A round speaker sounds better than a square speaker. Acoustic measurements. We measure the frequency response with improvised means What is the frequency response measured

Today you can find columns of almost any shape. But how does this affect the sound? Let's look at the basic shapes of loudspeakers, and why a round speaker will sound better than a square or cylindrical one.

For the final A amplitude - H atomic X characteristic ( frequency response) A bushy C systems ( AC) is influenced by many factors. This is the frequency response of the speaker, its quality factor, the selected type and material of the case, damping, etc. etc. But today we will consider another interesting nuance that makes adjustments to the final frequency response - shape of the acoustic system.

What is the shape of AS

By itself, the shape of the column on the outside does not really matter, the important thing is that it determines the shape of the internal volume of the speaker. At low frequencies, the linear dimensions of the body are smaller than the wavelength of sound, so the shape of the internal volume does not matter.

But at medium frequencies, diffraction effects make a significant contribution. For simplicity, the following refers to a closed acoustic structure.

By diffraction effects is meant the mutual amplification and damping of reflected and direct sound waves inside the column.

Sharp corners, depressions and protrusions adversely affect the frequency response of speakers. On them, the unevenness of the sound field is maximum.

But rounding and leveling have a positive effect on the shape of the frequency response. To be more precise, more rounded shapes have minimal effect on the linearity of the frequency response.

Cylindrical loudspeakers

The worst the results are given by a case in the form of a horizontal cylinder ( rice. A )

The position of the center of the radiating head is conventionally depicted by a dot.

The uneven frequency response of the column shown in figure a reaches 10 dB at the first maximum (~500Hz). This is due to the fact that the wavelength is comparable to the linear dimensions of the case. The next highs correspond to double, triple, and so on. frequencies.

This pattern arises due to the reflection between the front ( with speaker) and the rear walls of the case. This leads to the appearance of an interference pattern between them. The specific frequencies of the maxima and minima depend on the actual dimensions of the column.

Speaker shaped like a cylinder, but with a dynamic head on the side panel ( rice. b) has a more uniform frequency response. The front panel in this case creates a scattered field in the internal volume. The upper and lower walls have little effect, because are not on the same axis as the emitter.

Round column and square column

Cubic body ( rice. V) also creates a highly uneven frequency response. In this case, a close interference pattern arises.

Spherical acoustics has the most minimal effect on the frequency response shape ( rice. G). In a case of this shape, sound scattering occurs equally in all directions.

However, the manufacture of a round column is a rather laborious process. Although the use of modern materials such as plastics makes this task easier.

But still, plastic is not the best material for a high-quality speaker cabinet.

How to improve the sound of a non-circular speaker

A positive result is the use of mastics. If such materials are applied to corners and joints, this will lead to their rounding. Thanks to this, the frequency response of the speakers will become linear.

Also, to improve the frequency response, damping of the internal volume by absorbing materials is used. They dampen excess sound waves, so there are fewer reflections.

Even spherical acoustics, which have the best frequency response, have a decline in the low-frequency region. The most effective solution to this problem would be .

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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.

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, 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 harmonic distortion (THD) values ​​of 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 has practically no effect on 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 common 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.

It is known that dynamic processes can be represented by frequency characteristics (FC) by expanding the function into a Fourier series.

Suppose there is some object and it is required to determine its frequency response. During the experimental removal of the frequency response, a sinusoidal signal with amplitude A in = 1 and a certain frequency w is fed to the input of the object, i.e.

x (t) \u003d A in sin (wt) \u003d sin (wt).

Then, after passing through transients at the output, we will also have a sinusoidal signal of the same frequency w, but of a different amplitude A out and phase j:

y(t) = A out sin(wt + j)

For different values ​​of w, the values ​​of A out and j, as a rule, will also be different. This dependence of amplitude and phase on frequency is called the frequency response.

Types of frequency response:

·

y” “s 2 Y etc.

Let's define the derivatives of the frequency response:

y'(t) = jw A out e j (w t + j) = jw y,

y”(t) = (jw) 2 A out e j (w t + j) = (jw) 2 y, etc.

This shows the correspondence s = jw.

Conclusion: frequency responses can be built from transfer functions by replacing s = jw.

To build the frequency response and phase response, the following formulas are used:

, ,

where Re(w) and Im(w) are the real and imaginary parts of the expression for the AFC, respectively.

Formulas for obtaining AFC by AFC and PFC:

Re(w) = A(w) . cos j(w), Im(w) = A(w) . sinj(w).

The frequency response graph is always located in one quarter, because frequency w > 0 and amplitude A > 0. The PFC graph can be located in two quarters, i.e. phase j can be either positive or negative. The AFH schedule can run through all quarters.

When graphing the frequency response according to the known AFC, several key points corresponding to certain frequencies are highlighted on the AFC curve. Next, the distances from the origin of coordinates to each point are measured and the frequency response graph is plotted: vertically - measured distances, horizontally - frequencies. The construction of the AFC is carried out in a similar way, but not distances are measured, but angles in degrees or radians.

