Useful tips. Power: how many watts does the speaker need. Experience in improving the parameters of a high-frequency speaker When operating an amplifier with a wideband speaker, you have to look for some kind of compromise value for the recovery time

If you ask me why this is necessary, then I will not answer you - then this article is not for you. If everything is in order with your motivation, then I offer you to familiarize yourself with some of the results I obtained with the modest means and knowledge that I have.

To begin with, the guinea pig, who is he?

Our patient is a high-frequency speaker with a 3GD-31 cone diaphragm. The main complaint against it is the significant unevenness and unevenness of the frequency response. Those. In addition to the unevenness of about 10 dB between the maximum peak and trough, there are many smaller irregularities, as a result of which the frequency response is similar to a forest. I decided not to present the measured characteristics at the beginning of the article, because... It will be more clear to place them next to the final ones obtained after all design changes.
The main idea of ​​my actions, or rather two main ideas, is, firstly, to add sound-absorbing elements inside the volume of the speaker in order to suppress resonances that arise in a closed volume with solid walls that easily reflect sound without noticeably absorbing its energy, which is the body of the said speaker. The second idea is to process the diffuser material itself (no, not with A. Vorobyov’s liquid ;-)), but with varnish, resulting in a composite material that is superior to the original (paper) in rigidity, but is not inferior to it in damping its own resonances, which reduces bending deformation of the diffuser during its operation and thereby helps to reduce resonant peaks and dips in the frequency response.

What's going through my head?

The fact is that I have been conducting similar experiments for a long time and have received quite a lot of confirmation of the correctness and usefulness of my approach, but all the results were rather scattered. This was partly a consequence of a lack of experience in acoustic measurements (and more in the interpretation of the results obtained), partly a consequence of the incomplete development of the idea itself and the general plan of action. And so, when this whole mosaic formed in my head into a more or less complete picture, I decided to conduct the experiment from beginning to end, simultaneously making all the measurements.

So what was done?

To begin with, the speaker was disassembled. To do this, the speaker coil leads were unsoldered from the terminals on the case, then, after soaking with acetone, the cardboard sealing ring was separated and the diffuser itself was peeled off from the metal “funnel” of the case in the same way. Next, the diffuser was removed from the housing and put aside for now.
First, the speaker housing was processed. Sectors were cut out of cloth about 3 mm thick, precisely covering the inner surface of the body, which was a truncated cone. At the bottom (the smaller base of the truncated cone) a circle was cut from the same material with a hole in the middle for the coil. After this, the inner surface of the body and the surface of the cloth blanks were smeared with one layer of Moment glue and almost immediately (since it dries very quickly and when I finished spreading the cloth patterns, the layer on the body had already dried) pressed against each other. Here is a photo of the resulting semi-finished product.

At that moment, the idea occurred to me that not only resonances in the volume of the case, but also in the walls themselves, could be to blame for a broken frequency response. the body is a kind of bell made of stamped sheet metal. To measure its resonances, I used the following technique. Having placed the case on a soft base, with the magnet facing down, I installed the microphone directly above it, turned on the sound recording and hit the outside of the case several times with the plastic handle of a screwdriver. Then I selected the most successful signal (in terms of level) from the recording and imported it into LspLab for analysis. Results a little later. Then, in order to dampen the body, it was covered on the outside with rubber from an ancient bicycle inner tube, using the same technology as the previous cloth covering. Then, after complete drying - a day later, tests were carried out again, using the same method as above. However, the sound from the impact was much weaker, so I automatically hit a little harder than during the first measurement - because of this, the signal level during the second measurement, in my opinion, turned out to be somewhat overestimated, but this does not play a significant role in this case . So, here are the first comparative results - the transient response of the speaker cabinet (in the form of a sonogram). Below is the original version.

It is clearly visible that after the modification, all resonances above 3 kHz were suppressed by a level of more than 20 dB! From this image it seems that the main resonance at 1200Hz (by the way, interestingly, the main resonance of the speaker cone is located exactly at the same frequency) has become much stronger. This is not true, because the program normalizes the levels on the sonogram so that the “strongest” signals become red, but this scale is valid only within one graph, and there are two of them in the image, so the red on the top graph is 20 dB weaker than the red on the bottom graph! Here is another - already more familiar graph - the frequency response of both measurements.

It can be seen that the damping efficiency increases with frequency and the suppression at frequencies of 3 kHz and higher exceeds 30 dB! And this despite the fact that, as I already said, in the second dimension I hit the body harder! To you, those who like to “calm down” speaker boxes, a note - I give it to you!

The diffuser was coated (not impregnated, but coated) with nitro varnish (of all the materials tested for this purpose, it had the best effect on the properties of the speakers). On the inside there is only one layer, on the outside there are three. But, of course, these were not the kind of layers that are not painted on walls! When applying the first layer with a soft brush, the surface is only moistened, and not much. The second and third layers are a little thicker, but in total, the three layers are so thin that the fibrous structure of the paper is still visible underneath them.

Before assembly, a “donut” of cotton wool was additionally inserted into the cavity between the body and the diffuser in order to achieve maximum sound absorption in the volume if possible. The following picture shows the body prepared for assembly.

