Home Steering Increasing the power of the amplifier on the TDA7294 chip. Low frequency amplifier (LF) on the TDA7250 chip Amplifier circuits on one chip

Increasing the power of the amplifier on the TDA7294 chip. Low frequency amplifier (LF) on the TDA7250 chip Amplifier circuits on one chip

I'd say it's just a super simple amp that contains all four elements and puts out 40 watts of power into two channels!
4 parts and 40 W x 2 power output Karl! This is a godsend for car enthusiasts, since the amplifier is powered by 12 Volts, the full range is from 8 to 18 Volts. It can be easily integrated into subwoofers or speaker systems.
Everything is accessible today thanks to the use of modern element base. Namely the chip - TDA8560Q.

This is a PHILIPS chip. Previously, the TDA1557Q was in use, on which you can also build a stereo amplifier with an output power of 22 W. But it was later modernized by updating the output stage and the TDA8560Q appeared with an output power of 40 W per channel. Also similar is the TDA8563Q.

Car amplifier circuit on a chip

The diagram shows a microcircuit, two input capacitors and one filter capacitor. The filter capacitor is specified with a minimum capacity of 2200 uF, but the best solution would be to take 4 of these capacitors and parallel them, this will ensure more stable operation of the amplifier at low frequencies. The microcircuit must be installed on a radiator, the larger the better.

Building a simple amplifier



It is also possible to increase the number of components in the circuit that increase reliability during operation, but not fundamentally.


Five more details have been added here, I’ll explain why. Two 10K Ohm resistors will remove the hum if there are long wires going to the circuit. A 27 K Ohm resistor and a 47 uF capacitor provide a smooth start of the amplifier without clicks. A 220 pF capacitor will filter out high-frequency noise traveling along the power wires. So I recommend modifying the circuit with these nodes; it won’t be superfluous.
I would also like to add that the amplifier develops full power only at a 2 Ohm load. At 4 Ohms there will be somewhere around 25 W, which is also very good. So our Soviet acoustics will be rocked.
Low-voltage, single-polar power supply provides additional advantages: it can be used in car speakers, but at home it can be powered from an old computer power supply.
The minimum number of components allows you to build in an amplifier to replace an old one that has failed on a microcircuit of other brands.

I first noticed the LT1210 chip on Datagor. It was given to me after the New Year's competition as a gift. The first practical design on the LT1210 was a single-supply ear amplifier for our Goldsmith DAC. .
Then this development moved into the hands of Yooree, where, in addition to the name ASTREAM, it received bipolar power supply and some uh... circuit design changes.

The studied properties of the chip and the reviews of those who repeated Yuri’s WHH prompted me to install a linear amplifier for mini-monitors. The main goal was to develop a stable, easily repeatable design that does not require adjustment. Along the way, issues such as problems with resistance to reactive load, increased output constant at WHH, etc. were resolved.
When developing the amplifier, such an important factor as the optimal layout of “grounds” and PP was taken into account. A version with double the output power has been developed.

From Datagor:
The bug fix was a great success! The result was the correct WHH, or rather WK - Waft Killer.
There are even two options - UMZCH Waft Killer 7 and the more powerful Waft Killer 15.
Datagor.RU recommends these particular versions of UMZCH on LT1210 for assembly.

Update- see the bridge version there WK60!!!


What do you think is shown in the photo? So, we don’t give hints from the back rows!

In the meantime, we are looking in a search engine for the inscription on the board, I will tell you what it is. This is the UcD250 module from Hypex Electronics.
Nothing special. Class D, 250 W declared power. Normal, right?
Have the Chinese painted their Watts again? No, today everything is honest and real.
This is the insides of the EveAudio near-field monitor, designed for professional studio work.
The size of the module can be estimated from the photo; for scale, use a regular AA battery.

Digitally controlled preamplifier-switcher. We use programming through the Arduino shell, electronic potentiometers from Microchip, and TFT graphics.


It was not my plan to develop and assemble this device. Well, there’s just no way! I already have two preamps. Both suit me quite well.
But, as usually happens for me, a coincidence of circumstances or a chain of certain events, and now a task has emerged for the near future.

Greetings to Datagor readers again! In the second part we will deal with the construction of a 6-channel volume control.

The regulator consists of two main chips: an ATiny26 microcontroller and a specialized TDA7448 chip. I added a volume indicator (a line of 7 LEDs) to roughly know what level is set, because an infinitely rotating encoder acts as a control knob.


And then I decided to try 5.1 surround sound. But on a budget, without sacrifices. And off we go! I started disassembling, tinkering, designing, assembling, sawing, drilling... In general, I started pumping the system.
I present the results in two parts to dear readers.

By chance, the Arcturus-006-stereo record player fell into my hands. Therefore, there was an urgent need for a phono stage. On the Internet I came across scheme by A. Bokarev, for which I decided to make a much-needed device.
At the back of the player there are two output connectors (SG-5/DIN): one from the built-in phono stage (500mV), the second bypassed for connecting to an external one (5mV). When using the built-in phono stage, a jumper is installed in the second output.

I didn’t like the characteristics of the built-in corrector, and when I turned it on, it turned out that it was faulty - I only heard a 50 Hz hum in the speakers. There was no desire to restore it, so I disconnected the built-in corrector board completely.
I'll listen to my version.


Photo source: vega-brz.ru


The Arctur-006-stereo electric player of the highest complexity group has been produced by the Berdsk Radio Plant since 1983. The player is made on the basis of a two-speed EPU G-2021, with an ultra-low-speed electric motor and direct drive. There is a pressure regulator and a roll force compensator, adjustment of the disc rotation speed using a strobe light, auto-stop, micro-lift, speed switch and auto-return of the tonearm at the end of the record.

This project considers amplifiers for headphones on mass-produced microcircuits, such as BA5415A and BA5417.


I refrained from philosophical discussions about which of the presented sound reproduction schemes is “more correct.” The purpose of the experiments is different - to provide worthy schemes for repetition, and enthusiastic readers will make their own choice and share their impressions.

- despite its relative simplicity, it provides fairly high parameters. Actually, to tell the truth, “chip” amplifiers have a number of limitations, so “loose” amplifiers can provide higher performance. In defense of the microcircuit (otherwise why do I use it myself and recommend it to others?) we can say:

Simple and effective scheme

  • the circuit is very simple
  • and very cheap
  • and requires virtually no adjustment
  • and you can assemble it in one evening
  • and the quality is superior to many amplifiers of the 70s ... 80s, and is quite sufficient for most applications (and even modern systems under $300 can be inferior to it)
  • thus, the amplifier is suitable for both beginners and experienced radio amateurs (for example, I once needed a multi-channel amplifier to test an idea. Guess what I did?).

