Accurate diagnosis. Frequency response of acoustic systems. Description of calculation and interpretation methods

Before you get to the review combos for playing outside I would like to figure out the main thing. How is the sound we hear formed?
During the formation process, sound goes approximately this way:

Pickup or microphone --->
preamp --->
equalizer/effects set --->
power amplifier --->
acoustic system.

We have an acoustic system (speaker) at the output. And although the speaker takes up very little space in the picture, it forms the sound, and therefore determines a lot.

In other words: if the speaker system is bad, then no matter what high-quality signal comes from the PA, we will hear what the speaker deigns to transmit. 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 produce high-quality sound and convey well what you are playing. Many combos suffer from this drawback.
However:

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

In the early 90s, when Hi-End first began to appear in Russia, there was a wonderful empirical formula about 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 correctly chosen acoustics that is the fundamental basis 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 sound, parameters, configuration - purpose are the first three.
I would also like to immediately mention the parameters that are indicated on the speakers and by which they can be selected. (And we’ll delve into the essence of each of them and how each part of the speaker affects it - a little later.)

SPEAKER PARAMETERS:

"Sensitivity"- this is the standard sound pressure (SPL) that the loudspeaker develops. 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 specified in the speaker documentation).
The higher the sensitivity of the speaker system, the louder the sound it can produce for a given power input. Having AC with high sensitivity, you can have not too much powerful amplifier, and on the contrary, in order to “stimulate” speakers with low sensitivity, a higher power amplifier will be required.
A numerical sensitivity value, for example, 90 dB/W/m, means that this speaker is capable of creating a sound pressure of 90 dB at a distance of 1 m from the speaker with an input power of 1 W. The sensitivity of conventional speakers ranges from 84 to 102 dB. Conventionally, sensitivity 84-88 dB can be called low, 89-92 dB - medium, 94-102 dB - high. If 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. Therefore, some companies specify “anechoic” sensitivity for their speakers, measured in an anechoic chamber. It is clear that anechoic sensitivity is a more “honest” characteristic.

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

Frequency response - amplitude-frequency characteristic of the speaker.
Shows the sound pressure level of the speaker depending on the frequency being reproduced. Usually presented in graph form. Here is an example of the frequency response for the Celestion Vintage 30 speaker:

“Irregularity of frequency response”- shows uneven amplitude in the range of reproduced frequencies. Typically between 10 and 18 dB.

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

"Impedance" (RESISTANCE)- complete electrical resistance speakers, usually 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 power input. Refers to the amount of power that a loudspeaker can withstand for an extended period of time without damaging the cone surround, overheating the voice coil, or other problems.

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

Now you can consider how each part of the speaker affects the parameters of the speaker and the sound as a whole. :) But more on this in the following articles.

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

Kirill Trufanov
Guitar workshop.

We continue our tradition and publish another article in the “testing methods” series. Articles such as these serve as both a general theoretical framework to help readers gain an introduction to the topic, and 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 necessary? The fact is that sound and acoustics are practically the most complex of all the 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 some gaps in the knowledge of all amateurs good sound. So, let's start with the basic terms and concepts that any novice audiophile must know.

Basic terms and concepts

A short 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 happens that the average human ear can distinguish signals in the range from 20 to 20,000 Hz (or 20 kHz). This fairly substantial range, in turn, is usually divided into 10 octaves(can be divided by any other quantity, but 10 is accepted).

In general octave is a frequency range whose boundaries 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 two with an added zero at the end in their pure form. Actually, why do you need knowledge of octaves? It is necessary in order to stop the confusion about what should be called lower, middle or some other bass and the like. The generally accepted set of octaves clearly determines who is who to the nearest hertz.

Octave number	Lower limit, Hz	Upper limit, Hz	Name	Title 2
			Deep Bass
			Mid Bass	Subcontrol
			Upper Bass
			Lower middle
			Actually the middle
			Upper middle
			Bottom top
			Middle 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 "Title 2". This contains the names of the octaves that are highlighted by musicians. These “strange” people have no concept of deep bass, but they have one octave above - from 20480 Hz. That is why there is such a discrepancy in numbering and names.

Now we can talk more specifically about the frequency range of speaker systems. We should start with some unpleasant news: there is no deep bass in multimedia acoustics. The vast majority of music lovers have simply never heard 20 Hz at a level of -3 dB. And now the news is pleasant and unexpected. There are no such frequencies in a real signal either (with some exceptions, of course). An exception is, for example, a recording from an IASCA Competition judge's 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 judges will decorate the system that wins over the Vikings with awards, like a Christmas tree with toys. And with real signal everything is simpler: bass drum - from 40 Hz. Hefty Chinese drums also start from 40 Hz (among them, however, there is one megadrum. So it starts playing as early as 30 Hz). Live double bass is generally from 60 Hz. As you can see, 20 Hz is not mentioned here. Therefore, you don’t have to worry about the absence of such low components. They are not needed to listen to real music.

The figure shows a spectrogram. There are two curves on it: purple DIN and green (from old age) IEC. These curves display 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 noticed that listener preferences and music had changed somewhat. This is reflected in the great and mighty DIN standard. As you can see, there are much more high frequencies. But there was no increase in bass. Conclusion: no need to chase super-bass systems. Moreover, the desired 20 Hz was not put in the box there anyway.

Characteristics of acoustic systems

Now, knowing the alphabet of octaves and music, you can begin to understand the frequency response. Frequency response (amplitude-frequency response) - dependence of the oscillation amplitude at the device output on the frequency of the input harmonic signal. That is, the system is supplied with a signal at the input, the level of which is taken as 0 dB. From this signal, speakers with an amplification path do what they can. What they usually end up with is not a straight line at 0 dB, but a somewhat broken line. The most interesting thing, by the way, is that everyone (from audio enthusiasts to audio manufacturers) strives for a perfectly flat frequency response, but they are afraid to “strive.”

Actually, what is the benefit of the frequency response and why do the authors of TECHLABS constantly try to measure this curve? The fact is that it can be used to establish real frequency range boundaries, and not those whispered by the “evil marketing spirit” to the manufacturer. It is customary to indicate at what signal drop the boundary frequencies are still played. If not specified, it is assumed that the standard -3 dB was taken. This is where the catch lies. It is enough not to indicate at what drop the boundary 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 from it, although approximately, you can understand what problems the selected system will have. Moreover, the system as a whole. The frequency response suffers from all elements of the path. To understand how the system will sound according to the schedule, you need to know the elements of psychoacoustics. In short, the situation is like this: a person speaks within medium frequencies. That’s why he perceives them best. And at the corresponding octaves the graph should be the most even, since distortions in this area put a lot of pressure on the ears. The presence of tall narrow peaks is also undesirable. The general rule here is that peaks are heard better than valleys, 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 measuring it.

