoofer frequency response (if any) + total system frequency response, quality analysis; matching triphonics, the effect of phase inverter resonance;

Axial frequency response depending on the tone controls (if any);

Frequency response of a phase inverter, analysis;

The spectrum of harmonic distortion;

The frequency response of the speakers separately (for example, bass and treble), if necessary.

7. Audition

First, we give the first subjective assessment of the nature of the sound, indicate whether the volume is sufficient for various playback modes. We note the features of the acoustics in each of the typical applications - cinema (for 5.1 systems we focus on the quality of positioning), music and games. We indicate the type of room for listening, its area and volume, as well as the degree of exactingness of this acoustics to the room. Next, we analyze the sound of the speakers using the list of characteristics and terminology described above. We try to avoid subjective comments and at every opportunity we make a footnote to the measurement result, which confirmed one or another feature of the sound. In general, the entire analysis of the sound is done in the key of linking with the measurements. Be sure to pay attention to the following parameters:

The nature of the work of acoustics in each of the key frequency ranges, how much one or another range is accentuated;

The nature and quality of the stereo effect (the width of the stage, the positioning of sound sources and instruments on it), for acoustics 5.1, an assessment of the spatial positioning is given separately. Do not forget to place the acoustics correctly (the angle to the front pair is 45 degrees, the distance is slightly more than the stereo base, the rear pair is twice as close to the listener as the front one, all the speakers are at ear level);

Detail, sound transparency, "graininess" (post-pulse activity at medium and high frequencies);

The presence of color and its character in different ranges, timbre balance and naturalness of sound;

The clarity of the sound attack (impulse response) and separately - the operation of the subwoofer (if any);

Saturation of the signal with harmonics (warmth or coldness of sound);

Micro- and macrodynamics of sound, detail of background sounds, "openness" or "tightness" of sound (dynamic range width, transient response quality GG);

Optimal tone settings.

Here, a general assessment of acoustics is given, first of all, the correspondence of the solutions used in it to the final result and price category. It is estimated how successful the acoustics are, the perspective is suitable as a "blank" for modifications. A list of the pros and cons of the system is given.

Conclusion

The assiduous reader, having completed reading this article, probably brought out something new and interesting for himself. We did not try to embrace the immensity and cover all possible aspects of the analysis of acoustic systems and, moreover, the theory of sound, we will leave this to specialized publications, each of which has its own view on the line where physics ends and shamanism begins. But now all aspects of testing acoustics by the authors of our portal should be very clear. We never tire of repeating that sound is a subjective matter, and it is impossible to be guided when choosing acoustics by tests alone, but we hope that our reviews will greatly help you. Have a good sound, dear readers!

Comparative testing of Edifier and Microlab stereo speakers (April 2014)

Power

Under the word power in colloquial speech, many mean "power", "strength". Therefore, it is only natural that consumers associate power with loudness: “The more power, the better and louder the speakers will sound.” However, this popular belief is fundamentally wrong! It is far from always that a 100 W speaker will play louder or better than the one that has “only” 50 W power. The power value, rather, speaks not about the volume, but about the mechanical reliability of the acoustics. The same 50 or 100 watts is not loud at all published by the column. Dynamic heads themselves have low efficiency and convert only 2-3% of the power of the electrical signal supplied to them into sound vibrations (fortunately, the volume of the emitted sound is quite enough to create sound accompaniment). The value indicated by the manufacturer in the passport of the speaker or the system as a whole only indicates that when a signal of the specified power is applied, the dynamic head or speaker system will not fail (due to critical heating and interturn short circuit of the wire, “biting” of the coil frame, rupture of the diffuser , damage to flexible hangers of the system, etc.).

Thus, the power of the speaker system is a technical parameter, the value of which is not directly related to the loudness of the acoustics, although it is associated with some dependence. The nominal power values ​​of dynamic heads, amplifying path, acoustic system can be different. They are indicated, rather, for orientation and optimal pairing between the components. For example, an amplifier of much less or much more power can disable the speaker in the maximum positions of the volume control on both amplifiers: on the first - due to the high level of distortion, on the second - due to the abnormal operation of the speaker.