In any case, a poorly made and incorrectly tuned amplifier in bulk will sound worse than a microcircuit amplifier. And our task is to make a very good amplifier. It should be noted that the sound of the amplifier is very good (if it is made correctly and powered correctly); there is information that some company produced Hi-End amplifiers based on the TDA7294 chip! And our amplifier is no worse!!!

- this is practically a repetition of the connection diagram proposed by the manufacturer. And this is no coincidence - who knows better how to turn it on. And there certainly won’t be any surprises due to non-standard activation or operating mode.

Input path

The input circuit R1C1 is a low pass filter (LPF), cutting off everything above 90 kHz. It is impossible without it - the 21st century is, first of all, the century of high-frequency interference. The cutoff frequency of this filter is quite high. But this is on purpose - I don’t know what this amplifier will be connected to. If there is a volume control at the input, then it will be just right - its resistance will be added to R1, and the cutoff frequency will decrease (the optimal value of the volume control resistance is ~10 kOhm, more is better, but the regulation law will be violated).

Next, the R2C2 chain performs the exact opposite function - it does not allow frequencies below 7 Hz to enter the input. If this is too low for you, the C2 capacity can be reduced. If you get too carried away with reducing capacity, you may be left without any low frequencies at all. For the full audio range, C2 must be at least 0.33 µF. And remember that capacitors have a fairly wide range of capacitances, so if it says 0.47 microfarads, it could easily turn out to be 0.3! And further. At the lower end of the range, the output power is reduced by 2 times, so it is better to choose it lower:

C2[uF] = 1000 / (6.28 * Fmin[Hz] * R2[kOhm])

Resistor R2 sets the input resistance of the amplifier. Its value is slightly larger than according to the datasheet, but this is also better - too low an input impedance may “not be liked” by the signal source. Please note that if a volume control is turned on in front of the amplifier, then its resistance should be 4 times less than R2, otherwise the law of volume control will change (the volume value depends on the angle of rotation of the control). The optimal value of R2 lies in the range of 33...68 kOhm (higher resistance will reduce noise immunity).

Audio amplifier circuit on a chip, namely, the amplifier switching circuit is not inverting. Resistors R3 and R4 create a negative feedback circuit (NFC). The gain is:

Ku = R4 / R3 + 1 = 28.5 times = 29 dB

Gain

This is almost equal to the optimal value of 30 dB. You can change the gain by changing resistor R3. Please note that you cannot make Ku less than 20 - the microcircuit can excite itself. It’s also not worth doing it more than 60 - the depth of feedback will decrease and the distortion will increase. With the resistance values ​​indicated in the diagram, with an input voltage of 0.5 volts, the output power at a 4 ohm load is 50 W. If the sensitivity of the amplifier is not enough, then it is better to use a preamplifier.

The resistance values ​​are slightly higher than those recommended by the manufacturer. Firstly, this increases the input impedance, which is nice for the signal source (to obtain maximum DC balance, R4 must be equal to R2). Secondly, it improves the operating conditions of the electrolytic capacitor C3. And thirdly, it enhances the beneficial effects of C4. More about this. Audio amplifier circuit on a chip works in the following sequence: capacitor C3 in series with R3 creates 100% OOS for direct current (since its resistance to direct current is infinite, and Ku is equal to unity). In order for the influence of C3 on the amplification of low frequencies to be minimal, its capacitance must be quite large. The frequency at which the influence of C3 becomes noticeable is:

f [Hz] = 1000 / (6.28 * R3 [kOhm] * C3 [uF]) = 1.3 Hz

Reducing distortion

This frequency should be very low. The fact is that C3 is electrolytic polar, and it is supplied with alternating voltage and current, which is very bad for it. Therefore, the lower the value of this voltage, the less distortion introduced by C3. For the same purpose, its maximum permissible voltage is chosen to be quite large (50V), although the voltage on it does not exceed 100 millivolts. It is very important that the cutoff frequency of the R3C3 circuit is much lower than the input circuit R2C2. After all, when the influence of C3 manifests itself due to an increase in its resistance, then the voltage on it increases (the output voltage of the amplifier is redistributed between R4, R3 and C3 in proportion to their resistances). If at these frequencies the output voltage drops (due to a drop in the input voltage), then the voltage at C3 does not increase. In principle, you can use a non-polar capacitor as C3, but I can’t say for sure whether this will improve the sound or worse: a non-polar capacitor is “two in one” polar capacitors connected back-to-back.

Capacitor C4 bypasses C3 at high frequencies: electrolytes have another drawback (in fact, there are many drawbacks, this is the price to pay for the high specific capacitance) - they do not work well at frequencies above 5-7 kHz (expensive ones are better, for example Black Gate, which costs 7-7 kHz). 12 euros per piece works well at 20 kHz). Film capacitor C4 “takes over the high frequencies,” thereby reducing the distortion introduced to them by capacitor C3. The larger the C4 capacity, the better. And its maximum operating voltage can be relatively small.

Amplifier stability

Circuit C7R9 increases the stability of the amplifier. In principle, the amplifier is very stable, and you can do without it, but I came across instances of microcircuits that worked worse without this circuit. Capacitor C7 must be designed for a voltage no lower than the supply voltage.

Audio amplifier circuit on a chip, and in particular capacitors C8 and C9 carry out the so-called volt boost. Through them, part of the output voltage flows back into the pre-final stage and is added to the supply voltage. As a result, the supply voltage inside the chip is higher than the voltage of the power supply. This is necessary because the output transistors provide an output voltage 5 volts less than the voltage at their inputs. Thus, in order to get 25 volts at the output, you need to apply a voltage of 30 volts to the gates of the transistors, but where can you get it? So we take it from the exit. Without a voltage boost circuit, the output voltage of the microcircuit would be 10 volts less than the supply voltage, but with this circuit it would be only 2-4. Film capacitor C9 takes over the work at high frequencies, where C8 performs worse. Both capacitors must withstand a voltage not lower than 1.5 times the supply voltage.

Controlling Mute and StdBy modes

Resistors R5-R8, capacitors C5, C6 and diode D1 control the Mute and StdBy modes when the power is turned on and off (see Mute and StandBy modes in the TDA7294/TDA7293 chip). They provide the correct sequence for turning on/off these modes. True, everything works fine even if they are in the “wrong” sequence, so you need such control more for your own pleasure.

Capacitors C10-C13 filter the power. Their use is mandatory - even with the best power supply, the resistance and inductance of the connecting wires can affect the operation of the amplifier. With these capacitors, no wires are a problem (within reasonable limits)! There is no need to reduce the capacity. Minimum 470 µF for electrolytes and 1 µF for film ones. When installing on a board, it is necessary that the leads are as short as possible and well soldered - do not skimp on solder. All these capacitors must withstand a voltage not lower than 1.5 times the supply voltage.