Phase frequency response (PFC) shows the change in the phase of the harmonic signal reproduced by the speaker depending on the frequency. Can be uniquely calculated from the frequency response using the Hilbert transform. The ideal phase response, which says that the system has no phase-frequency distortions, is a straight line passing through the origin of coordinates. Acoustics with such a phase response are called phase-linear. For a long time this characteristic was not paid attention to, 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 Spectral Attenuation (CSF) - 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, from the GLC graph one can accurately say which regions of the spectrum will decay at what speed after the pulse, that is, the graph allows one to identify delayed resonances of the AS.

If the KZS has many resonances after the upper middle, then such acoustics will subjectively sound “dirty”, “with sand on the high frequencies”, etc.

AC impedance - this is the total electrical resistance of the speaker, including the resistance of the filter elements (complex value). This resistance contains not only active resistance, but also the reactance of capacitors and inductances. Since reactance depends on frequency, impedance is also entirely dependent on it.

If we talk about impedance as numerical value, completely devoid of complexity, then they speak out about its module.

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

Considering that impedance depends on frequency and is not constant, you can easily determine from it how difficult the acoustics are for an amplifier. Also, from the graph you can tell what kind of acoustics it is (ZYa - closed box), FI (with a bass reflex), how individual sections of the range will be reproduced.

Sensitivity - see Thiel-Small parameters.

Coherence - coordinated occurrence of several oscillatory or wave processes in time. This means that the signal from different GG acoustic systems will arrive at the listener simultaneously, that is, it indicates the safety of phase information.

Listening Room Meaning

The listening room (among audiophiles is often shortened to KdP), and its conditions are extremely important. Some put the CDP in first place in importance, and only after that - acoustics, amplifier, source. This is somewhat justified, since the room is capable of doing whatever it wants with the graphs and parameters measured by the microphone. Peaks or dips in the frequency response may appear that were not observed during measurements in a quiet room. Both the phase response (following the frequency response) and the transient characteristics will change. In order to understand where such changes come from, we need to introduce the concept of room modes.

Room mods are the 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 (CLR) is completely unpredictable. There are, of course, calculations that allow us to evaluate the influence of various modes on sound. But they exist for an empty room with an idealized finish. Therefore, it is not worth presenting them here; they have no practical value in everyday life.

However, you must 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. Bass boominess (resonance at 35-100 Hz) - bright representative the appearance of resonances in response to a low frequency signal in a standard room of 16-20 m2. High frequencies give rise to slightly different problems: diffraction and interference of sound waves appear, which make the directivity characteristics of the speakers frequency-dependent. That is, the directionality of the speakers becomes increasingly narrow with increasing frequency. It follows that maximum comfort will be received by the listener at the intersection of the acoustic axes of the speakers. And only him. 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 if the control panel is muffled. For this, various sound-absorbing materials are used - from thick curtains and carpets to special slabs and cunning configurations of walls and ceilings. The quieter the room, the more the speakers contribute to the sound, and not the reflections from your favorite computer desk and pot of geraniums.

Recipes for placing speakers in a room

Vandersteen recommends placing speakers along the long 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 the long wall successively into three, five, seven and nine parts, draw the corresponding lines perpendicular to this wall. Do the same with the side wall. The intersection points 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 whether the stands under the speakers are stable: if necessary, use spikes or conical legs;

Check how solid the wall behind the speaker is. If the wall is flimsy and makes noise, place the speaker in front of a strong (solid) wall.

The stereo image does not extend 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:

select the optimal height for the speakers (use stands) and your listening position.

Sharp annoying sound in the mid and high frequencies:

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

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

Distortions

From subjectivism we need to move on to technical concepts. It's worth starting with distortions. They are divided into two large groups: linear and nonlinear distortions. Linear distortion do not create new spectral components of the signal; they change only the amplitude and phase components. (They distort the frequency response and phase response, respectively.) Nonlinear distortion make changes to the signal spectrum. Their number in the signal is presented in the form of nonlinear distortion and intermodulation distortion coefficients.

Harmonic distortion factor (THD, THD - total harmonic distortion) is an indicator characterizing the degree to which the voltage or current shape differs from the ideal sinusoidal shape. In Russian: a sinusoid is supplied to the input. At the output, it does not resemble itself, since the path introduces changes in the form of additional harmonics. The degree of difference between the signal at the input and 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 f 1 And f 2(based on the recommendation of IEC 268-5, frequencies are taken for measurements f 1 and f 2, such that f 1 < f 2/8. You can take another relationship between frequencies). Intermodulation distortion is assessed quantitatively by spectral components with frequencies f 2±(n-1) f 1, where n=2,3,... At the system output, the number of extra harmonics is compared and the percentage of the spectrum they occupy is estimated. The result of the comparison is the intermodulation distortion coefficient. If measurements are carried out for several n (usually 2 and 3 are sufficient), 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

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

A few axioms:

Volume does not depend only on power. It also depends on the sensitivity of the speaker itself. And for an acoustic system, 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 unpleasant. Maximum power for a long time is highly likely to damage something dynamically. Manufacturer's warranty! Power should be understood as an unattainable limit. Only less. Not equal, and certainly not more;

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

The power of the speaker system can be electrical or acoustic. It is unrealistic to see the acoustic power on a box with acoustics. Apparently, so as not to scare off the client with a small number. The fact is that efficiency (coefficient useful action) GG (loudspeaker heads) in very good case reaches 1%. The usual value is up to 0.5%. Thus, the acoustic power of a 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 powers that can be seen on acoustics are: RMS, PMPO. This is electrical power.

RMS(Root Mean Squared - root mean square value) - average value of the input electrical power. Power measured in this way has a meaning. It is measured by applying a sine wave with a frequency of 1000 Hz, limited from above by a given value of total harmonic distortion (THD). It is imperative to study what level of nonlinear distortion the manufacturer considered acceptable, so as not to be deceived. It may turn out that the system is stated at 20 watts per channel, but the measurements were carried out at 10% SOI. As a result, it is impossible to listen to acoustics at this power. Also, the speakers can play at RMS power for a long time.