Power can be measured in various ways and under various test conditions. There are generally accepted standards for these measurements. Let us consider in more detail some of them, which are most often used in the characteristics of products of Western firms:

RMS (Rated Maximum Sinusoidal power- installed maximum sinusoidal power). Power is measured by applying a sinusoidal signal with a frequency of 1000 Hz until a certain level of non-linear distortion is reached. Usually in the passport for the product it is written like this: 15 W (RMS). This value says that the speaker system, when a 15 W signal is applied to it, can work for a long time without mechanical damage to the dynamic heads. For multimedia acoustics, higher power values ​​in W (RMS) compared to Hi-Fi speakers are obtained due to measurements at very high harmonic distortions, often up to 10%. With such distortions, it is almost impossible to listen to the soundtrack due to strong wheezing and overtones in the dynamic head and speaker cabinet.

PMPO(Peak Music Power Output Peak Music Power). In this case, the power is measured by applying a short-term sinusoidal signal with a duration of less than 1 second and a frequency below 250 Hz (typically 100 Hz). This does not take into account the level of non-linear distortion. For example, the speaker power is 500 W (PMPO). This fact indicates that the speaker system, after reproducing a short-term low-frequency signal, did not have mechanical damage to the dynamic heads. Popularly, the units of power measurement W (PMPO) are called “Chinese watts” due to the fact that power values ​​with this measurement technique reach thousands of watts! Imagine - active speakers for a computer consume 10 V * A electrical power from the AC mains and develop at the same time a peak musical power of 1500 W (PMPO).

Along with Western standards, there are also Soviet standards for various types of power. They are regulated by the current GOST 16122-87 and GOST 23262-88. These standards define concepts such as rated, maximum noise, maximum sinusoidal, maximum long-term, maximum short-term power. Some of them are indicated in the passport for Soviet (and post-Soviet) equipment. Naturally, these standards are not used in world practice, so we will not dwell on them.

We draw conclusions: the most important in practice is the value of the power indicated in W (RMS) at values ​​of harmonic distortion (THD) equal to 1% or less. However, comparing products even by this indicator is very approximate and may not have anything to do with reality, because the sound volume is characterized by the sound pressure level. That's why informativeness of the indicator "power of the acoustic system" zero.

Sensitivity

Sensitivity is one of the parameters specified by the manufacturer in the characteristics of acoustic systems. The value characterizes the intensity of the sound pressure developed by the column at a distance of 1 meter when a signal with a frequency of 1000 Hz and a power of 1 W is applied. Sensitivity is measured in decibels (dB) relative to the hearing threshold (zero sound pressure level is 2*10^-5 Pa). Sometimes the designation is used - the level of characteristic sensitivity (SPL, Sound Pressure Level). At the same time, for brevity, dB / W * m or dB / W ^ 1/2 * m is indicated in the column with units of measurement. It is important to understand, however, that sensitivity is not a linear proportionality factor between sound pressure level, signal strength and distance to the source. Many companies list the sensitivity characteristics of dynamic heads, measured under non-standard conditions.

Sensitivity is a characteristic that is more important when designing your own speaker systems. If you do not fully understand what this parameter means, then when choosing multimedia acoustics for a PC, you can not pay much attention to sensitivity (fortunately, it is not often indicated).

frequency response

Frequency response (frequency response) in the general case is a graph showing the difference in the amplitudes of the output and input signals over the entire range of reproducible frequencies. The frequency response is measured by applying a sinusoidal signal of constant amplitude as its frequency changes. At the point on the graph where the frequency is 1000 Hz, it is customary to plot the level of 0 dB on the vertical axis. The ideal option is in which the frequency response is represented by a straight line, but in reality, acoustic systems do not have such characteristics. When considering the graph, you need to pay special attention to the amount of unevenness. The greater the amount of unevenness, the greater the frequency distortion of the timbre in the sound.