Separation of input and output ground

And finally, resistor R10. It serves to separate the input and output land. “On the fingers” its purpose can be explained as follows. A large current flows from the amplifier output through the load to ground. It may happen that this current, flowing through the “ground” conductor, will also flow through the section through which the input current flows (from the signal source, through the amplifier input, and then back to the source along the “ground”). If the resistance of the conductors was zero, then there would be no problem. But the resistance, although small, is not zero, so a voltage will appear at the resistance of the “ground” wire (Ohm’s law: U=I*R), which will add up to the input. Thus, the output signal of the amplifier will go to the input, and this feedback will not bring anything good, only all sorts of nasty things. The resistance of resistor R10, although small (the optimal value is 1...5 Ohms), is much greater than the resistance of the ground conductor, and through it (the resistor) hundreds of times less current will flow into the input circuit than without it.

In principle, if the board layout is good (and I have a good one), this will not happen, but on the other hand, something similar can happen on a “macro scale” along the signal source-amplifier-load circuit. A resistor will help in this case too. However, it can be completely replaced with a jumper - it was used based on the principle “it’s better to be safe than sorry.”

Power supply

Audio amplifier circuit on a chip is powered by bipolar voltage (i.e. these are two identical sources connected in series, and their common point is connected to ground).

The minimum supply voltage according to the datasheet is +- 10 volts. I personally tried to power it from +-14 volts - the microcircuit works, but is it worth doing this? After all, the output power is scanty! The maximum supply voltage depends on the load resistance (this is the voltage of each arm of the source):

This dependence is caused by the permissible heating of the microcircuit. If the microcircuit is installed on a small radiator, it is better to reduce the supply voltage. The maximum output power received from the amplifier is approximately described by the formula:

where the units are: V, Ohm, W (I will study this issue separately and describe it), and Uip is the voltage of one arm of the power source in silent mode.

Power supply power

The power supply power should be 20 watts more than the output power. The rectifier diodes are designed for a current of at least 10 Amps. The capacitance of the filter capacitors is at least 10,000 µF per arm (less is possible, but the maximum power will decrease and distortion will increase).

It must be remembered that the rectifier voltage at idle is 1.4 times higher than the voltage on the secondary winding of the transformer, so do not burn the microcircuit! A simple but fairly accurate program for calculating a power supply:

PCB layout

Audio amplifier circuit on a chip, the board of which is wired taking into account all the requirements for wiring high-quality amplifiers. The entrance is separated as far as possible from the exit, and is enclosed in a “screen” of divided earth - entrance and exit. The power supply paths ensure maximum efficiency of filter capacitors (at the same time, the length of the leads of capacitors C10 and C12 should be minimal). In my experimental board, I installed terminal blocks for connecting the input, output and power supply - there is a place for them (capacitor C10 may get in the way a little), but for stationary structures it is better to solder all these wires - it’s more reliable.

In addition to low resistance, wide tracks also have the advantage that they are more difficult to peel off when overheated. And when manufacturing using the “laser-ironing” method, if a square of 1 mm x 1 mm is not “printed” anywhere, then it’s not a big deal - the conductor won’t break anyway. In addition, a wide conductor holds heavy parts better (while a thin conductor can simply peel off from the board).

There is only one jumper on the board. It lies under the pins of the microcircuit, so it needs to be mounted first, and leave enough space under the pins so that it doesn’t short out.

All resistors except R9 with a power of 0.12 W, Capacitors C9, C10, C12 K73-17 63V, C4 I used K10-47V 6.8 uF 25V (I had it lying around in the closet... With such a capacitance, even without capacitor C3, the cutoff frequency in the OOS circuit it turns out 20 Hz - where deep bass is not needed, one such capacitor is quite enough). However, I recommend using all capacitors of the K73-17 type. I consider the use of expensive “audiophile” ones to be economically unjustified, and cheap “ceramic” ones will give worse sound (this is in theory, in principle - please just remember that some of them can withstand a voltage of no more than 16 volts and cannot be used as a C7). Any modern electrolytes will do. Audio amplifier circuit on a chip has on the printed circuit board polarity symbols for connecting all electrolytic capacitors and a diode. Diode - any low-power rectifier that can withstand a reverse voltage of at least 50 volts, for example 1N4001-1N4007. It is better not to use high-frequency diodes.

In the corners of the board there is space for holes for M3 mounting screws - you can fasten the board only to the chip body, but it is still more reliable to secure it with screws.

Heat sink for microcircuit

The microcircuit must be installed on a radiator with an area of ​​at least 350 cm2. More is better. In principle, it has thermal protection built into it, but it’s better not to tempt fate. Even if active cooling is assumed, the radiator must still be quite massive: with pulsed heat release, which is typical for music, heat is more effectively removed by the heat capacity of the radiator (i.e., a large cold piece of iron) rather than by dissipation into the environment.

The metal housing of the microcircuit is connected to the negative side of the power supply. This gives rise to two ways to install it on a radiator:

Through an insulating gasket, the radiator can be electrically connected to the housing.
Directly, in this case the radiator is necessarily electrically isolated from the body.

The second option (my favorite) provides better cooling, but requires caution, such as not removing the chip while the power is on.

In both cases, you need to use heat-conducting paste, and in the 1st option it should be applied both between the microcircuit body and the gasket, and between the gasket and the radiator.

Audio amplifier circuit on a microcircuit - setup

Communication on the Internet shows that 90% of all problems with equipment are due to its “not being adjusted.” That is, having soldered yet another circuit and having failed to fix it, the radio amateur gives up on it and publicly declares the circuit bad. Therefore, setup is the most important (and often the most difficult) stage in creating an electronic device.

A properly assembled amplifier does not need adjustment. But, since no one guarantees that all parts are absolutely in good working order, you need to be careful when you turn it on for the first time.

The first switch-on is carried out without load and with the input signal source turned off (it is better to short-circuit the input with a jumper). It would be nice to include fuses of about 1A in the power circuit (both in the plus and minus between the power source and the amplifier itself). Briefly (~0.5 sec.) Apply the supply voltage and make sure that the current consumed from the source is small - the fuses do not burn out. It is convenient if the source has LED indicators - when disconnected from the network, the LEDs continue to light for at least 20 seconds: the filter capacitors are discharged for a long time by the small quiescent current of the microcircuit.

Chip quiescent current

If the current consumed by the microcircuit is large (more than 300 mA), then there can be many reasons: short circuit in installation; poor contact in the “ground” wire from the source; “plus” and “minus” are confused; the pins of the microcircuit touch the jumper; microcircuit is faulty; capacitors C11, C13 are soldered incorrectly; capacitors C10-C13 are faulty.