Another change was made to the coil terminals. Initially, the thin wires of the coil winding itself were soldered to copper rivets on the diffuser (and hefty drops of solder were soldered!), which should create a new resonant system from the mass of all this metal and the rigidity of the part of the diffuser on which it was all stuck. I didn’t like this state of affairs at all, so I decided to redo everything. I unsoldered the coil wires from the rivets, drilled them out and soldered the leads connecting the coil to the external terminals directly to the voice coil wires. The next picture, although not of very good quality, shows the new state of affairs. The remaining holes are sealed with paper circles.

Now I will give the summary result.

To begin with, here is the frequency response of the original speaker and it after the modification. The bold lines show the frequency response and frequency response after modification.

At first glance, I did not achieve much success. Well, the dip at 4 kHz decreased by about 3 dB, the peak at 9 kHz decreased by a couple of dB, and the frequency response leveled off from 12 to 20 kHz. It can be attributed to random phenomena - the resonances in the diffuser were successfully redistributed. However, it should be said that this speaker was not very successful for the purposes of my experiment - it initially had almost the maximum quality for its design. For comparison, I will give a similar pair of frequency response for another sample - worse.

This is where all the miraculous effects of improvement on the face are! However, I am not using this speaker as the basis for this article because in this case this is all the data that I received, but I collected more information on the speaker described above.

Now I want to give the transient characteristics of the speaker. They are the same as for the body - in the form of sonograms, in my opinion this is more clear.

It is clearly visible that the original speaker has delayed resonances in the region of 5 and 10 kHz, reaching a duration of up to 1.3 ms. After modification, firstly, they are shortened by 1.5 times, and secondly, they crumble into many smaller ones, both in intensity and duration. Above 10 kHz they are not there at all - they have disappeared. In general, the impulse response has improved much more noticeably than the frequency response.
Based on this experiment, as well as several previous ones, I came to the conclusion that the varnish coating mainly affects the performance of the speaker in the highest frequency range, and various sound-absorbing materials work in the mid-frequency range.
Body damping did not appear to have a significant effect on the results.

In conclusion, I want to say that this article was written mainly with the purpose of introducing people who do not have the means of instrumental assessment of the objective parameters of speakers to the influence that specific actions have on a specific sample speaker.
As a result of these experiments, another idea arose to further improve the parameters. It will be the basis for further experiments and, if they are successful, the topic of the next such article.

There are many different types of sound emitters, but the most common are electromagnetic type emitters, or as they are also called, speakers.

Speakers are the main structural elements of acoustic systems (AS). Unfortunately, one speaker is not capable of reproducing the entire audible frequency range. Therefore, for full-range reproduction in acoustic systems, several speakers are used, where each is designed to reproduce its own frequency band. The operating principles of low-frequency (LF) and high-frequency (HF) speakers are the same; the differences lie in the implementation of individual structural elements.

The operating principle of the speaker is based on the interaction of an alternating magnetic field created by a current flowing through the wire of a magnetic coil with the magnetic field of a permanent magnet.

Despite the comparative simplicity of the design, speakers intended for use in high-quality acoustic systems have a large number of important parameters on which the final sound of the acoustic system depends.

The most important indicator characterizing a speaker is the reproduced frequency band. It can be indicated as a pair of values ​​(lower limit and upper limit frequency), or given in the form of an amplitude-frequency response (AFC). The second option is more informative. The frequency response is a graphical dependence of the sound pressure level created by a speaker at a distance of 1 meter along the working axis on frequency. The frequency response allows you to evaluate the frequency distortions introduced by the speaker into the original signal, and also, in the case of using the speaker as part of a multi-band system, to identify the optimal value of the crossover filter frequency. It is the frequency response that allows a speaker to be classified as low-frequency, mid-frequency or high-frequency.

Selecting a subwoofer

For LF speakers, in addition to the frequency response, an essential group of indicators are the so-called Thiel-Small parameters. Based on them, the acoustic design parameters for the speaker (speaker system housing) are calculated. The minimum set of parameters is resonant frequency - fs, total quality factor - Qts, equivalent volume - Vas.

The Thiel-Small parameters describe the behavior of the speaker in the piston-action region (below 500Hz), considering it as an oscillating system. Together with the acoustic design (AO), the speaker is a high-pass filter (HPF), which allows the use of mathematical tools borrowed from filter theory in calculations.

An assessment of the Thiel-Small values ​​of the speaker parameters, and first of all, the total quality factor Qts, allows us to judge the advisability of using the speaker in acoustic systems with one or another type of acoustic design (AO). For speakers with phase-inverted acoustic design, speakers with a total quality factor of up to 0.4 are mainly used. It is worth noting that phase-inverted systems are the most demanding, from a design point of view, compared to speakers that have a closed and open AO. This design is sensitive to errors made in calculations and in the manufacture of the housing, as well as when using unreliable values ​​for the parameters of the woofer.

When choosing a woofer, the Xmax parameter plays an important role. Xmax shows the maximum permissible displacement of the cone, at which a constant number of turns of voice coil wire is maintained in the gap of the speaker magnetic circuit (see figure below).

For satellite speaker systems, speakers with Xmax = 2-4mm are suitable. For subwoofers, speakers with Xmax=5-9mm should be used. At the same time, the linearity of the conversion of electrical vibrations into acoustic ones at high powers (and, accordingly, large vibration amplitudes) is maintained, which manifests itself in more efficient radiation of low frequencies.