Western manufacturers prefer to indicate the range of reproducible frequencies, which is a "squeeze" of information from the frequency response: only cutoff frequencies and unevenness are indicated. Suppose it is written: 50 Hz - 16 kHz (± 3 dB). This means that this acoustic system in the range of 50 Hz - 16 kHz has a reliable sound, and below 50 Hz and above 15 kHz, the unevenness increases sharply, the frequency response has a so-called "blockage" (a sharp drop in characteristics).

What does it threaten? Reducing the level of low frequencies implies a loss of juiciness, saturation of the bass sound. The rise in the bass region causes a sensation of mumbling and buzzing of the speaker. In the blockages of high frequencies, the sound will be dull, unclear. High-frequency rises mean the presence of annoying, unpleasant hissing and whistling overtones. In multimedia speakers, the frequency response unevenness is usually higher than in the so-called Hi-Fi acoustics. All advertising statements of manufacturing companies about the frequency response of a speaker of the type 20 - 20,000 Hz (theoretical limit of possibility) should be treated with a fair amount of skepticism. In this case, the uneven frequency response is often not indicated, which can be unimaginable values.

Since manufacturers of multimedia acoustics often "forget" to indicate the uneven frequency response of the speaker system, when meeting with a speaker characteristic of 20 Hz - 20,000 Hz, you need to keep your eyes open. There is a good chance of buying something that does not even provide more or less uniform response in the 100 Hz - 10,000 Hz frequency band. It is impossible to compare the range of reproducible frequencies with different irregularities at all.

Harmonic distortion, harmonic distortion

Kg coefficient of harmonic distortion. The acoustic system is a complex electro-acoustic device that has a non-linear gain characteristic. Therefore, the signal after the passage of the entire audio path at the output will necessarily have non-linear distortions. One of the most obvious and easiest to measure is harmonic distortion.

The coefficient is a dimensionless quantity. Specified either as a percentage or in decibels. Conversion formula: [dB] = 20 log ([%]/100). The higher the harmonic distortion value, the worse the sound is usually.

Kg speakers largely depends on the power of the signal fed to them. Therefore, it is foolish to draw conclusions in absentia or compare speakers only by the harmonic coefficient, without resorting to listening to the equipment. In addition, for the operating positions of the volume control (usually 30..50%), the value is not indicated by manufacturers.

Total electrical resistance, impedance

The electrodynamic head has a certain resistance to direct current, depending on the thickness, length and material of the wire in the coil (such resistance is also called resistive or reactive). When a musical signal, which is an alternating current, is applied, the head impedance will change depending on the frequency of the signal.

Impedance(impedans) is the total electrical resistance to alternating current, measured at a frequency of 1000 Hz. Typically, speaker impedance is 4, 6, or 8 ohms.

In general, the value of the total electrical resistance (impedance) of the speaker system will not tell the buyer about anything related to the sound quality of a particular product. The manufacturer indicates this parameter only so that the resistance is taken into account when connecting the speaker system to the amplifier. If the speaker impedance is lower than the amplifier's recommended load value, the sound may be distorted or short-circuit protected; if higher, the sound will be much quieter than with the recommended resistance.

Speaker box, acoustic design

One of the important factors affecting the sound of a speaker system is the acoustic design of the radiating dynamic head (speaker). When designing acoustic systems, the manufacturer usually faces the problem of choosing an acoustic design. There are more than a dozen types of them.

Acoustic design is divided into acoustically unloaded and acoustically loaded. The first implies a design in which the oscillation of the diffuser is limited only by the rigidity of the suspension. In the second case, the oscillation of the diffuser is limited, in addition to the rigidity of the suspension, by the elasticity of the air and acoustic resistance to radiation. Acoustic design is also divided into single and double action systems. The single action system is characterized by the excitation of the sound going to the listener by means of only one side of the cone (the radiation of the other side is neutralized by the acoustic design). The dual action system involves the use of both surfaces of the cone in the formation of sound.