For graphical plotting of the AFC, it is necessary to know the type of AFC and PFC. At the same time, several points corresponding to certain frequencies are allocated on the frequency response and phase response. For each frequency, the amplitude A is determined by the frequency response, and the phase j is determined by the phase response. Each frequency corresponds to a point on the AFC, the distance to which from the origin is A, and the angle relative to the positive semiaxis Re is equal to j. The marked points are connected by a curve.

Example: .

For s = jw we have

= = = =

Introduction It is unlikely that I would make a discovery by naming the topic of testing computer acoustics as one of the most unpopular in the computer press. If we analyze most of the reviews, we can come to the conclusion that all of them are purely descriptive in nature and consist, as a rule, of recompiling press releases with rewriting the main technical parameters, admiring the hull performance, and extremely subjective final assessments that are not supported by any evidence. The reason for this “dislike” is the lack of specialized measuring instruments at the disposal of testers, such as audio analyzers, sensitive microphones, millivoltmeters, audio signal generators, etc. Such a set of equipment costs decent 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 "correct ears" and, preferably, have an idea of ​​high-quality sound not from his domestic 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 please the user's ear with a reliable transmission of timbres, accurately conveying the emotional content of the sound picture, it should at least not distort the sound of a number of instruments, not bring discomfort to the mind of the listener. Objectively, the human ear, of course, levels out most of the distortions, isolating and restoring the sound picture even from the crackle of the speaker of a radio broadcasting loudspeaker, however, when listening to the same work on better 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, and 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 really high-quality acoustic systems has been steadily growing. To make it easier for you to choose, we decided to develop this topic on the pages of our website, and in order for the reviews not to be purely subjective, 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 manufactured by French firm Euraudio. Let's take a closer look at this device.

Audio analyzer Euraudio PRO600S

The Euraudio PRO600S audio analyzer is a compact mobile device designed for performing electro-acoustic 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 similar devices 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, allowing you to use the device away from electrical mains (the battery charge is enough for approximately four hours of battery life). An interesting fact: it was this mobile audio analyzer that was adopted by the installers of car acoustics, which is why the option of powering the device from the cigarette lighter is provided. For stationary use, an external 12V power supply is connected to the PRO600S.
To measure acoustic parameters in the settings of the audio analyzer, either the built-in or a connected external microphone is selected, and for electrical measurements, a line input is selected. The built-in microphone is used in cases where high measurement accuracy is not required (for example, when setting up the system for the first time). If the task is to take more accurate parameters, or there is a need for a special positioning of the microphone to the speaker speaker, external highly sensitive microphones can be connected to the device. We have two such microphones at our disposal. The first is a Neutrik microphone (a successful replacement for the built-in microphone), the second is a special Linearx M52 microphone designed to measure high sound pressure levels (High-SPL Microphone). The connectors of these external microphones comply with the AES / EBU standard (if I'm not mistaken, these are abbreviations from the American Electromechanical Society / European Broadcasting Union) and are connected to the XLR connector of the audio analyzer via a special shielded adapter cable.

Neutrik microphone



Linearx M52 high-SPL microphone



Connector for external microphone

The audio analyzer's line input allows measurements of electrical (and acoustic) circuits. This input can be connected to the line outputs of preamplifiers, 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 measuring with the line input, the LCD displays the levels in dBv.

Mode of measurement of electrical circuits by linear input

The device is controlled using an elementary on-screen menu system and a few buttons on its front panel. The 5-inch monochrome LCD display has a resolution of 240x128 dots for easy reading. In other cases, when the audio analyzer is not being used "in the field", a printer or computer can be connected to it. To do this, it has interface ports IEEE1284 (LPT) and RS-232 (COM).

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

The input source selection in the Input Selection menu is between the Internal Microphone, 1/3 Oct External Microphone, High-SPL External Microphone, or Line Input.

Selecting an input source

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

Selecting the weighing method



Sound competition mode

In general terms, in order not to bother the reader with theoretical material, it happens like this. The acoustic signal received by the audio analyzer from the microphone is sent to its band-pass filters, which are engaged in amplifying some frequencies and smoothing (attenuating) others. These filters are sort of loads. There are two types of loading, which are denoted 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 (1000 - 1400 Hz), on the contrary, are amplified. Mode "L" (Linear) means no loading.

Curves "A" and "C"

Next, I will try to most popularly state the essence of measuring the frequency response.