PMPO(Peak Music Power Output - peak music output power). What is the benefit of a person knowing that his system may suffer a short, less than a second, low frequency sine wave with high power? However, manufacturers are very fond of this option. After all, on plastic speakers the size of a child’s fist there can be a proud number of 100 Watts. There were no healthy boxes of Soviet S-90s lying around! :) Oddly enough, such figures have very little relation to the real PMPO. Empirically (based on experience and observations) you can obtain approximately real watts. Let's take the Genius SPG-06 as an example (PMPO-120 Watt). It is necessary to divide PMPO into 10 (12 Watts) and 2 (number of channels). The output is 6 watts, which is similar to the real figure. Once again: this method is not scientific, but is based on the author’s observations. Usually works. In reality, this parameter is not so large, and the huge numbers are based only on the wild imagination of the marketing department.

Thiel-Small parameters

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

Speaker resonant frequency Fs(Hz) - resonance frequency of a 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 below the resonant frequency the speaker practically does not sound (the sound pressure level drops strongly and sharply).

Equivalent Volume Vas(liters) - the useful volume of the housing required for the speaker to operate. Depends only on the diffuser area (Sd) and the flexibility of the suspension. It is important because, when working, the speaker relies not only on the suspension, but also on the air inside the box. If the pressure is not what is needed, then the speaker will not work perfectly.

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

Resistance DC Re(Ohm). There is nothing special to explain here. Resistance of the head winding to direct current.

Mechanical quality factor Qms- the ratio of elastic and viscous forces of the speaker; elasticity is considered only for the mechanical elements of the speaker. It is made up 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(m2) - measured, roughly speaking, with a ruler. It has no secret meaning.

Sensitivity SPL(dB) - 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 (typical). 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 the bass mug).

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

Impedance Z(Ohm) is 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. Resistance depends on frequency. Thus, impedance is the ratio of the complex voltage amplitude and the complex current at a certain frequency. (Frequency dependent complex impedance, in other words).

Peak power Pe(Watt) is PMPO, which is discussed above.

Weight 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 hardness Cms(meters/newton) - flexibility of the moving system of the loudspeaker head, displacement under the influence of mechanical load (for example, a finger that aims to poke the speaker). The higher the parameter, the softer the suspension.

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

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

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

Fs increases with increasing rigidity of the suspension and decreases with increasing mass of the moving system;

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

Qts increases with increasing rigidity of the suspension and mass of the moving system and decreases with increasing power B.L..

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

Testing methodology

Frequency response 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 method for measuring frequency response so that the reader does not have any additional questions. Secondly, we will tell you in detail how to perceive the resulting graphs and what can be said from the given dependencies, as well as what should not be said. Let's start with the methodology.

Measurement microphone Nady CM-100

Our technique for measuring frequency response is quite traditional and differs little from the generally accepted principles of 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. We use a high-precision microphone as a measuring microphone. condenser microphone Behringer ECM-8000 with an omnidirectional pattern (omnidirectional), at a relatively low price it has quite good parameters. So to speak, this is the “heart” of our system. This tool designed specifically for use with modern technology as part of budget measurement laboratories. We also have at our disposal a similar microphone, the Nady CM-100. The characteristics of both microphones practically repeat each other, however, we always indicate with which microphone a particular frequency response was measured. As an example, here are the stated technical characteristics of the Nady CM-100 microphone:

impedance: 600 Ohm;

sensitivity: -40 dB (0 dB = 1 V/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 microphone preamplifier

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

frequency range: 5-50,000 Hz;

microphone gain: 60 dB;

microphone input impedance: 1 kOhm;

instrument gain: 40 dB;

instrument input impedance: 100 kOhm;

power supply: 9 V AC, 300 mA.

Sound card ESI Juli@

For further analysis, the signal from the output of the amplifier is fed to the input of a computer audio interface, which uses an ESI Juli@ PCI card. This solution can easily be classified as a semi-professional device or even a professional one. entry level. 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 range: 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 input and 1 MIDI output;

interface: PCI;

synchronization: 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 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 carried out in an average living room with standard reverberation. The area of ​​the room is 34 m2, the volume is 102 m3. The use of an anechoic chamber, naturally, increases the accuracy of the result obtained, 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 also tangible advantages to this: for example, the frequency response in a real room will always be far from the frequency response that 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 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, subject to free distribution and constant updates it is very practical and provides high measurement accuracy. In fact, it has already become a standard among test packages all over 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 they often diverge somewhat from each other. For example, the basic standards and measurement methods are given in several very significant documents at once: outdated GOSTs of the USSR (GOST 16122-87 and GOST 23262-88), IEC recommendations (publications 268-5, 581-5 and 581-7), German DIN standard 45500, as well as American AES and EIA regulations.

We make our measurements as follows. The acoustic system (AS) is installed in the center of the room at the maximum distance from walls and three-dimensional 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 in such a way that the microphone “looks” at approximately the central point between the midrange and tweeter speakers. The resulting frequency response is called the characteristic taken on the direct axis, and in classical electroacoustics it is considered one of the most important parameters. It is believed that the fidelity of reproduction directly depends on the unevenness of the frequency response. However, read about this below. We also always measure the angular characteristics of the system. Ideally, it is necessary to obtain a whole set of dependencies in the vertical and horizontal planes in increments of 10...15 degrees. Then it is quite reasonable to draw conclusions about the directional pattern of the speakers and give advice on the correct placement in space. In fact, the angular frequency response is no less important than the frequency response along the straight axis, since they determine the nature of the sound reaching the listener after reflection from the walls of the room. According to some reports, the share of reflections at the listening point reaches 80% or more. We also remove all possible characteristics of the path with all available frequency adjustments, modes such as 3D, etc.

Simplified flowchart of the measurement process

You can tell a lot from these graphs...

Subjective listening

So, the frequency response graphs have been obtained. What can you say after studying them in detail? In fact, a lot can be said, but it is impossible to unambiguously evaluate the system based on these dependencies. Not only is the frequency response not a very informative characteristic, and a whole series of additional measurements are required, for example, impulse response, transient response, cumulative spectrum attenuation, etc., but even from these comprehensive dependencies it is quite difficult to give an unambiguous assessment of acoustics. Strong evidence of this can be found in the official statement of the AES (Journal of AES, 1994) that subjective assessment is simply necessary to obtain a complete picture of the acoustic system in combination with objective measurements. In other words, a person can hear a certain artifact, but it is possible to understand where it comes from only by making a series of precise measurements. Sometimes measurements help to identify an insignificant defect that can easily slip past your ears when listening, and you can “catch” it only by focusing your attention on this particular range.