Making sure that audio amplifier circuit on a chip maintains a normal quiescent current, feel free to turn on the power and measure the constant voltage at the output. Its value should not exceed +-0.05 V. High voltage indicates problems with C3 (less often with C4), or with the microcircuit. There have been cases when the “ground-to-ground” resistor was either poorly soldered or had a resistance of 3 kOhms instead of 3 ohms. At the same time, the output was constant 10...20 volts. By connecting an AC voltmeter to the output, we make sure that the AC voltage at the output is zero (this is best done with the input closed, or simply with the input cable not connected, otherwise there will be noise at the output). The presence of alternating voltage at the output indicates problems with the microcircuit, or circuits C7R9, C3R3R4, R10. Unfortunately, conventional testers often cannot measure the high-frequency voltage that appears during self-excitation (up to 100 kHz), so it is best to use an oscilloscope here.

If everything is in order here, we connect the load, check again for the absence of excitation with the load, and that’s it - you can listen!

Additional testing

But it’s better to do another test. The fact is that the most disgusting type of amplifier excitation, in my opinion, is “ringing” - when excitation appears only in the presence of a signal, and at a certain amplitude. Because it is difficult to detect without an oscilloscope and a sound generator (and it is not easy to eliminate), and the sound deteriorates enormously due to huge inter-modulation distortion. Moreover, this is usually perceived by ear as a “heavy” sound, i.e. without any additional overtones (since the frequency is very high), so the listener does not know that his amplifier is being excited. He just listens and decides that the microcircuit is “bad” and “doesn’t sound.”

If audio amplifier circuit on a microcircuit correctly assembled and a normal power source this should not happen.

However, sometimes it happens, and the C7R9 chain is precisely what struggles with such things. BUT! In a normal microcircuit, everything is OK even in the absence of C7R9. I came across copies of the microcircuit with ringing, in them the problem was solved by introducing the C7R9 circuit (that’s why I use it, even though it’s not in the datasheet). If such a nasty thing occurs even if you have a C7R9, then you can try to eliminate it by “playing” with the resistance (it can be reduced to 3 ohms), but I would not recommend using such a microcircuit - it’s some kind of defect, and who knows? what else will come out in it.

The problem is that the "ringing" can only be seen on an oscilloscope, which is when audio amplifier circuit on a chip receives a signal from a sound generator (you may not notice it on real music) - and not all radio amateurs have this equipment. (Although, if you want to do this business well, try to notice such devices, at least use them somewhere). But if you want high-quality sound, try to check it on the devices - “ringing” is the most insidious thing, and can damage the sound quality in a thousand ways. My boards:


"Desktop" test of the amplifier

Audio amplifier circuit on a chip after preliminary switching on the table, it showed that the circuit and printed circuit board are absolutely working! No additional settings were made after assembly according to the diagram! very satisfied, I recommend!

Preliminary switching on of the amplifier on the table showed that the circuit and printed circuit board are absolutely working! No additional settings were made after assembly according to the diagram! very satisfied, I recommend!

– The neighbor stopped knocking on the radiator. I turned the music up so I couldn't hear him.
(From audiophile folklore).

The epigraph is ironic, but the audiophile is not necessarily “sick in the head” with the face of Josh Ernest at a briefing on relations with the Russian Federation, who is “thrilled” because his neighbors are “happy.” Someone wants to listen to serious music at home as in the hall. For this purpose, the quality of the equipment is needed, which among lovers of decibel volume as such simply does not fit where sane people have a mind, but for the latter it goes beyond reason from the prices of suitable amplifiers (UMZCH, audio frequency power amplifier). And someone along the way has a desire to join useful and exciting areas of activity - sound reproduction technology and electronics in general. Which in the age of digital technology are inextricably linked and can become a highly profitable and prestigious profession. The optimal first step in this matter in all respects is to make an amplifier with your own hands: It is UMZCH that allows, with initial training on the basis of school physics on the same table, to go from the simplest designs for half an evening (which, nevertheless, “sing” well) to the most complex units, through which a good rock band will play with pleasure. The purpose of this publication is highlight the first stages of this path for beginners and, perhaps, convey something new to those with experience.

Protozoa

So, first, let's try to make an audio amplifier that just works. In order to thoroughly delve into sound engineering, you will have to gradually master quite a lot of theoretical material and not forget to enrich your knowledge base as you progress. But any “cleverness” is easier to assimilate when you see and feel how it works “in hardware.” In this article further, too, we will not do without theory - about what you need to know at first and what can be explained without formulas and graphs. In the meantime, it will be enough to know how to use a multitester.

Note: If you haven’t soldered electronics yet, keep in mind that its components cannot be overheated! Soldering iron - up to 40 W (preferably 25 W), maximum allowable soldering time without interruption - 10 s. The soldered pin for the heat sink is held 0.5-3 cm from the soldering point on the side of the device body with medical tweezers. Acid and other active fluxes cannot be used! Solder - POS-61.

On the left in Fig.- the simplest UMZCH, “which just works.” It can be assembled using both germanium and silicon transistors.

On this baby it is convenient to learn the basics of setting up an UMZCH with direct connections between cascades that give the clearest sound:

  • Before turning on the power for the first time, turn off the load (speaker);
  • Instead of R1, we solder a chain of a constant resistor of 33 kOhm and a variable resistor (potentiometer) of 270 kOhm, i.e. first note four times less, and the second approx. twice the denomination compared to the original according to the scheme;
  • We supply power and, by rotating the potentiometer, at the point marked with a cross, we set the indicated collector current VT1;
  • We remove the power, unsolder the temporary resistors and measure their total resistance;
  • As R1 we set a resistor with a value from the standard series closest to the measured one;
  • We replace R3 with a constant 470 Ohm chain + 3.3 kOhm potentiometer;
  • Same as according to paragraphs. 3-5, V. And we set the voltage equal to half the supply voltage.

Point a, from where the signal is removed to the load, is the so-called. midpoint of the amplifier. In UMZCH with unipolar power supply, it is set to half its value, and in UMZCH with bipolar power supply - zero relative to the common wire. This is called adjusting the amplifier balance. In unipolar UMZCHs with capacitive decoupling of the load, it is not necessary to turn it off during setup, but it is better to get used to doing this reflexively: an unbalanced 2-polar amplifier with a connected load can burn out its own powerful and expensive output transistors, or even a “new, good” and very expensive powerful speaker.

Note: components that require selection when setting up the device in the layout are indicated on the diagrams either with an asterisk (*) or an apostrophe (‘).

In the center of the same fig.- a simple UMZCH on transistors, already developing power up to 4-6 W at a load of 4 ohms. Although it works like the previous one, in the so-called. class AB1, not intended for Hi-Fi sound, but if you replace a pair of these class D amplifiers (see below) in cheap Chinese computer speakers, their sound improves noticeably. Here we learn another trick: powerful output transistors need to be placed on radiators. Components that require additional cooling are outlined in dotted lines in the diagrams; however, not always; sometimes - indicating the required dissipative area of ​​the heat sink. Setting up this UMZCH is balancing using R2.