Since the acoustic design of the speaker practically does not affect the high-frequency and mid-frequency dynamic heads, we will talk about the most common options for low-frequency acoustic design of the cabinet.

The acoustic scheme, called the "closed box", is very widely applicable. Refers to the loaded acoustic design. It is a closed case with a speaker cone displayed on the front panel. Advantages: good frequency response and impulse response. Disadvantages: low efficiency, need for a powerful amplifier, high level of harmonic distortion.

But instead of fighting the sound waves caused by the back side of the cone, they can be used. The most common variant of the double-acting systems is the phase inverter. It is a pipe of a certain length and section, built into the body. The length and cross section of the phase inverter are calculated in such a way that at a certain frequency, an oscillation of sound waves is created in it, in phase with the oscillations caused by the front side of the diffuser.

For subwoofers, an acoustic circuit with the generally accepted name "resonator box" is widely used. Unlike the previous example, the speaker cone is not displayed on the case panel, but is located inside, on the partition. The speaker itself does not directly participate in the formation of the low-frequency spectrum. Instead, the diffuser only excites low-frequency sound vibrations, which then multiply in volume in the phase inverter pipe, which acts as a resonant chamber. The advantage of these constructive solutions is high efficiency with small dimensions of the subwoofer. Disadvantages are manifested in the deterioration of phase and impulse characteristics, the sound becomes tiring.

The best choice would be medium-sized speakers with a wooden case, made according to a closed circuit or with a bass reflex. When choosing a subwoofer, you should pay attention not to its volume (by this parameter, even inexpensive models usually have a sufficient margin), but to reliable reproduction of the entire low frequency range. In terms of sound quality, speakers with a thin body or very small sizes are most undesirable.

Before going to the review combos for outdoor play I would like to get to the bottom of it. How is the sound we hear formed?
Sound in the process of formation goes something like this:

Pickup or Microphone --->
preamplifier --->
equalizer / effects set --->
power amplifier --->
acoustic system.

The acoustic system (speaker) is located at the output. And although the speaker takes up very little space in the picture, it forms the sound, and therefore determines it in many ways.

In other words: if the acoustic system is worthless, then no matter what high-quality signal comes from the PA, we will hear what the AU deigns to convey. It is worth noting that sometimes manufacturers of portable amps forget about this, installing completely mediocre speakers on their designs, which are simply not able to make sound of high quality and convey well what you are playing. Many combos suffer from this shortcoming.
However:

ACOUSTICS FIRST OF ALL DETERMINES THE SOUND OF THE SYSTEM!
And it is its most important component.
In general, it is strange that in the musical environment there is a lot of talk about, wood and guitars, effects sets, prev. amplifiers and power amplifiers, wires, but very little is mentioned about speakers and acoustic systems.
For me, this question arose, first of all, when I began to analyze the problems of poor sounding of portable equipment. The main trouble is small slurred, cheap speakers with poor sensitivity.

In the early 90s, when Hi-End first began to appear in Russia, there was a wonderful empirical formula for the distribution of resources. It looked something like this: 50% - acoustics, 10% - all cables, 40% - source and amplifier.
And this is generally true, because. it is the right acoustics that is the fundamental principle around which you can build your system and get high-quality sound.

And so, let's Let's move on to the speakers:

The main parts of the speaker are a magnet, a coil, a membrane (diffuser), a frame (basket, diffuser holder). The main components that affect the sound, parameters, configuration - purpose are the first three.
I also want to mention right away the parameters that are indicated on the speakers and by which they can be selected. (And let's get into the essence of each of them and how each part of the speaker affects it - a little later.)