First, you need to break the entire frequency range into characteristic sections so that it is clear what we are talking about. Agree, when we say “mid frequencies”, it is not clear how much it is: 300 Hz or 1 kHz? Therefore, we suggest using a convenient division of the entire sound range into 10 octaves, described in the previous section.

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

ah-ax (rhymes with "rah" - Hurray). The coloring of vowels caused by a peak in the frequency response around 1000 Hz.

Airy - airiness. Refers to high frequencies that sound light, gentle, open, with a feeling of unlimited top end. A property of a system that has a very smooth response at high frequencies.

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

Boomy - read the word "boom" with a long "m". Characterizes an excess of mid-bass, often with a predominance of a narrow low-frequency band (very close to “one-note-bass” - bass on one note).

Boxy (literally “boxy”): 1) characterized by “oh” - the coloring of the vowels, as if the head is speaking inside the box; 2) used to describe the upper bass/low mids of speakers with excessive cabinet wall resonances.

Bright, brilliant - bright, with shine, sparkling. An often misused term in audio, it describes the degree of hardness of the edge of the sound being reproduced. 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 there is excess of it.

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

Chesty - from chest (chest). A pronounced density or heaviness when reproducing 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 sound is usually caused by HF roll-off above 10 kHz.

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

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

Cool - cool. Moderately lacking in density and warmth due to monotonic decay starting at 150 Hz.

Crisp - crisp, clearly defined. Precisely localized and detailed, sometimes excessively due to the peak in the mid-HF range.

Cupped-hands - a mouthpiece made of palms. Coloration with a nasal sound or, in extreme cases, sound through a megaphone.

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

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

Discontinuity (literally - gap). Change in timbre or color during the transition of a signal 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 middle. The sound has a lot of bass and high frequencies, 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. An audible HF roll-off effect after 5 kHz.

her - rhymes with we. Coloration of vowels caused by a peak in the frequency response around 3.5 kHz.

eh - as in "bed". Coloration of vowels caused by a short rise in frequency response in the region of 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 mid and upper bass. Excessively warm, more "warm".

Forward, forwardness (literally - brought to the fore, moving forward). A reproduction quality that gives the impression that sound sources are closer than they were when recorded. Typically this is the result of a hump in the midrange plus the narrow directivity of the speakers.

Glare (literally - dazzling, sparkling). An unpleasant quality of hardness or brightness due to excessive low or mid-high energy.

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

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. "aw" coloring, characteristic of many acoustic systems that have a mid-frequency horn driver.

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

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

Humped (literally - hunched over). Characterizes the sound pushed forward (in terms of spatial characteristics). Overall sound sluggish, meager. Caused by a broad rise in the mid frequencies and a fairly early fall in the lows and highs.

ih - as in the word "bit". Coloration of vowels caused by a peak in the frequency response around 3.5 kHz.

Laid-back (literally - pushed back, pushed back). Depressed, distant sounding, with exaggerated depth, usually due to a saucer-shaped midrange.

Lean - thin, skinny, frail. The effect of a slight downward decline in frequency response, starting from 500 Hz. Less pronounced than “cool” - cool.

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

Loose - loose, loose, unstable. Refers to poorly defined/washed out and poorly controlled bass. Problems with damping of the amplifier or dynamic drivers/acoustic design of speakers.

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

Muffled - muted. It sounds very sluggish, dull, and has no high frequencies in the spectrum at all. The result is a roll-off of high frequencies above 2 kHz.

Nasal (literally - nasal, nasal). It sounds similar to talking with a stuffy or pinched nose. Similar to the coloring of the vowel "eh". In loudspeaker systems, this is often caused by a measured pressure peak in the upper midrange followed by a dip.

oh - pronunciation as in "toe". The coloring of a vowel caused by a wide spike in the frequency response in the region of 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 head, room resonances can also appear.

oo - pronunciation as in the word "gloom". The coloring of the vowel is caused by a wide surge in the frequency response in the region of 120 Hz.

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

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

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

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

Seamless (literally - without a seam, from a single/solid piece). There are no noticeable discontinuities throughout the entire audible range.

Seismic - seismic. Describes the reproduction of low frequencies that make the floor seem to vibrate.

Sibilance (literally - whistling, hissing). Coloration emphasizing the vocal sound "s". It may be associated with a monotonic rise in the frequency response from 4-5 kHz or with a wide surge in the 4-8 kHz band.

Silvery - silvery. Somewhat harsh, but clear sound. It gives the flute, clarinet, and violas an edge, but the gong, bells, and triangles can be obtrusive and excessively sharp.

Sizzly - hissing, whistling. The frequency response rises in the region of 8 kHz, adding hissing (whistle) 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 vagueness and illegibility in the lower range.

Solid-state sound - transistor sound, semiconductor sound. A combination of sonic qualities common to most solid-state amplifiers: deep, tight bass, slightly offset bright stage character and clearly defined, detailed treble.

Spitty (literally - spitting, snorting, hissing). A sharp “ts” is a coloring that overemphasizes musical overtones and sibilants. Similar to the surface noise of a vinyl record. Usually, the result is a sharp peak in the frequency response in the extreme HF region.

Steely - steely, steely. Describes shrillness, harshness, importunity. Similar to "hard", but to a greater extent.

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

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

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

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

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

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

Wooly - lethargic, vague, shaggy. Refers to loose, loose, poorly defined bass.

Zippy - lively, fast, energetic. Slight emphasis in the upper octaves.

So, now, looking at the given frequency response, you can characterize the sound in 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.

What material is the acoustics tested on? When choosing 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, the set of test disks traditionally includes:

DVDs with films 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 genres;

MP3 discs with compressed music, material that is mainly listened to on MM acoustics;

special test CDs and HDCDs of audiophile quality.