On the right in Fig.- not yet a 350 W monster (as was shown at the beginning of the article), but already quite a solid beast: a simple amplifier with 100 W transistors. You can listen to music through it, but not Hi-Fi, operating class is AB2. However, it is quite suitable for scoring a picnic area or an outdoor meeting, a school assembly hall or a small shopping hall. An amateur rock band, having such a UMZCH per instrument, can perform successfully.

There are 2 more tricks in this UMZCH: firstly, in very powerful amplifiers, the drive stage of the powerful output also needs to be cooled, so VT3 is placed on a radiator of 100 kW or more. see. For output VT4 and VT5 radiators from 400 sq.m. are needed. see. Secondly, UMZCHs with bipolar power supply are not balanced at all without load. First one or the other output transistor goes into cutoff, and the associated one goes into saturation. Then, at full supply voltage, current surges during balancing can damage the output transistors. Therefore, for balancing (R6, guessed it?), the amplifier is powered from +/–24 V, and instead of a load, a wirewound resistor of 100...200 Ohms is switched on. By the way, the squiggles in some resistors in the diagram are Roman numerals, indicating their required heat dissipation power.

Note: A power source for this UMZCH needs a power of 600 W or more. Anti-aliasing filter capacitors - from 6800 µF at 160 V. In parallel with the electrolytic capacitors of the IP, 0.01 µF ceramic capacitors are included to prevent self-excitation at ultrasonic frequencies, which can instantly burn out the output transistors.

On the field workers

On the trail. rice. - another option for a fairly powerful UMZCH (30 W, and with a supply voltage of 35 V - 60 W) on powerful field-effect transistors:

The sound from it already meets the requirements for entry-level Hi-Fi (if, of course, the UMZCH works on the corresponding acoustic systems, speakers). Powerful field drivers do not require a lot of power to drive, so there is no pre-power cascade. Even more powerful field-effect transistors do not burn out the speakers in the event of any malfunction - they themselves burn out faster. Also unpleasant, but still cheaper than replacing an expensive loudspeaker bass head (GB). This UMZCH does not require balancing or adjustment in general. As a design for beginners, it has only one drawback: powerful field-effect transistors are much more expensive than bipolar transistors for an amplifier with the same parameters. Requirements for individual entrepreneurs are similar to previous ones. case, but its power is needed from 450 W. Radiators – from 200 sq. cm.

Note: there is no need to build powerful UMZCHs on field-effect transistors for switching power supplies, for example. computer When trying to “drive” them into the active mode required for UMZCH, they either simply burn out, or the sound is weak and “no quality at all.” The same applies to powerful high-voltage bipolar transistors, for example. from line scan of old TVs.

Straight up

If you have already taken the first steps, then it is quite natural to want to build Hi-Fi class UMZCH, without going too deep into the theoretical jungle. To do this, you will have to expand your instrumentation - you need an oscilloscope, an audio frequency generator (AFG) and an AC millivoltmeter with the ability to measure the DC component. It is better to take as a prototype for repetition the E. Gumeli UMZCH, described in detail in Radio No. 1, 1989. To build it, you will need a few inexpensive available components, but the quality meets very high requirements: power up to 60 W, band 20-20,000 Hz, frequency response unevenness 2 dB, nonlinear distortion factor (THD) 0.01%, self-noise level –86 dB. However, setting up the Gumeli amplifier is quite difficult; if you can handle it, you can take on any other. However, some of the currently known circumstances greatly simplify the establishment of this UMZCH, see below. Bearing in mind this and the fact that not everyone is able to get into the Radio archives, it would be appropriate to repeat the main points.

Schemes of a simple high-quality UMZCH

The Gumeli UMZCH circuits and specifications for them are shown in the illustration. Radiators of output transistors – from 250 sq. see for UMZCH in Fig. 1 and from 150 sq. see for option according to fig. 3 (original numbering). Transistors of the pre-output stage (KT814/KT815) are installed on radiators bent from 75x35 mm aluminum plates with a thickness of 3 mm. There is no need to replace KT814/KT815 with KT626/KT961; the sound does not noticeably improve, but setup becomes seriously difficult.

This UMZCH is very critical to power supply, installation topology and general, so it needs to be installed in a structurally complete form and only with a standard power source. When trying to power it from a stabilized power supply, the output transistors burn out immediately. Therefore, in Fig. Drawings of original printed circuit boards and setup instructions are provided. We can add to them that, firstly, if “excitement” is noticeable when you first turn it on, they fight it by changing the inductance L1. Secondly, the leads of parts installed on boards should be no longer than 10 mm. Thirdly, it is extremely undesirable to change the installation topology, but if it is really necessary, there must be a frame shield on the side of the conductors (ground loop, highlighted in color in the figure), and the power supply paths must pass outside it.

Note: breaks in the tracks to which the bases of powerful transistors are connected - technological, for adjustment, after which they are sealed with drops of solder.

Setting up this UMZCH is greatly simplified, and the risk of encountering “excitement” during use is reduced to zero if:

  • Minimize interconnect installation by placing the boards on radiators of powerful transistors.
  • Completely abandon the connectors inside, performing all installation only by soldering. Then there will be no need for R12, R13 in a powerful version or R10 R11 in a less powerful version (they are dotted in the diagrams).
  • Use oxygen-free copper audio wires of minimum length for internal installation.

If these conditions are met, there are no problems with excitation, and setting up the UMZCH comes down to the routine procedure described in Fig.

Wires for sound

Audio wires are not an idle invention. The need for their use at present is undeniable. In copper with an admixture of oxygen, a thin oxide film is formed on the faces of metal crystallites. Metal oxides are semiconductors and if the current in the wire is weak without a constant component, its shape is distorted. In theory, distortions on myriads of crystallites should compensate each other, but very little (apparently due to quantum uncertainties) remains. Sufficient to be noticed by discerning listeners against the background of the purest sound of modern UMZCH.

Manufacturers and traders shamelessly substitute ordinary electrical copper instead of oxygen-free copper - it is impossible to distinguish one from the other by eye. However, there is an area of ​​application where counterfeiting is not clear: twisted pair cable for computer networks. If you put a grid with long segments on the left, it will either not start at all or will constantly glitch. Momentum dispersion, you know.

The author, when there was just talk about audio wires, realized that, in principle, this was not idle chatter, especially since oxygen-free wires by that time had long been used in special-purpose equipment, with which he was well acquainted by his line of work. Then I took and replaced the standard cord of my TDS-7 headphones with a homemade one made from “vitukha” with flexible multi-core wires. The sound, aurally, has steadily improved for end-to-end analogue tracks, i.e. on the way from the studio microphone to the disc, never digitized. Vinyl recordings made using DMM (Direct Metal Mastering) technology sounded especially bright. After this, the interconnect installation of all home audio was converted to “vitushka”. Then completely random people, indifferent to the music and not notified in advance, began to notice the improvement in sound.