If you have decided to make a speaker system with your own hands, you will inevitably be faced with the question of choosing branded components, including the frequency of the speakers. Without experience in using products from different manufacturers, it is sometimes difficult to make the best choice. You have to be guided by many factors and compare according to many parameters, not only those related to passport characteristics. ACTON speakers will successfully complement your speaker system because, in addition to high quality, they have a number of advantages:

• have an optimal price/quality ratio in their segment;
• the speakers are specially designed for professional speakers used for dubbing social and cultural events;
• documentation for the manufacture of housings has been developed for speakers;
• interaction between the consumer and the manufacturer is carried out directly without intermediaries, which avoids problems with the availability of any spare parts and components;
• information support on the design of speakers;
• high reliability of ACTON speakers.

You can familiarize yourself with the model range of ACTON speakers.

Selecting a tweeter

When choosing a tweeter, the frequency response determines the lower frequency of the range it reproduces. It is necessary that the frequency band of the tweeter somewhat overlaps the frequency band of the woofer.

Some tweeters are designed to work in conjunction with a horn. Unlike direct-radiation tweeters (or tweeters, as they are called), horn tweeters, due to the properties of the horn, have a lower cutoff frequency of the reproduced audio range. The lower limiting frequency of such a high-frequency speaker can be approximately 2000-3000 Hz, which makes it possible in many cases to abandon the midrange speaker in the speaker system.

Due to their design, tweeters tend to have higher sensitivity than woofers. Therefore, at the filter design stage, an attenuator (suppressor) circuit is provided in it, which is necessary to reduce excess radiation, which brings the sensitivity values ​​of the high-frequency and low-frequency speakers to the same level.

When choosing a tweeter, it is important to consider its power, which is selected based on the power of the woofer. In this case, the power of the HF speaker is taken lower than the power of the LF speaker, which follows from the analysis of the spectral density of the audio signal, corresponding to pink noise (which has a decline towards high frequencies). For a practical calculation of the power dissipated by the high-frequency dynamics in speakers with a crossover frequency of 3-5 kHz, you can use the calculator on our website.

Let us remind you that HF ​​speakers cannot be used without a high-pass filter (HPF), which limits the penetration of the low-frequency part of the spectrum.

Speaker Damage Factors

In the event of abnormal operating conditions, mechanical and electrical damage to the speakers is possible. Mechanical damage occurs when the amplitude of diffuser vibrations exceeds the permissible amplitude, which depends on the mechanical properties of the elements of the moving system. The most critical frequency zone for such damage is near and below the mechanical resonance frequency of the speaker, i.e. where the amplitude of oscillations is maximum. Electrical damage occurs as a result of irreversible overheating of the voice coil. The most critical frequency band for damage of this kind corresponds to the band located near the electro-mechanical resonance of the speaker. Both types of damage occur as a result of exceeding the maximum permissible electrical power supplied to the speaker. In order to avoid such consequences, the maximum power value is standardized.

There are several standards, using which manufacturers normalize the power of their products. The closest from the point of view of real conditions in the case of using an acoustic system for sounding public events is the AES standard. Power according to this standard is defined as the square of the rms voltage in a certain pink noise band that the speaker can withstand for at least 2 hours, divided by the minimum impedance value Zmin. The standard regulates the presence of the speaker in “free air” without a housing. When testing, some manufacturers place the speaker in a housing, thus bringing its operating conditions closer to real conditions, which, from their point of view, leads to more objective results. The known power value of the speaker serves as a guide when choosing an amplifier whose power should correspond to the power value of the AES speaker.

It is worth noting that the real value of the power supplied to the speaker is difficult to estimate without special measurements and can vary widely even with the same setting of the volume control on sound path devices.

This can be influenced by many factors, such as:

• Spectrum of the reproduced signal (musical genre, frequency and dynamic range of the musical work, predominant musical instruments);
• Characteristics of passive filter circuits and active crossovers that limit the spectrum of the original signal entering the speakers;
• Using an equalizer and other frequency correction devices in the audio path;
• Amplifier operating mode (appearance of nonlinear distortion and clipping);
• Acoustic system housing design;
• Amplifier malfunction (the appearance of a constant component in the spectrum of the amplified signal)

The following measures increase the reliability of operation of speaker systems:

• Reducing the upper limit frequency of the woofer speaker using a low-pass filter (LPF). In this case, the part of the signal spectrum that makes a significant contribution to heating the coil is limited;
• Limits the frequency band below the bass reflex tuning frequency using LOW-PASS (high-pass filter) circuits. This measure limits the amplitude of vibrations of the diffuser outside the operating range of the speakers on the low-frequency side, preventing mechanical damage to the woofer;
• Adjusting the high-frequency high-frequency speaker to a higher frequency;
• Design of speaker enclosures that provide the best conditions for natural convection of speakers;
• Elimination of operation of speakers with an amplifier operating in nonlinear distortion and clipping mode;
• Preventing the occurrence of loud switching clicks, “winding up” of the microphone;
• Using a limiter in the audio path.