Let's take a closer look at the test discs. Their purpose is to identify shortcomings in acoustic systems. There are test discs with a test signal and with musical material. Test signals are generated reference frequencies (allowing you to determine by ear the boundary values ​​of the reproduced range), white and pink noise, a signal in phase and antiphase, and so on. The popular test disk seems to us the most interesting F.S.Q. (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 mixed with precision. We use two licensed HDCD discs (recorded at 24-bit and 88 kHz sampling frequency) - Audiophile Reference II (First Impression Music) and HDCD Sampler (Reference Recordings), as well as a CD sampler of classical music, Reference Classic, from the same label, Reference Recordings .

AudiophileReference II(the disc allows you to evaluate such subjective characteristics as musical resolution, involvement, emotionality and presence, 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 find magnificent vocals, powerful Chinese drums, deep string bass, and on a truly high-quality system you can get real listening pleasure.

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

ReferenceClassic shows us the real strong point of Reference Recordings - chamber music recordings. The main purpose of the disc is to test the system for faithful 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 direct current resistance. For household equipment, the most common numbers are 4 and 8 ohms. In automotive technology, speakers with a resistance of 2 ohms are often found. The resistance of good monitor headphones can reach hundreds of ohms. From a physics point of view, this resistance is determined by the properties of the conductor from which the coil is wound. However, speakers, like headphones, are designed to operate with audio frequency alternating current. It is clear that as the frequency changes, 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 software package LSPCad 5.25, 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

Sampling rates: 96000, 88200, 64000, 48000, 44100, 32000, 22050, 16000, 1025, 8000 Hz and user selectable Custom.

Window: Half Offset

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

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 entire system (speaker/AC+resistor) is connected through an AF power amplifier to the output of the same sound card. We use the ESI Juli@ interface for these purposes. The program is very convenient because it does not require careful and lengthy setup. Just calibrate the sound levels and press the "Measure" button. In a split second we see the finished graph. Next comes its analysis; in each specific case we pursue different goals. So, when studying the woofer, we are interested in 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 isolation 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 whose impedance sags below 2 ohms will not be to the taste of almost any amplifier. These things should be known and taken into account.

Nonlinear distortions. Measurement technique and interpretation

Total Harmonic Distortion (THD) is a critical factor when evaluating speakers, 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 nonlinear 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. Typically, 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 nonlinear distortion factor is normalized in several frequency bands. For example, for the zero complexity group according to GOST 23262-88, the requirements of which significantly exceed minimum requirements 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 numbers, of course, characterize the system as a whole, but the phrase “THE = 1%” still says little. A striking example: tube amplifier with a non-linear distortion coefficient of about 10% can sound much better than a transistor amplifier with the same coefficient of less than 1%. The fact is that lamp distortion is mainly caused by those harmonics that are screened by 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 what the signal spectrum of a specific acoustic looks like at a reference frequency of 5 kHz

In principle, you can look at the distribution of harmonics across the spectrum using any analyzer, both hardware and software. The same programs RMAA or TrueRTA do this without any problems. As a rule, we use the first one. The test signal is generated using a simple generator; several test points are used. For example, nonlinear distortions that increase 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 timbral balance, wheeze, have extraneous sounds, etc. Also, these measurements make it possible to evaluate the acoustics in more detail in combination with other measurements, and check the correctness of the calculation of the separation filters, because the nonlinear distortions of the speaker increase greatly outside its operating range.

Article structure

Here we will describe the structure of the article on acoustic systems. Despite the fact that we try to make reading as pleasant as possible and do not squeeze ourselves into a certain framework, articles are compiled taking into account this plan, so that the structure is clear and understandable.

1. Introduction

Here we write general information about the company (if we are getting to know it for the first time), general information about the product line (if we are taking it for a test for the first time), and we give an outline of the current state of the market. If the previous options are not suitable, 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 gaming 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 performance characteristics, we make a short introduction (for example, “we can expect serious YYY parameters from acoustics costing XXX”). The table type and set of parameters are as follows:

For systems2.0

Parameter	Meaning
output power, W (RMS)
External dimensions of speakers, WxDxH, mm
Gross weight, kg
Net weight, kg
Speaker diameter, mm
Speaker resistance, Ohm
Supply voltage, V
frequency range, Hz
Frequency response unevenness in the operating range, +/- dB
Low frequency adjustment, dB
Crosstalk, dB
Signal to noise ratio, dB
Completeness
Average retail price, $

For systems2.1

Parameter	Meaning
Output power of satellites, 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 resistance, Ohm
Magnetic shielding, availability
Supply voltage, V
High frequency adjustment, dB
Low frequency adjustment, 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)
Center channel output power, W (RMS)
Subwoofer output power, W (RMS)
Total output power, W (RMS)
SOI at rated power, %
External dimensions of front satellites, WxDxH, mm
External dimensions of 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 resistance, 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
High frequency adjustment, dB
Low frequency adjustment, dB
Crosstalk, dB
Signal to noise ratio, dB
Completeness
Average retail price, $

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

3. Packaging and accessories

We describe the delivery package and box, at least two photographs. Here we evaluate the completeness of the kit, describe the nature of the cables included in the kit, and, if possible, estimate their cross-section/diameter. We conclude that the kit corresponds to the price category, convenience and packaging design. We note the presence of a Russian-language operating manual and 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 solutions in terms of potential impact on sound (remembering to add the word "allegedly"). We evaluate the quality of workmanship, the presence of legs/spikes, grill/acoustic fabric in front of the diffusers. We are looking for fastenings, the possibility of installation on a stand/shelf/wall.

Describes ergonomics and impressions of working with acoustics (excluding listening). It is noted whether there is a click when turned on, whether the wires are long enough, and whether all controls are convenient to use. Implementation of controls (analog sliders or knobs, digital encoders, toggle switches, etc.) Several photographs of controls, remote control if available, photos of speakers in a setting or in comparison with ordinary objects. Convenience and speed of switching, the need to check phasing, whether the instructions help, 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. Design

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

Type of acoustic design (open, closed box, bass reflex, passive radiating, transmission line, etc.) + general photo of the internal structure;

Dimensions and internal volume of the case, assume the compatibility of the AO with the GG;

Location of loudspeaker heads (SG), method of attachment to acoustic design;

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

Availability of mechanical damping, quality of its execution and materials used + photo;

The shape and dimensions of the bass reflex (if any), its location (estimated effect on the sound) and the manufacturer’s likely adaptations to eliminate jet noise + photo;

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

The GGs used are the type, material of manufacture (paper, impregnated silk, aluminum, plastic, etc.), the nature of the diffuser surface (conical, exponential surface, corrugated, with “stiffening ribs,” etc.) and the protective cap (flat , “acoustic bullet”, etc.), suspension (rubber, paper, etc.), degree of suspension rigidity), coil diameter, tweeter cooling, markings, resistance + photo of each GG;

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

Signal cable connectors - types, quantity, quality.