How to make interconnect wires from twisted pair, see next. video.

Video: do-it-yourself twisted pair interconnect wires

Unfortunately, the flexible “vitha” soon disappeared from sale - it did not hold well in the crimped connectors. However, for the information of readers, flexible “military” wire MGTF and MGTFE (shielded) is made only from oxygen-free copper. Fake is impossible, because On ordinary copper, tape fluoroplastic insulation spreads quite quickly. MGTF is now widely available and costs much less than branded audio cables with a guarantee. It has one drawback: it cannot be done in color, but this can be corrected with tags. There are also oxygen-free winding wires, see below.

Theoretical Interlude

As we can see, already in the early stages of mastering audio technology, we had to deal with the concept of Hi-Fi (High Fidelity), high fidelity sound reproduction. Hi-Fi comes in different levels, which are ranked according to the following. main parameters:

  1. Reproducible frequency band.
  2. Dynamic range - the ratio in decibels (dB) of the maximum (peak) output power to the noise level.
  3. Self-noise level in dB.
  4. Nonlinear distortion factor (THD) at rated (long-term) output power. The SOI at peak power is assumed to be 1% or 2% depending on the measurement technique.
  5. Unevenness of the amplitude-frequency response (AFC) in the reproducible frequency band. For speakers - separately at low (LF, 20-300 Hz), medium (MF, 300-5000 Hz) and high (HF, 5000-20,000 Hz) sound frequencies.

Note: the ratio of absolute levels of any values ​​of I in (dB) is defined as P(dB) = 20log(I1/I2). If I1

You need to know all the subtleties and nuances of Hi-Fi when designing and building speakers, and as for a homemade Hi-Fi UMZCH for the home, before moving on to these, you need to clearly understand the requirements for their power required to sound a given room, dynamic range (dynamics), noise level and SOI. It is not very difficult to achieve a frequency band of 20-20,000 Hz from the UMZCH with a roll off at the edges of 3 dB and an uneven frequency response in the midrange of 2 dB on a modern element base.

Volume

The power of the UMZCH is not an end in itself; it must provide the optimal volume of sound reproduction in a given room. It can be determined by curves of equal loudness, see fig. There are no natural noises in residential areas quieter than 20 dB; 20 dB is the wilderness in complete calm. A volume level of 20 dB relative to the threshold of audibility is the threshold of intelligibility - a whisper can still be heard, but music is perceived only as the fact of its presence. An experienced musician can tell which instrument is being played, but not what exactly.

40 dB - the normal noise of a well-insulated city apartment in a quiet area or a country house - represents the intelligibility threshold. Music from the threshold of intelligibility to the threshold of intelligibility can be listened to with deep frequency response correction, primarily in the bass. To do this, the MUTE function (mute, mutation, not mutation!) is introduced into modern UMZCHs, including, respectively. correction circuits in UMZCH.

90 dB is the volume level of a symphony orchestra in a very good concert hall. 110 dB can be produced by an extended orchestra in a hall with unique acoustics, of which there are no more than 10 in the world, this is the threshold of perception: louder sounds are still perceived as distinguishable in meaning with an effort of will, but already annoying noise. The volume zone in residential premises of 20-110 dB constitutes the zone of complete audibility, and 40-90 dB is the zone of best audibility, in which untrained and inexperienced listeners fully perceive the meaning of the sound. If, of course, he is in it.

Power

Calculating the power of equipment at a given volume in the listening area is perhaps the main and most difficult task of electroacoustics. For yourself, in conditions it is better to go from acoustic systems (AS): calculate their power using a simplified method, and take the nominal (long-term) power of the UMZCH equal to the peak (musical) speaker. In this case, the UMZCH will not noticeably add its distortions to those of the speakers; they are already the main source of nonlinearity in the audio path. But the UMZCH should not be made too powerful: in this case, the level of its own noise may be higher than the threshold of audibility, because It is calculated based on the voltage level of the output signal at maximum power. If we consider it very simply, then for a room in an ordinary apartment or house and speakers with normal characteristic sensitivity (sound output) we can take the trace. UMZCH optimal power values:

  • Up to 8 sq. m – 15-20 W.
  • 8-12 sq. m – 20-30 W.
  • 12-26 sq. m – 30-50 W.
  • 26-50 sq. m – 50-60 W.
  • 50-70 sq. m – 60-100 W.
  • 70-100 sq. m – 100-150 W.
  • 100-120 sq. m – 150-200 W.
  • More than 120 sq. m – determined by calculation based on on-site acoustic measurements.

Dynamics

The dynamic range of the UMZCH is determined by curves of equal loudness and threshold values ​​for different degrees of perception:

  1. Symphonic music and jazz with symphonic accompaniment - 90 dB (110 dB - 20 dB) ideal, 70 dB (90 dB - 20 dB) acceptable. No expert can distinguish a sound with a dynamics of 80-85 dB in a city apartment from ideal.
  2. Other serious music genres – 75 dB excellent, 80 dB “through the roof”.
  3. Pop music of any kind and movie soundtracks - 66 dB is enough for the eyes, because... These opuses are already compressed during recording to levels of up to 66 dB and even up to 40 dB, so that you can listen to them on anything.

The dynamic range of the UMZCH, correctly selected for a given room, is considered equal to its own noise level, taken with the + sign, this is the so-called. signal-to-noise ratio.

SOI

Nonlinear distortions (ND) of UMZCH are components of the output signal spectrum that were not present in the input signal. Theoretically, it is best to “push” the NI under the level of its own noise, but technically this is very difficult to implement. In practice, they take into account the so-called. masking effect: at volume levels below approx. At 30 dB, the range of frequencies perceived by the human ear narrows, as does the ability to distinguish sounds by frequency. Musicians hear notes, but find it difficult to assess the timbre of the sound. In people without a hearing for music, the masking effect is observed already at 45-40 dB of volume. Therefore, an UMZCH with a THD of 0.1% (–60 dB from a volume level of 110 dB) will be assessed as Hi-Fi by the average listener, and with a THD of 0.01% (–80 dB) can be considered not distorting the sound.