Note that speaker systems that are used for professional sound recording (especially in discotheques) are often forced to operate at high power. During operation, the heating of the speaker voice coil can reach 200 degrees, and the elements of the magnetic circuit - 70 degrees. Long-term operation at extreme conditions leads to the fact that the speakers “burn”. This may be caused by exceeding the permissible electrical power supplied to the speaker, or by a faulty amplifier. In many ways, the safety of the set depends on the qualifications of the DJ. Because of this, no matter which speaker you choose, you need to consider the availability of repair kits. At the same time, the situation is further complicated by the fact that, as a rule, not one speaker burns out at the same time, but several, which disables the entire set. Considering all of the above, we conclude that the question of the timing and cost of delivery of repair kits is also extremely important at the stage of selecting speakers for speakers.

Harmonic theory

Amplitude compression

What to do?

Overloading (clipping) power amplifiers- a common occurrence. This article discusses overload caused by an increased input signal level, which results in the output signal being limited.

Having analyzed the “phenomenon” of this kind of overload, which allegedly causes damage to the speakers, we will try to prove that the true culprit is amplitude compression (compression) of the signal.

WHY DO LOUDSPEAKERS NEED PROTECTION?

All loudspeaker heads have operating power limits. Exceeding this power results in damage to the loudspeakers (LS). These damages can be divided into several types. Let's take a closer look at two of them.

The first type is excessive displacement of the GG diffuser. The GG diffuser is a radiating surface that moves as a result of an applied electrical signal. This surface can be conical, domed or flat. The vibrations of the diffuser excite vibrations in the air and emit sound. According to the laws of physics, in order to produce a louder sound or reproduce lower frequencies, the diffuser must oscillate with a larger displacement amplitude, while approaching its mechanical boundaries. If it is forced to move further it will result in excessive deflection. This most often occurs with low-frequency GGs, although it can happen with mid-frequency and even high-frequency GGs (if the low frequencies are not limited enough). Thus, excessive displacement of the diffuser most often leads to mechanical damage to the head.

The second enemy of the GG is thermal energy resulting from thermal losses in the voice coils. No device is 100% efficient. As for the GG, 1 W of input power is not converted into 1 W of acoustic power. Almost most GGs have an efficiency of less than 10%. Losses caused by low efficiency are transformed into heating of the voice coils, causing their mechanical deformation and loss of shape. Overheating of the voice coil frame causes weakening of its structure, and even complete destruction. In addition, overheating can cause the glue to foam and enter the air gap, causing the voice coil to no longer move freely. Eventually, the voice coil winding may simply blow out like a fuse link. It is absolutely obvious that this cannot be allowed.

Determining the permissible power of multi-band speakers has always been a serious problem for users and developers. Users who replace damaged tweeters most often

They are convinced that what happened is not their fault. It would seem that the output power of the amplifier is 50 W, and the power of the speaker is 200 W, and, nevertheless, the high-frequency speaker fails after some time. This problem forced engineers to figure out why this was happening. Many theories have been put forward. Some of them have been scientifically confirmed, others remain as theories.

Let's consider several views on the situation.

HARMONICS THEORY

Studies of the energy distribution across the signal spectrum have shown that, regardless of the type of music, the level of high-frequency energy in the sound signal is much lower than the level of low-frequency energy. This fact makes it even more difficult to figure out why tweeters are damaged. It would seem that if the amplitude of high frequencies is lower, then the low-frequency speakers should be damaged first, and not the high-frequency speakers.

Speaker manufacturers also use this information when developing their products. Understanding the energy spectrum of music allows them to significantly improve the sound of tweeters by using lighter moving systems, as well as using thinner wire in the voice coils. In speakers, the power of high-frequency speakers usually does not exceed 1/10 of the total power of the speaker itself.

But because in the low-frequency (LF) range there is more musical energy than in the high-frequency (HF) range, which means, due to its low power, high-frequency energy cannot cause damage to the high-frequency speakers. Therefore, the source of high frequencies powerful enough to damage tweeters is somewhere else. So, where is he located?

It has been suggested that if there are sufficient low-frequency components in the audio signal to overload the amplifier, it is likely that, as a result of limiting the output signal, high-frequency distortion will be sufficiently powerful to damage the tweeter.

Table 1. Harmonic amplitudes 100 Hz square wave, 0 dB = 100 W

This theory became quite widespread in the early 70s and gradually began to be perceived as “dogma”. However, as a result of studies of the reliability and security of power amplifiers under typical conditions, as well as the practice of operating amplifiers and speakers by typical users, it turned out that overload is a common occurrence and it is not as noticeable to the ear as most people think. The response of amplifier overload indicators is usually delayed and does not always accurately indicate the real overload. In addition, many amplifier manufacturers deliberately slow down their response based on their own ideas about how much distortion must occur for the indicator to light up.

More advanced and better sounding amplifiers, incl. Amplifiers with soft clipping will also damage tweeters. However, more powerful amplifiers cause less damage to tweeters. These facts further strengthen the theory that the source of damage to high-frequency speakers is still amplifier overload (clipping). It would seem that there is only one conclusion - clipping is the main reason for damage to high-frequency speakers.

But let's continue to explore this phenomenon.

AMPLITUDE COMPRESSION

When the amplitude of a sinusoidal signal is limited, the amplifier introduces large distortions into the original signal, and the shape of the resulting signal resembles the shape of a rectangle. In this case, an ideal rectangle (meander) has the highest level of higher harmonics. (see Figure 1). A less clipped sine wave has harmonics of the same frequencies, but at a lower level.