We illustrate the following with diagrams and graphs:

Amplifier chip(s) - table with key characteristics, their analysis for compliance with performance characteristics and speakers, if possible - provide a graph of power versus SOI and a photo, maybe a photo of the radiator;

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

Separation filter - we sketch the circuit, indicate the order of the filter (and, accordingly, the attenuation of the signal), and draw a conclusion about its justification; application (if appropriate measurements are available), we calculate the cutoff frequency if we subsequently measure resonance and/or Z-characteristic;

We calculate the resonant frequency of the bass reflex, present the formula and justify its use.

6. Measurements

We make the following measurements and provide an analysis for each of them, making 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 speaker dispersion;

Frequency response of the subwoofer (if any) + total frequency response of systems, quality analysis; trifonic matching, influence of bass reflex resonance;

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

Frequency response of the bass reflex, analysis;

Harmonic distortion spectrum;

Frequency response of speakers separately (for example, LF and HF), if necessary.

7. Audition

First, we give the first subjective assessment of the nature of the sound, indicating whether the volume is sufficient for various playback modes. We note the peculiarities 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 listening room, its area and volume, as well as the degree of demands of the given acoustics on 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, make a reference to the measurement result that confirmed this or that sound feature. In general, all sound analysis is done in conjunction with measurements. Be sure to pay attention to the following parameters:

The nature of the acoustics in each of the key frequency ranges, the extent to which one or another range is emphasized;

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

Detail, sound transparency, “grain” (post-pulse activity at mid and high frequencies);

The presence of color and its character in different ranges, timbral balance and natural sound;

Clarity of sound attack (impulse response) and separately - subwoofer operation (if any);

Signal saturation with harmonics (warmth or coldness of sound);

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

Optimal values tone adjustments.

Here we give a general assessment of the acoustics, first of all, the compliance of the solutions used in it with the final result and price category. It is assessed whether the acoustics are successful, promising, and suitable as a “blank” for modifications. A list of pros and cons of the system is given.

Conclusion

The assiduous reader, having completed reading this article, probably learned 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, especially, sound theory; we will leave this to specialized publications, each of which has its own view of the line where physics ends and shamanism begins. But now all aspects of acoustics testing by the authors of our portal should be extremely clear. We never tire of repeating that sound is a subjective matter, and you cannot be guided by tests alone when choosing acoustics, but we hope that our reviews will greatly help you. Have a good sound, dear readers!

IntroductionIt is unlikely that I will make a discovery by calling the topic of testing computer acoustics one of the most unpopular in the computer press. If we analyze most reviews, we can come to the conclusion that they are all purely descriptive in nature and consist, as a rule, of recompiling press releases with rewriting of the main technical parameters, admiring the body's performance, and extremely subjective final assessments, not supported by any evidence. The reason for this “dislike” is the lack at the disposal of testers of such specialized measuring instruments as audio analyzers, sensitive microphones, millivoltmeters, sound signal generators, etc. Such a set of equipment costs a lot of money, and for this reason not every test laboratory can afford it (especially that computer acoustics costs disproportionately little compared to similar measuring equipment). In addition, the tester, of course, must have the “right ears” and, preferably, have an idea of ​​​​quality sound not from his home music center, but from the sound of a symphony orchestra in the conservatory hall, for example. Be that as it may, although computer acoustics do not pretend to take the place of hi-end and delight the user’s ears with a reliable transmission of timbres, accurately conveying the emotional content of the sound picture, they should at least not distort the sound of a number of instruments and not introduce discomfort into the listener’s consciousness. Objectively, the human ear, of course, neutralizes most distortions, isolating and restoring the sound picture even from the crackle of a radio broadcast loudspeaker, but when listening to the same work on higher-quality acoustics, the listener begins to distinguish new and additional details, some musical shades (like that “...if you look with the naked eye, you can see three stars!..”). Probably for this reason too, the choice of computer acoustics should be approached more seriously and consciously.
Recently, the number of users who want to equip their computer with truly high-quality speaker systems has been steadily growing. To make the task of choosing easier for you, we decided to develop this topic on the pages of our website, and in order for the reviews not to be purely subjective in nature and not based only on the personal preferences of the author-tester, F-Center equipped the test laboratory with a special device - the PRO600S audio analyzer produced French company Euraudio. Let's look at this device in a little more detail.

Audio analyzer Euraudio PRO600S

The Euraudio PRO600S audio analyzer is a compact mobile device designed for performing electroacoustic measurements in real time. Its body is made of durable plastic, and ergonomic protrusions on the sides provide a certain comfort when working “in the field”. For stationary installation on a tripod, a special mount is provided in the bottom of the device. In general, there are quite a lot of devices with similar purposes in the world, however, the main and advantageous difference between the Euraudio PRO600S is its complete autonomy. The audio analyzer has its own battery inside, which allows you to use the device away from electrical networks (the battery charge lasts approximately four hours of battery life). An interesting fact: this particular mobile audio analyzer is adopted by car audio installers, which is why there is an option to power the device from the cigarette lighter. For stationary use, an external 12V power supply is connected to the PRO600S.
To measure acoustic parameters, either a built-in or a connected external microphone is selected in the audio analyzer settings, and for electrical measurements, a linear input is selected. The built-in microphone is used in cases where high measurement accuracy is not required (for example, during initial system setup). If the task is to take more precise parameters, or there is a need for special positioning of the microphone to the speaker, you can connect external highly sensitive microphones to the device. We have two such microphones at our disposal. The first is a microphone from Neutrik (a successful replacement for the built-in microphone), the second is a special Linearx M52 microphone designed for measuring high levels sound pressure (High-SPL Microphone). The connectors on these external microphones are AES/EBU (American Electromechanical Society/European Broadcasting Union, if I'm not mistaken) and connect to the audio analyzer's XLR connector via a special shielded adapter cable.