SPEAKER PARAMETERS:

"Sensitivity" is the standard sound pressure (SPL) developed by the loudspeaker. It is measured at a distance of 1 meter with an input power of 1 watt at a fixed frequency (usually 1 kHz, unless otherwise noted in the speaker documentation).
The higher the sensitivity of the speaker system, the louder the sound it can produce for a given input power. Having speakers with high sensitivity, you can have a not too powerful amplifier, and vice versa, in order to “shake” speakers with low sensitivity, you need an amplifier with more power.
A sensitivity value such as 90dB/W/m means that the speaker is capable of producing 90dB sound pressure at 1m from the speaker with 1W input power. The sensitivity of conventional speakers ranges from 84 to 102 dB. Conventionally, the sensitivity of 84-88 dB can be called low, 89-92 dB - medium, 94-102 dB - high. If the measurements are carried out in a normal room, then the sound reflected from the walls is mixed with the direct radiation of the speakers, increasing the sound pressure level. For this reason, some companies list "anechoic" sensitivity for their speakers, measured in an anechoic chamber. It is clear that anechoic sensitivity is a more "honest" characteristic.

"frequency range" indicates the frequency limits within which the deviation of sound pressure does not exceed certain limits. Usually these limits are indicated in such a characteristic as “frequency response unevenness”.

AFC - amplitude-frequency characteristic of the speaker.
Shows the sound pressure level of the loudspeaker in relation to the reproduced frequency. Usually presented as a graph. Here is an example of a frequency response for a Celestion Vintage 30 speaker:

"Uneven frequency response"- shows the unevenness of the amplitude in the range of reproducible frequencies. Typically 10 to 18 dB.

(Correction - yes, ± 3dB - this is the speaker characteristic necessary for a more “honest” signal reproduction in the specified range.)

"Impedance" (RESISTANCE) is the electrical impedance of the speaker, typically 4 or 8 ohms. Some speakers have an impedance of 16 ohms, some are not standard values. 2, 6, 10, 12 ohm.

"Rated electrical power" RMS (Rated Maxmum Sinusoidal) - constant long-term input power. Denotes the amount of power that a loudspeaker can handle for an extended period of time without damage to the cone surround, overheating of the voice coil, or other annoyances.

"Peak Electrical Power"- maximum input power. Indicates the amount of power that the loudspeaker can withstand for a short time (1-2 seconds) without risk of damage.

Now you can consider how each of the parts of the speaker affects the parameters of the speaker and the sound - in general. :) But more on that in the following articles.

Other speaker parameters are such as diaphragm size and material. And their influence on properties and sound. Let's look at it in another article.

Kirill Trufanov
Guitar workshop.

According to the “canned” GOST (16122-78), any type of acoustic system is characterized by such indicators as sensitivity, reproducible frequency range and uneven frequency response (AFC) in this range. What to pay attention to first of all? And can everything be verified by algebra?

The sensitivity is measured when a sinusoidal voltage with an amplitude of 1 V of a certain frequency is applied to the acoustic system, while the microphone is located at a distance of 1 m. Then, by measuring the developed sound pressure sequentially, step by step, over the entire audible frequency range (by default, 20–20,000 Hz), we obtain AFC by sensitivity.

The range of reproducible frequencies is determined on the basis of the obtained frequency response. For example, if the global roll-off starts at 100 Hz at low frequencies, reaching, say, -40 dB at 60 Hz, then the lower limit of the operating range is based on some roll-off given by the rules adopted in a particular country. Thus, in our example, the lower limit of the ill-fated range can be 80 Hz, or maybe 70 Hz, as the rules require.

The unevenness of the frequency response is calculated similarly to the standard deviation in mathematical statistics, that is, first the average value of the amplitude is estimated within the frequency range, and then the bumpiness of the frequency response curve around the obtained average is calculated. The more uneven, the worse. Ideally, the frequency response is a straight line with no slope, but nothing is perfect in the real world.

The use of frequency response measured by sensitivity is convenient for assessing unevenness, but is completely unacceptable when comparing acoustic systems with different electrical impedance, which, in turn, depends on frequency. As a result of different impedance, acoustic systems consume different power when the same voltage is applied (the relationship between power, resistance, current and voltage can be found in a physics textbook). In other words, the average value of the amplitude "by sensitivity" for such acoustic systems will be, to put it mildly, "some in the forest, some for firewood." Therefore, the International Electrotechnical Commission (IEC), when measuring the frequency response, requires that not a voltage be supplied, but an electrical power equal to 1 W. The acoustic system will radiate a different (sound) power, roughly speaking, in accordance with the "personal" efficiency at different frequencies.