Take a look at the spectral composition of a square wave signal with a frequency of 100 Hz and a power of 100 W presented in Table 1.

As you can see, the power reaching the tweeter after passing this signal through an ideal crossover with a cutoff frequency of 1 kHz is less than 2 W (0.83 + 0.589 = 1.419 W). It's not much. And do not forget that in this case a hard, ideal overload of a 100-watt amplifier is simulated, capable of turning a sine into a meander. A further increase in overload will no longer increase harmonics.

Rice. 1. Harmonic components of a 100 Hz square wave relative to a 100 Hz sine wave

The results of this analysis indicate that even if a weak high-frequency speaker with a power of 5-10 W is used in a 100W speaker, harmonic damage to it is impossible, even if the signal takes the form of a meander. However, the speakers are still damaged.

This means that we need to find something else that could cause such failures. So what's the matter?

The reason is the amplitude compression of the signal.

Compared to older amplifier models, modern high-quality amplifiers have greater dynamic range and sound better when driven. Therefore, users are more tempted to overdrive amplifiers and clip them on low-frequency dynamic peaks, because in this case, large audible distortions do not occur. This results in compression of the dynamic characteristics of the music. The volume of high frequencies increases, but the volume of low frequencies does not. This is perceived by ear as an improvement in sound brightness. Some may interpret this as an increase in volume without an accompanying change in sound balance.

For example, we will increase the signal level at the input of a 100-watt amplifier. Low frequency components will be limited to 100 W due to overload. As the input level is further increased, the high-frequency components will rise until they also reach the 100 W clipping point.

Look at fig. 2, 3 and 4. The graphs are graduated in volts. At an 8-ohm load, 100 W corresponds to a voltage of 40 V. Before the limitation, the low-frequency components have a power of 100 W (40 V), and the high-frequency components only have a power of 5-10 W (9-13 V).

Let's assume that a music signal with low-frequency and high-frequency components is fed to a 100-watt amplifier (8 ohms). We use a mixture of a low-level HF sinusoidal signal with a high-level LF signal (see Fig. 2). The level of the high-frequency components supplied to the tweeter is at least 10 dB lower than the level of the low-frequency components. Now let’s increase the volume until the signal is limited (+3 dB overload, see Fig. 3).

Rice. 2. A low-level, high-frequency sine wave mixed with a burst of high-level, low-frequency sine wave

Rice. 3. Output of a 100-watt amplifier with 3 dB of overload

Rice. 4. Output of a 100-watt amplifier with 10 dB of overload

Please note that, judging by the waveform, only the low-frequency components were limited, and the level of the high-frequency components simply increased. Of course, clipping generates harmonics, but their level is significantly lower than that of the meander we discussed earlier. The amplitude of the HF components increased by 3 dB in relation to the LF (this is equivalent to the amplitude compression of the signal by 3 dB).

When the amplifier is overloaded by 10 dB, the amplitude of the HF components will increase by 10 dB. Thus, every 1 dB increase in volume causes an increase in the amplitude of the HF components by 1 dB. The growth will continue until the power of the RF components reaches 100W. Meanwhile, the peak level of low-frequency components cannot exceed 100 W (see Fig. 4). This graph corresponds to almost 100% compression, because... there is almost no difference between the HF and LF components.

Now it is easy to see how much the power of the RF signal exceeds the power of a 5-10-watt tweeter. It is true that overloading will generate additional harmonics, but they will never reach the level of the amplified original high-frequency signals.

You might think that the signal distortion would be unbearable. Don't fool yourself. You will be amazed to learn how high the overload limit is, above which it will no longer be possible to listen to anything. Just turn off the overload indicator on the amplifier and see to what level you turn the amplifier's volume control. If you measure the level of the amplifier's output signal with an oscilloscope, the level of overload will surprise you. An overload level of 10 dB on low-frequency components is common.

WHAT TO DO?

If we can protect amplifiers from overloading (clipping), we can use speakers more efficiently. To prevent overload and resulting amplitude compression, any modern amplifier must use the so-called. slip limiters. They prevent the aforementioned amplitude compression because When the threshold value is reached at any frequency, the level of all frequencies decreases by the same amount.

In external limiters, the response threshold (threshold) is set by the user. Fine tune

This threshold for limiting amplifiers is quite difficult. In addition, the clipping level of amplifiers is not a constant value. It changes depending on the supply voltage, AC resistance, and even the nature of the signal. The limiter threshold must continuously monitor these factors. The most correct solution would be to tie the threshold to the amplifier overload signal.

It is quite logical to build a limiter inside the amplifier. In modern amplifiers, it is easy to determine the moment when an overload occurs with great accuracy. It is to this that the so-called built-in amplifiers react. slip limiters. As soon as the amplifier's output signal reaches the overload level, the control circuit turns on the limiter control element.

The second parameter, after the response threshold, inherent in any limiter is the response and release times. More important is the recovery time after overload (release time).

There are two options for using amplifiers:

• work as part of a multi-band amplifier complex,

• work on broadband speakers.