Neutrik microphone



High-SPL microphone Linearx M52



Jack for connecting an external microphone

The linear input of the audio analyzer allows you to measure electrical (and acoustic) circuits. This input can be connected to the line outputs of preamps, mixing consoles, CD players, equalizers, etc. The only exceptions are the outputs of power amplifiers, the high electrical potential of which can damage the electronics of the device. When making measurements using the line input, the levels are indicated on the LCD display in dBV.

Mode for measuring electrical circuits using linear input

The device is controlled using a basic on-screen menu system and a few buttons on its front panel. The five-inch monochrome LCD display has a resolution of 240x128 pixels, providing easy reading. In other cases, when the audio analyzer is not used in the field, you can connect a printer or computer to it. For this purpose, it has IEEE1284 (LPT) and RS-232 (COM) interface ports.

On the rear panel of the audio analyzer there is: line input (1), built-in microphone (2), power switch (3), connector for connecting an external power source (4), COM port (5), LPT port (6)

Source selection input signal in the Input Selection menu is made between the built-in microphone (Internal Microphone), external one-third octave microphone (1/3 Oct External Microphone), external High-SPL microphone or Line Input.

Selecting the input source

There are several measurement modes: a mode for identifying the amplitude-frequency characteristics of an acoustic system, the maximum sound pressure level, a competition mode with scoring, and a mode for measuring electrical paths. The "weighing" or "weighting" method is selected from the Weighting SPL menu, which consists of the A-weighting, C-weighting and Linear items.

Selecting a weighing method



Sound competition mode

In general terms, so as not to bore the reader theoretical material, it goes like this. The acoustic signal received by the audio analyzer from the microphone is sent to its bandpass filters, which amplify some frequencies and smooth (attenuate) others. These filters are a kind of load. There are two types of loading, which are designated by the letters “A” and “C” (A- and C-weighting). Curve "A" is determined by the approximate inverse value of 40 phon ("phon" is a unit of equivalent loudness equal to 1 decibel) of the equivalent loudness contour, and curve "C" is determined by 100 phon. Here, low frequencies are attenuated, and the frequencies of the speech range (1,000 - 1,400 Hz) are, on the contrary, amplified. Mode "L" (Linear) indicates no load.

Curves "A" and "C"

Next, I will try to explain in the most popular way the essence of measuring the frequency response.

Frequency response measurement using Euraudio PRO600S

So, the device allows you to measure the amplitude-frequency characteristics of acoustic systems by sound pressure in real time. If we take it purely hypothetically, then the process of measuring the frequency response could be organized as follows: by sequentially changing the frequency of the signal at the input, measure the current value of the sound pressure at the output. To obtain a “non-blurry” idea of ​​the shape of the frequency response, it is necessary to carry out such measurements on at least thirty segments of the frequency scale of the sound spectrum, spaced no further than a third of an octave from each other. This “manual” measurement mode will take a significant amount of time, which can only be done when testing a single speaker, and even then if you do not resort to any additional adjustments in the process (so as not to then run through all frequencies again). That is why acoustic laboratories use the method of measuring the frequency response by sound pressure in real time (RTA - Real Time Analyzing). Here, instead of separate signals, a single signal is supplied to the system input, uniformly saturated across the entire frequency spectrum of the audio range (from 20 to 20,000 Hz), which is called “pink noise”. To the ear, such a signal resembles the sound of an untuned radio or the noise of a waterfall. The acoustic system reproduces “pink noise”, which, in turn, is picked up by the microphone of the audio analyzer, after which it is sent to its bandpass filters, which cut out a narrow frequency band (each of its own) from the spectrum, the width of which is a third of an octave. For example, the first filter is set to a band from 20 to 25 Hz, the second – from 25 to 31.5 Hz, etc. Boosted signal for each band of the range is displayed on the LCD display of the audio analyzer in the form of a level column. To cover the frequency range from 20 to 20,000 Hz, thirty bandpass filters will be required. It is clear that the device indicator should display all thirty levels. Most of the Euraudio PRO600S's LCD display is taken up by these third-octave bars, covering the audio range from 25 to 20,000 Hz. On the device display, the frequency scale is displayed in logarithmic form, which corresponds to the expression of pitch in octaves proportional to the logarithm of the frequency ratio (the screen resolution is such that one pixel on the device display is equal to one decibel).
On the right side of the screen there is an indicator of the overall sound pressure level, which is designed as a level column with a digital value duplicated on top. The loading method used is indicated below this bar.

Real-time frequency response measurement mode for sound pressure

When measuring frequency response, it is possible to change the integration time, in other words, the response time of the audio analyzer to changes in the sound environment. There are three modes for this: Fast (125 ms), Slow (1 s) and Long (3 s). At any time, measurements can be paused, and the current readings of the audio analyzer will be “frozen”. Now, if you press one of the five numbered buttons, the display readings will be written to the memory cell corresponding to the button number. This option is left for transferring data from the audio analyzer to the printer.
The device comes with a CD containing the Euraudio utility program, which is quite simple. It is devoid of any analytical part and is required mainly to present test results on a computer. In addition, the program converts the readings of one-third octave filters into digital form, recording data with delimiters in text file(to convert to any known spreadsheet).

When measuring the frequency response, in order not to introduce distortion from the preamplifiers of any audio card, the speaker system under test is connected directly to the linear output of the CD player, and the “pink noise” test signal is read from a special IASCA CD.
The relative unevenness of the frequency response is determined as follows: based on the data obtained using an audio analyzer, the maximum difference between adjacent bandpass frequency filters is found, after which the difference between them is calculated. Taking into account the fact that our tests involve multimedia acoustic systems, the class of which is an order of magnitude different from the class of high-quality household audio equipment (many systems simply do not work in the range of 20 - 20,000 Hz), we decided to limit the calculation of frequency response unevenness to a segment from 50 to 15,000 Hz. Based on the frequency response unevenness indicator, we can talk about the quality of a particular acoustic system. The crossover frequency was determined visually from the measured frequency response. By the way, from the picture you can learn about the settings of the subwoofer’s bass reflex port and the tuning frequencies of the system’s bandpass filters.
The maximum sound pressure level was measured as follows: an SPL microphone is connected to the device, the appropriate measurement mode is selected from the menu, and the option to save peak values ​​is activated. Next, the SPL Competition test track is launched from the IASCA CD, which “forces” the system to operate at the highest possible acceptable values. During this stage, only the maximum achieved sound pressure level is displayed on the audio analyzer display (and remains as a peak). It is by this parameter that one can judge the ability of a particular acoustic system to “turn your insides” when listening at maximum volume levels.