Lamps

The last statement will probably cause rejection, even fury, among adherents of tube circuitry: they say, real sound is produced only by tubes, and not just some, but certain types of octal ones. Calm down, gentlemen - the special tube sound is not a fiction. The reason is the fundamentally different distortion spectra of electronic tubes and transistors. Which, in turn, are due to the fact that in the lamp the flow of electrons moves in a vacuum and quantum effects do not appear in it. A transistor is a quantum device, where minority charge carriers (electrons and holes) move in the crystal, which is completely impossible without quantum effects. Therefore, the spectrum of tube distortions is short and clean: only harmonics up to the 3rd - 4th are clearly visible in it, and there are very few combinational components (sums and differences in the frequencies of the input signal and their harmonics). Therefore, in the days of vacuum circuitry, SOI was called harmonic distortion (CHD). In transistors, the spectrum of distortions (if they are measurable, the reservation is random, see below) can be traced up to the 15th and higher components, and there are more than enough combination frequencies in it.

At the beginning of solid-state electronics, designers of transistor UMZCHs used the usual “tube” SOI of 1-2% for them; Sound with a tube distortion spectrum of this magnitude is perceived by ordinary listeners as pure. By the way, the very concept of Hi-Fi did not yet exist. It turned out that they sound dull and dull. In the process of developing transistor technology, an understanding of what Hi-Fi is and what is needed for it was developed.

Currently, the growing pains of transistor technology have been successfully overcome and side frequencies at the output of a good UMZCH are difficult to detect using special measurement methods. And lamp circuitry can be considered to have become an art. Its basis can be anything, why can’t electronics go there? An analogy with photography would be appropriate here. No one can deny that a modern digital SLR camera produces an image that is immeasurably clearer, more detailed, and deeper in the range of brightness and color than a plywood box with an accordion. But someone, with the coolest Nikon, “clicks pictures” like “this is my fat cat, he got drunk like a bastard and is sleeping with his paws outstretched,” and someone, using Smena-8M, uses Svemov’s b/w film to take a picture in front of which there is a crowd of people at a prestigious exhibition.

Note: and calm down again - not everything is so bad. Today, low-power lamp UMZCHs have at least one application left, and not the least important, for which they are technically necessary.

Experimental stand

Many audio lovers, having barely learned to solder, immediately “go into tubes.” This in no way deserves censure, on the contrary. Interest in the origins is always justified and useful, and electronics has become so with tubes. The first computers were tube-based, and the on-board electronic equipment of the first spacecraft was also tube-based: there were already transistors then, but they could not withstand extraterrestrial radiation. By the way, at that time lamp microcircuits were also created under the strictest secrecy! On microlamps with a cold cathode. The only known mention of them in open sources is in the rare book by Mitrofanov and Pickersgil “Modern receiving and amplifying tubes”.

But enough of the lyrics, let's get to the point. For those who like to tinker with the lamps in Fig. – diagram of a bench lamp UMZCH, intended specifically for experiments: SA1 switches the operating mode of the output lamp, and SA2 switches the supply voltage. The circuit is well known in the Russian Federation, a minor modification affected only the output transformer: now you can not only “drive” the native 6P7S in different modes, but also select the screen grid switching factor for other lamps in ultra-linear mode; for the vast majority of output pentodes and beam tetrodes it is either 0.22-0.25 or 0.42-0.45. For the manufacture of the output transformer, see below.

Guitarists and rockers

This is the very case when you can’t do without lamps. As you know, the electric guitar became a full-fledged solo instrument after the pre-amplified signal from the pickup began to be passed through a special attachment - a fuser - which deliberately distorted its spectrum. Without this, the sound of the string was too sharp and short, because the electromagnetic pickup reacts only to the modes of its mechanical vibrations in the plane of the instrument soundboard.

An unpleasant circumstance soon emerged: the sound of an electric guitar with a fuser acquires full strength and brightness only at high volumes. This is especially true for guitars with a humbucker-type pickup, which gives the most “angry” sound. But what about a beginner who is forced to rehearse at home? You can’t go to the hall to perform without knowing exactly how the instrument will sound there. And rock fans just want to listen to their favorite things in full juice, and rockers are generally decent and non-conflict people. At least those who are interested in rock music, and not shocking surroundings.

So, it turned out that the fatal sound appears at volume levels acceptable for residential premises, if the UMZCH is tube-based. The reason is the specific interaction of the signal spectrum from the fuser with the pure and short spectrum of tube harmonics. Here again an analogy is appropriate: a b/w photo can be much more expressive than a color one, because leaves only the outline and light for viewing.

Those who need a tube amplifier not for experiments, but due to technical necessity, do not have time to master the intricacies of tube electronics for a long time, they are passionate about something else. In this case, it is better to make the UMZCH transformerless. More precisely, with a single-ended matching output transformer that operates without constant magnetization. This approach greatly simplifies and speeds up the production of the most complex and critical component of a lamp UMZCH.

“Transformerless” tube output stage of the UMZCH and pre-amplifiers for it

On the right in Fig. a diagram of a transformerless output stage of a tube UMZCH is given, and on the left are pre-amplifier options for it. At the top - with a tone control according to the classic Baxandal scheme, which provides fairly deep adjustment, but introduces slight phase distortion into the signal, which can be significant when operating an UMZCH on a 2-way speaker. Below is a preamplifier with simpler tone control that does not distort the signal.

But let's get back to the end. In a number of foreign sources, this scheme is considered a revelation, but an identical one, with the exception of the capacitance of the electrolytic capacitors, is found in the Soviet “Radio Amateur Handbook” of 1966. A thick book of 1060 pages. There was no Internet and disk-based databases back then.

In the same place, on the right in the figure, the disadvantages of this scheme are briefly but clearly described. An improved one, from the same source, is given on the trail. rice. on right. In it, the screen grid L2 is powered from the midpoint of the anode rectifier (the anode winding of the power transformer is symmetrical), and the screen grid L1 is powered through the load. If, instead of high-impedance speakers, you turn on a matching transformer with regular speakers, as in the previous one. circuit, the output power is approx. 12 W, because the active resistance of the primary winding of the transformer is much less than 800 Ohms. SOI of this final stage with transformer output - approx. 0.5%

How to make a transformer?

The main enemies of the quality of a powerful signal low-frequency (sound) transformer are the magnetic leakage field, the lines of force of which are closed, bypassing the magnetic circuit (core), eddy currents in the magnetic circuit (Foucault currents) and, to a lesser extent, magnetostriction in the core. Because of this phenomenon, a carelessly assembled transformer “sings,” hums, or beeps. Foucault currents are combated by reducing the thickness of the magnetic circuit plates and additionally insulating them with varnish during assembly. For output transformers, the optimal plate thickness is 0.15 mm, the maximum allowable is 0.25 mm. You should not take thinner plates for the output transformer: the fill factor of the core (the central rod of the magnetic circuit) with steel will fall, the cross-section of the magnetic circuit will have to be increased to obtain a given power, which will only increase distortions and losses in it.