In the first case, either only the low-frequency band, or the mid-frequency and high-frequency bands can be supplied to the amplifier. When setting a long release time and operating the amplifier in the mid-high frequency bands, the “tails” of the limiter recovery can be noticeable by ear. And, conversely, with a short release time and operation in the low frequency band, signal shape distortions may occur.

When operating an amplifier on a wideband speaker, you have to look for some compromise value for the recovery time.

In this regard, amplifier manufacturers take two paths - either a compromise release time is selected, or a release time switch (SLOW-FAST) is introduced.

CONCLUSIONS:

I thought it would be useful and interesting to many. Information taken from the Internet.

The high-frequency speaker is also a tweeter, it is also a tweeter, the smallest in your car. Typically installed in door pillars. Size about 5cm in diameter.

Midrange speaker is a mid-range speaker.

LF-low-frequency speaker (bidbass)

One of the mandatory stages of setting up the sound in a car is selecting the optimal frequency separation between all emitting heads: LF, LF/MF, MF (if any) and HF. There are two ways to solve this problem.

Firstly, restructuring, and often complete reworking of the standard passive crossover, and secondly, connecting the speakers to an amplifier operating in multi-band amplification mode, the so-called options for switching on Bi-amp (two-way amplification) or Tri-amp (three-way amplification).

The first method requires serious knowledge of electroacoustics and electrical engineering, so for independent use it is only available to specialists and experienced amateur radio electronics engineers, but the second, although it requires a larger number of amplification channels, is also available to a less trained car enthusiast.

Moreover, the vast majority of power amplifiers sold are initially equipped with a built-in active crossover. In many models it is so developed that it successfully and with sufficiently high quality allows for multi-band connection of speakers with a large number of speakers. However, the lack of a developed crossover in an amplifier or head unit does not stop fans of this method of sounding the interior, since there are many external crossovers on the market that can solve these problems.

First of all, it should be said that we will not give you 100% universal recommendations, since they do not exist. In general, acoustics is a field of technology where experimentation and creativity play a large role, and in this sense, fans of audio technology are lucky. But to carry out an experiment, so that it doesn’t turn out like that crazy professor’s - with explosions and smoke - you must follow certain rules. The first rule is do no harm, and others will be discussed below.

The most difficult part is the inclusion of mid-range and/or high-frequency components. And the point here is not only that these ranges carry the maximum information load, being responsible for the formation of the stereo effect, sound stage, and are also highly susceptible to intermodulation and harmonic distortion if the crossover frequency is incorrectly set, but also that this frequency directly affects and reliability of midrange and tweeter speakers.

Turning on the HF head.

The choice of the lower limit frequency of the range of signals supplied to the HF head depends on the number of bands of the speaker system. When a two-way speaker is used, then in the most typical case, i.e. When the bass/midrange head is located in the doors, in order to raise the level of the sound stage, it is advisable to select the cutoff frequency as low as possible. Modern high-quality tweeters with a low resonant frequency FS (800-1500 Hz) can reproduce signals as low as 2000 Hz. However, most HF drivers in use have a resonant frequency of 2000-3000 Hz, so it should be remembered that the closer to the resonant frequency we set the crossover frequency, the greater the load placed on the HF driver.

Ideally, with a filter attenuation slope of 12 dB/octave, the separation between the crossover frequency and the resonant frequency should be greater than an octave. For example, if the resonant frequency of the head is 2000 Hz, then with a filter of this order the crossover frequency should be set to 4000 Hz. If you really want to choose a crossover frequency of 3000 Hz, then the slope of the filter attenuation characteristic should be higher - 18 dB/oct, or better yet - 24 dB/oct.

There is another issue to consider when setting the crossover frequency for a tweeter. The fact is that after matching the components for the reproduced frequency range, you also need to match them in level and phase. The latter, as always, is a stumbling block - it seems like everything was done correctly, but the sound is “not right.” It is known that a first-order filter will produce a phase shift of 90°, a second-order filter will produce a phase shift of 180° (antiphase), etc., so during setup, do not be lazy to listen to speakers with different switching polarities.

The human ear is very sensitive to the frequency range of 1500-3000 Hz, and in order to transmit it as well and clearly as possible, one should be extremely careful. It is possible to break (split) the sound range in this area, but you should think about how to properly eliminate the consequences of the unpleasant sound later. From this point of view, a three-way speaker system is more convenient and safer to set up, and the midrange speaker used in it allows not only to effectively reproduce the range from 200 to 7000 Hz, but also to more easily solve the problem of building a sound stage. In three-way speakers, the HF speaker is turned on at higher frequencies - 3500-6000 Hz, that is, obviously above the critical frequency band, and this makes it possible to reduce (but not eliminate) the requirements for phase matching.

Turning on the midrange head.

Before discussing the choice of frequency for dividing the mid- and low-frequency ranges, let us turn to the design features of the mid-range speakers. Recently, midrange speakers with dome diaphragms have become very popular among installers. Compared to cone midrange speakers, they provide a wider polar pattern and are easier to install since they do not require additional acoustic design. Their main disadvantage is their high resonant frequency, lying in the range of 450-800 Hz.

The problem is that the higher the lower limiting frequency of the signal band supplied to the midrange speaker, the smaller the distance between the midrange and low-frequency heads should be and the more critical it is where exactly the woofer is located and where it is oriented. Practice shows that dome midrange speakers can be switched on with a crossover frequency of 500-600 Hz without any problems with matching. As you can see, for most sold copies this is a fairly critical range, so if you decide on such a separation, the order of the separation filter should be quite high - for example, 4th.