I note that the concept of "overseas" sensitivity is somewhat different from what we inherited from the times of the USSR. Sensitivity “their way” is measured in decibels (dB), and “ours” - in pascals (N / m2). It is easy to recalculate from our relatively standard zero sound pressure level (210–5 Pa).

Special mention should be made of the optimal frequency resolution, or, to put it simply, the step between the measured frequency response points. Dusty from time to time, highly specialized meters of the standard-guested frequency response are made on an analog base and pass the frequency range at a speed that increases with increasing frequency. Thus, a dependence on frequency close to logarithmic is obtained. "Analog" frequency responses have good resolution at low frequencies, but poor resolution at high frequencies (there the speed of running is too high for the recorder to meticulously record the amplitude of the signal from the microphone). The speed schedule is determined by the approved rules, and, of course, by the dynamic capabilities of analog equipment. Advanced frequency responses are now calculated by means of special sound analyzers, in which both a high-precision figure and a low-noise analogue coexist. High-quality sound analyzers that meet all international measurement requirements are mind-bogglingly expensive. Not every Russian company can afford a measuring analyzer, paying as much for it as for a brand new foreign car. To complete the picture, I will mention the price of a measuring microphone with a preamplifier (not included in the analyzer package): two thousand evergreens still have to be met. On the other hand, an ingenious measurement methodology makes it possible in most cases to do without an acoustically muted chamber, since the cost of the latter for measuring the frequency response of acoustic systems is simply ruinous. The frequency resolution of such analyzers exceeds that required by the current rules, however, the possibility of variation is provided, so to speak, for research purposes. By the way, the frequency changes linearly (!), which gives a lot of advantages, and then the analyzer recalculates the accumulated array into a logarithmic scale for display on a standardized graph.

With software simulation of obtaining the frequency response on a computer (using a sound card), the master oscillator signal is replaced by a digitally simulated signal. As a rule, they use a sliding tone (sweep tone), which smoothly runs through all sound frequencies. In the simulated signal, the sound frequency increases almost identically to the classical frequency response meter. This digital signal is played in real time (without pauses), and the DAC of the audio card produces an analog signal, which is fed (through the amplifier) ​​to the speakers; further, the sound emitted by the speakers is recorded through a microphone with a preamplifier and recorded by the ADC of the same sound card. It is clear that the card must be really full duplex in order to simultaneously (in fact, with a delay) voice and record. Each transducer, amplifier and microphone (as well as the room as an acoustic resonator) has its own frequency response, therefore, in order to obtain the correct characteristics of the speakers themselves, either the frequency response of all transducers must be ideal, or all deviations must be taken into account. The signal recorded digitally is immediately processed by a program that can produce a change in time of either the peak magnitude or the RMS power of the recorded signal. And since it is known in advance how the frequency changes in this signal, the frequency response seems to be already in your pocket. However, in order to correctly determine both the peak magnitude and the RMS power, you need to set the time interval during which these things will be calculated. If you set a small interval, you will get a frequency response close to the real one, but distorted by all sorts of bad irregularities. If you set a large interval, you will get a frequency response that has nothing to do with the real one, but it is smooth, easily interpreted even by a teapot. Moreover, in the case of a fixed interval, the largest error from combing-alignment will emerge as the frequency increases logarithmically. It is clear that in order to improve the frequency resolution, it will be necessary to lengthen the simulated signal, and this will lead to a violation of the “hosted” rules for measuring the frequency response.