Maximum sound pressure level measurement mode

At the end of the testing, some measurement results were recorded in a table, looking at which it is quite easy to understand which system deserves attention. So, taking measurements using an audio analyzer allows us to judge the maximum sound pressure level, the relative unevenness of the frequency response, crossover frequencies and the actual range of reproduced frequencies by the acoustic system. Using the last parameter, you can check the discrepancies between the characteristics declared by the manufacturer and those obtained by us.

Impedance measurement

The audio analyzer, as I already said, is equipped with a linear input, designed in the form of an RCA connector. Thanks to this, the device allows you to go beyond just acoustic tests by measuring the sound pressure level when receiving data from a microphone. With this line-in input you can connect against electrical circuit speaker system and measure (approximately, of course), for example, impedance and harmonic distortion.
Impedance is very useful feature, with which you can check the speaker’s ability to operate correctly when this level gain and note the resonant frequencies of the woofer. To carry out the measurement, a “pink noise” test signal is applied to the input of the speaker amplifier. Take a look at the figure below: The amplifier should not be bridged (i.e. its negative pole should be common ground). 4 and 8 ohm resistors are used for calibration. First, a 4 Ohm resistor is selected and the volume is increased until readable signal levels appear on the audio analyzer display (usually this level is a straight line). After this, the 8 Ohm mode is selected and the levels are set for it. The switch is then set to test the speaker, and by comparing the two lines, its impedance across the entire acoustic range is estimated, finding its resonant frequency (or frequencies).

Impedance measurement circuit

Note: unfortunately, at the moment we did not have time to prepare a stand for impedance determination, so the results for this stage will be available a little later.

IASCA Competition Audio Test CD

Let me start with the fact that in the late 70s, acoustics manufacturers deliberately tried to draw analogies between audio equipment and... irons, very actively introducing into the minds of consumers sets of technical requirements, the fulfillment of which would guarantee (supposedly) the highest sound quality of the equipment. Even then, manufacturers who tried to rely only on objective parameters were called “objectivists”. However, in the early 80s, they were all disappointed in the form of a drop in demand and a general decline in sales volumes for audio equipment, despite the fact that “objective parameters” were constantly improving, and for some reason the sound quality, on the contrary, was getting worse. This general trend gave impetus to the birth of the subjectivist movement, whose slogan shocked many orthodox people: “If there are contradictions between objective parameters and subjective assessments, then the result of objective measurements should not be taken into account.” However, by today's standards, the then slogan of the subjectivists turned out to be quite balanced. Although auditory perception can fail us, it is nevertheless the most sensitive tool for assessing sound quality. The assessment itself cannot be given without listening to various test musical compositions (symphonic and instrumental music, boys' choir and famous tenor, jazz and rock compositions), so many record companies have developed special collections, like the one about which further narration.
Our test music disc can be called universal. It is used both to determine objective parameters (some tracks are used as a test signal source) and to construct subjective listening assessments. This is an IASCA Competition CD from a fairly well-known international association International Audio Sound Challenge Association.

There are 37 audio tracks on this disc, and some tracks are annotated in nature, bringing to the listener's attention what to pay attention to when listening. By the way, information about this disc is in the CDDB database, so after installation in the computer's CD player, the titles of all its tracks are downloaded from the Internet. The order in which records are placed on the disk is subject to a certain law, i.e. phonograms are divided into groups according to the sound characteristics being assessed (tonal purity, spectral balance, sound stage, etc.). Many recordings are taken from renowned music archives such as Telarc, Clarity, Reference, Sheffield and Mapleshade. Below is the IASCA Competition CD track list.

IASCA Competition CD playlist

Today you can find speakers of almost any shape. But how does this affect the sound? Let's consider the main forms of acoustic systems, and why round column will sound better than square or cylindrical.

To the final A amplitude - H frequency X characteristics ( frequency response) A bushy C systems ( AC) is influenced by many factors. Including the frequency response of the speaker, its quality factor, the selected type and material of the housing, damping, etc. etc. But today we will consider another interesting nuance that makes its own adjustment to the final frequency response - speaker system shape.

What is affected by the shape of the AS?

In itself, the shape of the speaker from the outside is not particularly important; the important thing is that it determines the shape of the internal volume of the speaker. On low frequencies, in which the linear dimensions of the body are less than the wavelength of sound, the shape of the internal volume does not matter, but at medium frequencies diffraction effects make a significant contribution. For simplicity, a closed acoustic design is assumed below.

Under diffraction effects This implies mutual amplification and damping of sound waves inside the speaker. The frequency response of speakers is negatively affected by sharp corners, depressions and protrusions, i.e. Maximum unevenness of the sound field is observed on them. But rounding and leveling have a positive effect on the shape of the frequency response. To be more precise, more rounded shapes have minimal impact on the linearity of the frequency response.

Cylindrical speakers frequency response

The worst results are obtained by a body in the form of a horizontal cylinder (Fig. a)
(The position of the center of the emitting head is conventionally depicted by a dot).

The unevenness of the speaker's frequency response reaches 10 dB at the first maximum (~500Hz). This is due to the fact that the wavelength corresponds (equal) to the linear dimensions of the body. The following maximums correspond to doubled, tripled, etc. frequencies. This picture arises due to the contribution of the front panel (on which the emitter is located). Reflection occurs between the front and rear panels, resulting in an interference pattern between them.

For this reason, a cylinder-shaped speaker with a dynamic head on the side panel (Fig. b) has a more uniform frequency response. Front panel in in this case creates a scattered field in the internal volume, and the upper and lower walls have little effect, because are not on the same axis with the emitter.

Round column and square column

The cubic-shaped housing (Fig. c) also creates a highly uneven frequency response, because an interference pattern also appears.

The most minimal impact The shape of the frequency response is influenced by spherical acoustics (Fig.d). In a housing of this shape, sound dissipation occurs equally in all directions.

However, making a round column is a rather labor-intensive process. Although the use of modern materials such as plastics simplifies this task, plastic is still not the best material for the body of a high-quality speaker system.

A positive result is obtained by using mastics and similar materials, the application of which to corners and joints leads to their rounding and linearization of the frequency response of the speakers. Also, to improve the frequency response, damping of the internal volume of the speaker system is used.

Even spherical acoustics, which have the best frequency response, have a decline in the low-frequency region. Most effective solution this problem may become .