It should be added that recently dome speakers with a resonant frequency of 300-350 Hz have begun to appear. They can be used starting at a frequency of 400 Hz, but for now the cost of such specimens is quite high.

The resonant frequency of midrange speakers with a cone diffuser lies in the range of 100-300 Hz, which allows them to be used starting at a frequency of 200 Hz (in practice, 300-400 Hz is more often used) and with a low-order filter, while the woofer/midrange speaker is completely freed from the need to work in the midrange. Reproduction of signals with frequencies from 300-400 Hz to 5000-6000 Hz without separation between speakers makes it possible to achieve pleasant, high-quality sound.

Turning on the woofer/midrange speaker.

Gradually we reached the low frequency range. Modern midrange/bass speakers allow you to operate effectively in the frequency range from 40 to 5000 Hz. The upper limit of its operating frequency range is determined by where the tweeter (in a 2-way speaker) or midrange speaker (in a 3-way speaker) starts to operate.

Many people are concerned about the question: is it worth limiting its frequency range from below? Well, let's figure it out. The resonant frequency of modern LF/MF speakers of a standard size of 16 cm lies in the range of 50-80 Hz and due to the high mobility of the voice coil, these speakers are not so critical to operating at frequencies below the resonant one. Nevertheless, reproducing frequencies below the resonant one requires some effort from it, which leads to a decrease in output in the range of 90-200 Hz, and in two-way systems, also in the quality of midrange transmission. Since the main energy of the bass drum strikes occurs in the frequency range from 100 to 150 Hz, the first thing you lose is a clearly defined punch. By limiting the range of signals reproduced by the low-frequency head to 60-80 Hz using a high-pass filter, you will not only allow it to work much cleaner, but also get a louder sound, in other words, better output.

Subwoofer.

It is better to assign signals with frequencies below 60-80 Hz to a separate speaker - a subwoofer. But remember that the sound range below 60 Hz is not localized in a car, which means the location of the subwoofer is not so important. If you have fulfilled this condition, and the sound of the subwoofer is still localized, then first of all you need to increase the order of the low-pass filter. You should also not neglect the infra-low frequency suppression filter (Subsonic, or FINCH). Do not forget that the subwoofer also has its own resonant frequency and by cutting off the frequencies below it, you achieve comfortable sound and reliable operation of the subwoofer. As practice shows, the pursuit of deep bass significantly increases the cost of a subwoofer. Believe me, if the sound system you have assembled reproduces the sound range from 50 to 16,000 Hz with good quality, this is quite enough to comfortably listen to music in the car.

Head pairing methods.

Quite often the question arises: should you have the same order of low-pass and high-pass filters? It’s not at all necessary, and it’s not even necessary at all. For example, if you have installed a two-way front speaker with a large speaker separation, then in order to compensate for the dips in the frequency response at the crossover frequency, the low-frequency/mid-range head is often included with a lower-order filter. Moreover, it is not even necessary that the cutoff frequencies of the high-pass filter and the low-pass filter coincide.

Let's say, to compensate for excess brightness at the point of separation, the bass/midrange head can operate up to 2000 Hz, and the tweeter - starting from 3000 Hz. It is important to remember that when using a first-order filter, the difference between the cutoff frequencies of the high-pass filter and the low-pass filter should be no more than an octave and decrease with increasing order. The same technique is used when pairing a subwoofer and midwoofer to attenuate standing waves (bass boom). For example, when setting the cutoff frequency of the low-pass filter of the subwoofer to 50-60 Hz, and the high-pass filter of the low-frequency/midrange head to 90-100 Hz, according to experts, unpleasant overtones caused by the natural rise in the frequency response in this frequency region due to the acoustic properties of the cabin are completely eliminated .

So even if the rule of transition from quantity to quality works in car audio, it is confirmed only in relation to the cost of individual components and man-years, which determine the experience and skill of the installer who will make the system reveal its sound potential.

The designs of high-frequency (HF) speakers are the most diverse. They can be ordinary, horn or dome. The main problem in their creation is the expansion of the direction of the emitted oscillations. In this regard, dome speakers have certain advantages. The diameter of the diffuser or radiating membrane of HF tweeters ranges from 10 to 50 mm. Often the tweeters are tightly closed at the back, which eliminates the possibility of modulation of their radiation by the radiation of low-frequency and mid-frequency emitters.

A typical miniature cone tweeter emits good high-frequency sounds, but has a very narrow radiation pattern - usually within an angle of 15 to 30 degrees (relative to the central axis). This angle is set when the speaker's output is typically reduced by -2 dB. The angle of deviation from both the horizontal and vertical axis is indicated. Abroad, this angle is called the angle of dispersion or dispersion of sound.

To increase the dispersion angle, diffusers or attachments for them are made in various shapes (spherical, horn-shaped, etc.). Much depends on the material of the diffuser. However, conventional tweeters are unable to emit sounds with frequencies noticeably higher than 20 kHz. Placing special reflectors in front of the tweeter (most often in the form of a plastic grid) allows you to significantly expand the radiation pattern. Such a grille is often an element of the acoustic frame of a tweeter or other emitter.