There is one more subtlety. Any physical device has a response delay in time. In particular, the loudspeaker cone cannot respond instantly to disturbances. The larger the mass of the diffuser and the stiffer its suspension, the worse the reaction is potentially worse. Look "under a magnifying glass" at the response of a microphone over time, such as impact, and you will see a very complex transient. Despite the noted problems, software simulation allows you to calculate the frequency response quite close to the standard, but now we are talking about something else. Looks like the standard is outdated! Of course, you can continue to better programmatically simulate prehistoric hardware frequency response meters, but let's look at the root. By increasing the frequency resolution, you get a clear explanation of what numerous frequency response interpreters have been breaking spears over for decades.

The most difficult and insidious lies in this. As you know, it is impossible in principle to accurately determine the frequency and time at the same time (the so-called Heisenberg uncertainty). That is, to determine the frequency value, it is necessary to observe the signal for a sufficient period of time. The larger this gap, the more accurately the frequency can be determined, and vice versa. And since the frequency in the test sweep-signal is constantly changing, the error will be the smaller, the slower the frequency increases. The graph of the change in the frequency value is known exactly, since it is embedded in the software procedure for generating a test signal or sound file. The latter is disorienting. The frequencies in the signal recorded by the microphone will float relative to the simulated and voiced signal due to numerous intermediate transformations. So again we come to the need to slow down the frequency change in the sweep signal.

Instead of a sliding tone test signal, white noise is often used. And it's safer for speakers, and easier in terms of processing. But ... Here again there are "buts". The Fast Fourier Transform (FFT) procedure is used to decompose the recorded signal into a spectrum. To minimize errors of a random nature, it is necessary to average the FFT results obtained at different points in time. The more spectra are averaged, the smaller the error in calculating the frequency response. To improve the frequency resolution, increase the length of the time window for the FFT, that is, increase the sample size. In an effort to get high resolution at low frequencies, the sample size is raised beyond 65536. However, at low frequencies, the speakers sound white noise components with an underestimated acoustic power. And this leads to implausible blockages on the bottoms of such a frequency response.

Finally, the frequency response can be obtained by generating a delta pulse and calculating the modulus of the complex FFT from the registered transfer function. Here it is necessary to select the pulse repetition interval in order to minimize the errors by averaging the spectra. For a number of reasons, this method is more suitable for ADCs than for loudspeakers.

It is easy to guess that the three characteristics listed above are stationary estimates, that is, they do not take into account the dynamics of the acoustic system. "That's where the dog rummaged!" Experts (both talented self-taught and arrogant snobs hatched from rich music lovers) often try to unambiguously interpret the frequency response zigzags, peeping into other people's cheat sheets and guided by their own auditory sensations. Interpretation is a thankless task, since the frequency response of two acoustic systems can resemble each other like twin brothers, but these systems will sound differently. And it is not a fact that the same sounding speakers in all cases will have a frequency response like two drops of water. Alas, there is no strict unambiguity here. Then it turns out that no one needs the measured frequency response and they don’t say anything at all? No, it's not. It should just be remembered that the standard frequency response is just a conditional simplified reflection of reality (a cut of a rough cast in its own way), although it is performed strictly according to certain rules, I note, also conditional. Sometimes the closeness of the obtained frequency response to the true frequency response is very good, and sometimes, alas, very bad. Let's put it on our nose: although the frequency response is the result of objective assessments-measurements, its interpretation is a subjective matter. Like “the law that drawbar. Wherever you turn, there you go.” In other words, the graph of the guest frequency response is akin to error messages issued by the current Windows: a false message or not, complete nonsense or a random mixture of truth and falsehood, only an experienced specialist can determine.

Speaker manufacturers themselves are quietly using dynamic characteristics (for example, based on wavelet-conversion) to understand and understand what and how to improve in their speakers. Buyers, on the other hand, are shown in the old fashioned way only stationary characteristics, that is, frozen in time. Moreover, they are often very competently ennobled and combed, so that people who are uninitiated in the secrets of specific columns do not have any extra questions.

As for active acoustic systems, unlike passive ones, the task becomes more complicated, since the dynamics of the built-in amplifier is added to the dynamics (behavior in time) of the speakers. And the latter, like any non-measuring amplifier, has a different non-linear distortion factor at different frequencies and power levels.