Frequency response measurement with 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 itself could be organized as follows: by successively changing the frequency of the signal at the input, measure the current value of the sound pressure at the output. To obtain a "not blurred" idea of ​​the shape of the frequency response, you need to make such measurements at least thirty segments of the frequency scale of the sound spectrum, spaced from each other no further than a third of an octave. Such a "manual" measurement mode will take a considerable amount of time, which can only be allowed when testing a single speaker, and even then, if you do not resort to any additional adjustments in the process (so as not to roll over all frequencies again). That is why acoustic laboratories use the method of measuring frequency response by sound pressure in real time (RTA - Real Time Analyzing). Here, instead of separate signals, a single signal is fed to the input of the system, uniformly saturated over 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 receiver or the sound 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 band-pass filters, which cut out a narrow frequency band (each with its own) from the spectrum, the width of which is a third of an octave. For example, the first filter is set to 20 to 25 Hz, the second is set to 25 to 31.5 Hz, and so on. The amplified signal for each band of the range is displayed on the LCD display of the audio analyzer in the form of a level bar. Thirty bandpass filters would be required to cover the frequency range from 20 to 20,000 Hz. It is clear that the indicator of the device should display all thirty levels. Most of the LCD on the Euraudio PRO600S is occupied by these one-third octave bars covering the audio range from 25 to 20,000 Hz. On the display of the device, the frequency scale is displayed in a logarithmic form, which corresponds to the expression of the pitch in octaves in proportion to the logarithm of the frequency ratio (the screen resolution is such that one pixel on the device display equals one decibel).
To the right of the screen 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 displayed under this column.

Real-time sound pressure frequency response measurement mode

When measuring the frequency response, it is possible to change the integration time (Integration Time), in other words, the response time of the audio analyzer to a change 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 suspended, 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 possibility is reserved for data transfer from the audio analyzer to the printer.
The device is supplied with a CD with the Euraudio utility program, which is quite simple. It is devoid of any analytical part and is required mainly for presenting test results on a computer. In addition, the program converts the readings of one-third-octave filters into digital form, writing data with separators in a text file (for conversion 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 line output of the CD player, and the "pink noise" test signal is read from a special IASCA CD.
The determination of the relative unevenness of the frequency response is carried out as follows: based on the data obtained using the audio analyzer, the maximum difference between adjacent band-pass filters is found, after which the difference between them is calculated. Taking into account the fact that multimedia acoustic systems take part in our tests, the class of which differs by an order of magnitude from the class of high-quality consumer audio equipment (many systems simply do not work in the range of 20 - 20,000 Hz), we decided to limit the calculation of the frequency response unevenness to a segment from 50 to 15,000 Hz. Based on the unevenness of the frequency response, we can talk about the quality of a particular speaker system. The frequency of the section was determined visually, according to the removed frequency response. By the way, from the picture you can also find out about the settings of the subwoofer phase inverter port and about the tuning frequencies of the system's band-pass filters.
The measurement of the maximum sound pressure level was carried out as follows: an SPL microphone was connected to the device, the corresponding measurement mode was selected from the menu, and the option to save peak values ​​was activated. Next, the SPL Competition test track is launched from the IASCA CD, which "forces" the system to work at the maximum possible allowable values. During this stage, the audio analyzer displays (and remains as a peak) only the maximum sound pressure level achieved. It is by this parameter that one can judge the ability of a particular speaker system to "turn your insides" when listening at maximum volume values.

Maximum sound pressure level measurement mode

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

Impedance measurement

The audio analyzer, as I said, is equipped with a line-in, designed as an RCA connector. Thanks to this, the device allows you to go beyond acoustic tests by measuring the sound pressure level when receiving data from a microphone. Using this line input, you can connect across the electrical circuit of the speaker system and measure (approximately, of course), for example, impedance and harmonic distortion.
Impedance is a very useful feature that can be used to test the speaker's ability to operate correctly at a given gain level and note the resonant frequencies of a subwoofer. To carry out the measurement, a "pink noise" test signal is applied to the input of the acoustic system amplifier. Take a look at the figure below: the amplifier must not be bridged (ie its negative pole must be common ground). 4 and 8 ohm resistors are used for calibration. First, a 4 ohm resistor is selected, the volume is increased until readable signal levels appear on the audio analyzer display (usually such a level is a straight line). After that, 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 over the entire acoustic range is estimated and the resonant frequency (or frequencies) is found.

Impedance measurement circuit

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

IASCA Competition CD Audio Test CD

To begin with, at the end of the 70s, acoustic manufacturers deliberately tried to draw analogies between audio equipment and ... irons, very actively introducing sets of technical requirements into the minds of consumers, the fulfillment of which guarantees (allegedly) 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 for audio equipment, despite the fact that the "objective parameters" were constantly improving, and the sound quality, for some reason, 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 evaluating sound quality. The assessment itself cannot be given without listening to various test musical compositions (symphonic and instrumental music, boys' choir and the 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 build subjective listening scores. 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, bringing to the listener what should be paid attention to when listening. By the way, information about this disc is in the CDDB database, so after installation in a computer CD player, the titles of all its tracks are downloaded from the Internet. The order in which records are placed on a disk obeys a certain law, i.e. phonograms are divided into groups according to the estimated sound characteristics (tonal purity, spectral balance, sound stage, etc.). Many of the recordings are from renowned music archives such as Telarc, Clarity, Reference, Sheffield and Mapleshade. Below is the track listing of the IASCA Competition CD.

IASCA Competition CD playlist