An eternal topic of debate is the question of whether it is necessary to emit frequencies above 20 kHz at all, since our ear cannot hear them, and even studio equipment often limits the effective range of sound signals at a level from 10 to 15-18 kHz. However, the fact that we do not hear such sinusoidal signals does not mean that they do not exist and do not affect the shape of the time dependences of real and rather complex audio signals with much lower repetition rates.

There is much convincing evidence that this shape is greatly distorted when the frequency range is artificially limited. One of the reasons for this is the phase shifts of various components of a complex signal. It is curious that our ear does not sense phase shifts themselves, but is able to distinguish signals with different forms of time dependence, even if they contain the same set of harmonics with the same amplitudes (but different phases). The nature of the frequency response decay and the linearity of the phase response even outside the effectively reproduced frequency range are of great importance.

Generally speaking, if we want to have uniform frequency response and phase response throughout the entire audio range, then the frequency range actually emitted by the acoustics should be noticeably wider than the audio one. All this fully justifies the development of broadband emitters by many leading companies in the field of electroacoustics.

Placement of HF emitters There is a problem - the result largely depends on where the heads are placed and how they are oriented. Let's talk about the HF head, or tweeter.

Features of HF heads From the theory of sound wave propagation it is known that as the frequency increases, the radiation pattern of the emitter narrows, and this leads to a narrowing of the optimal listening area. That is, it is possible to obtain a uniform tonal balance and the correct scene only in a small area of ​​​​space. Therefore, expanding the radiation pattern of the HF emitter is the main task of all loudspeaker designers. The weakest dependence of the radiation pattern on frequency is observed in dome tweeters. This type of HF emitters is the most common in automobile and household speakers. Other advantages of dome radiators are their small size and the absence of the need to create an acoustic volume, while the disadvantages include the low lower limit frequency, which lies in the range of 2.5-7 kHz. All these features are taken into account when installing a tweeter. The installation location is influenced by everything: the operating range of the tweeter, its directivity characteristics, the number of components installed (2- or 3-component systems), and even your personal taste. Let’s immediately make a reservation that there are no universal recommendations on this issue, so we cannot point the finger at you - they say, put it here and everything will be OK! However, today there are many standard solutions that are useful to familiarize yourself with. All of the following applies to non-processor circuits, but this is also true when using a processor; its presence simply provides much more opportunities to compensate for the negative impact of a non-optimal location.

Practical considerations. First, let us recall some canons. Ideally, the distance to the left and right tweeters should be the same, and the tweeters should be installed at the height of the listener's eyes (or ears). In particular, it is always best to move the tweeter heads as far forward as possible, since the further they are from the ears, there is less difference in the distances to the left and right drivers. The second aspect: the tweeter should not be far from the midrange or bass/midrange head, otherwise you will not get good tonal balance and phase matching (usually guided by the length or width of the palm). However, if the tweeter is set low, the sound stage falls down, and you seem to be above the sound. If the setting is too high, due to the large distance between the tweeters and midrange speakers, the integrity of the tonal balance and phase matching is lost. For example, when listening to a track with a recording of a piano piece, on low notes the same instrument will sound low, and on high notes it will soar sharply upward.

Directivity of the HF head. When you have figured out where to install the HF head, you should decide on its direction. As practice shows, to obtain the correct timbral balance, it is better to direct the tweeter towards the listener, and to obtain good depth of the sound stage, use reflection. The choice is determined by your personal feelings about the music you listen to. The main thing here is to remember that there can only be one optimal listening location.
It is advisable to orient the tweeter in space so that its central axis is directed towards the listener’s chin, that is, set a different angle of rotation for the left and right tweeters. There are two things to keep in mind when orienting a reflective tweeter. Firstly, the angle of incidence of the sound wave is equal to the angle of reflection, and secondly, by lengthening the path of the sound, we take the sound stage further, and if you get carried away, you can get the so-called tunnel effect, when the sound stage is far from the listener, as if at the end of a narrow corridor.

Setting method. Having outlined, in accordance with the recommendations given, the location of the RF heads, it is worth starting experiments. The fact is that no one will ever say in advance where exactly a 100% “hit” with your components will be ensured. The most optimal location will allow you to determine the experiment, which is quite simple to set up. Take any sticky material, for example, plasticine, double-sided tape, Velcro or model hot glue, put on your favorite music or test disc and, taking into account all of the above, start experimenting. Try different locations and orientation options in each. Before finally installing the high-frequency driver, it is better to listen a little more and correct it on plasticine.to nowhere.

Creative approach. Setting up and choosing the location of the tweeter has its own nuances for 2- and 3-component systems. In particular, in the first case, it is difficult to ensure close proximity of the tweeter and the low-frequency/mid-range emitter. But in any case, you shouldn’t be afraid to experiment - we’ve come across installations where HF heads ended up in the most unexpected places. Is there any point in having an additional pair of tweeters? For example, the American company Boston Acoustics produces sets of component speakers, where the crossover already has space for connecting a second pair of HF heads. As the developers themselves explain, the second pair is necessary to raise the level of the sound stage. In test conditions, we listened to them as an addition to the main pair of tweeters and were surprised how significantly the space of the sound stage expanded and the elaboration of nuances improved