In the core of an audio transformer operating with constant bias (for example, the anode current of a single-ended output stage) there must be a small (determined by calculation) non-magnetic gap. The presence of a non-magnetic gap, on the one hand, reduces signal distortion from constant magnetization; on the other hand, in a conventional magnetic circuit it increases the stray field and requires a core with a larger cross-section. Therefore, the non-magnetic gap must be calculated at the optimum and performed as accurately as possible.

For transformers operating with magnetization, the optimal type of core is made of Shp (cut) plates, pos. 1 in Fig. In them, a non-magnetic gap is formed during core cutting and is therefore stable; its value is indicated in the passport for the plates or measured with a set of probes. The stray field is minimal, because the side branches through which the magnetic flux is closed are solid. Transformer cores without bias are often assembled from Shp plates, because Shp plates are made from high-quality transformer steel. In this case, the core is assembled across the roof (the plates are laid with a cut in one direction or the other), and its cross-section is increased by 10% compared to the calculated one.

It is better to wind transformers without magnetization on USH cores (reduced height with widened windows), pos. 2. In them, a decrease in the stray field is achieved by reducing the length of the magnetic path. Since USh plates are more accessible than Shp, transformer cores with magnetization are often made from them. Then the core assembly is carried out cut to pieces: a package of W-plates is assembled, a strip of non-conducting non-magnetic material is placed with a thickness equal to the size of the non-magnetic gap, covered with a yoke from a package of jumpers and pulled together with a clip.

Note:“sound” signal magnetic circuits of the ShLM type are of little use for output transformers of high-quality tube amplifiers; they have a large stray field.

At pos. 3 shows a diagram of the core dimensions for calculating the transformer, at pos. 4 design of the winding frame, and at pos. 5 – patterns of its parts. As for the transformer for the “transformerless” output stage, it is better to make it on the ShLMm across the roof, because the bias is negligible (the bias current is equal to the screen grid current). The main task here is to make the windings as compact as possible in order to reduce the stray field; their active resistance will still be much less than 800 Ohms. The more free space left in the windows, the better the transformer turned out. Therefore, the windings are wound turn to turn (if there is no winding machine, this is a terrible task) from the thinnest possible wire; the laying coefficient of the anode winding for the mechanical calculation of the transformer is taken 0.6. The winding wire is PETV or PEMM, they have an oxygen-free core. There is no need to take PETV-2 or PEMM-2; due to double varnishing, they have an increased outer diameter and a larger scattering field. The primary winding is wound first, because it is its scattering field that most affects the sound.

You need to look for iron for this transformer with holes in the corners of the plates and clamping brackets (see figure on the right), because “for complete happiness,” the magnetic circuit is assembled as follows. order (of course, the windings with leads and external insulation should already be on the frame):

  1. Prepare acrylic varnish diluted in half or, in the old fashioned way, shellac;
  2. Plates with jumpers are quickly coated with varnish on one side and placed into the frame as quickly as possible, without pressing too hard. The first plate is placed with the varnished side inward, the next one with the unvarnished side to the first varnished, etc.;
  3. When the frame window is filled, staples are applied and bolted tightly;
  4. After 1-3 minutes, when the squeezing of varnish from the gaps apparently stops, add plates again until the window is filled;
  5. Repeat paragraphs. 2-4 until the window is tightly packed with steel;
  6. The core is pulled tightly again and dried on a battery, etc. 3-5 days.

The core assembled using this technology has very good plate insulation and steel filling. Magnetostriction losses are not detected at all. But keep in mind that this technique is not applicable for permalloy cores, because Under strong mechanical influences, the magnetic properties of permalloy irreversibly deteriorate!

On microcircuits

UMZCHs on integrated circuits (ICs) are most often made by those who are satisfied with the sound quality up to average Hi-Fi, but are more attracted by the low cost, speed, ease of assembly and the complete absence of any setup procedures that require special knowledge. Simply, an amplifier on microcircuits is the best option for dummies. The classic of the genre here is the UMZCH on the TDA2004 IC, which has been on the series, God willing, for about 20 years now, on the left in Fig. Power – up to 12 W per channel, supply voltage – 3-18 V unipolar. Radiator area – from 200 sq. see for maximum power. The advantage is the ability to work with a very low-resistance, up to 1.6 Ohm, load, which allows you to extract full power when powered from a 12 V on-board network, and 7-8 W when supplied with a 6-volt power supply, for example, on a motorcycle. However, the output of the TDA2004 in class B is not complementary (on transistors of the same conductivity), so the sound is definitely not Hi-Fi: THD 1%, dynamics 45 dB.

The more modern TDA7261 does not produce better sound, but is more powerful, up to 25 W, because The upper limit of the supply voltage has been increased to 25 V. The lower limit, 4.5 V, still allows it to be powered from a 6 V on-board network, i.e. The TDA7261 can be started from almost all on-board networks, except for the aircraft 27 V. Using attached components (strapping, on the right in the figure), the TDA7261 can operate in mutation mode and with the St-By (Stand By) function, which switches the UMZCH to the minimum power consumption mode when there is no input signal for a certain time. Convenience costs money, so for a stereo you will need a pair of TDA7261 with radiators from 250 sq. see for each.

Note: If you are somehow attracted to amplifiers with the St-By function, keep in mind that you should not expect speakers wider than 66 dB from them.

“Super economical” in terms of power supply TDA7482, on the left in the figure, operating in the so-called. class D. Such UMZCHs are sometimes called digital amplifiers, which is incorrect. For real digitization, level samples are taken from an analog signal with a quantization frequency that is no less than twice the highest of the reproduced frequencies, the value of each sample is recorded in a noise-resistant code and stored for further use. UMZCH class D – pulse. In them, the analogue is directly converted into a sequence of high-frequency pulse-width modulated (PWM), which is fed to the speaker through a low-pass filter (LPF).

Class D sound has nothing in common with Hi-Fi: SOI of 2% and dynamics of 55 dB for class D UMZCH are considered very good indicators. And TDA7482 here, it must be said, is not the optimal choice: other companies specializing in class D produce UMZCH ICs that are cheaper and require less wiring, for example, D-UMZCH of the Paxx series, on the right in Fig.

Among the TDAs, the 4-channel TDA7385 should be noted, see the figure, on which you can assemble a good amplifier for speakers up to medium Hi-Fi, inclusive, with frequency division into 2 bands or for a system with a subwoofer. In both cases, low-pass and mid-high-frequency filtering is done at the input on a weak signal, which simplifies the design of the filters and allows deeper separation of the bands. And if the acoustics are subwoofer, then 2 channels of the TDA7385 can be allocated for the sub-ULF bridge circuit (see below), and the remaining 2 can be used for MF-HF.

UMZCH for subwoofer

A subwoofer, which can be translated as “subwoofer” or, literally, “boomer,” reproduces frequencies up to 150-200 Hz; in this range, human ears are practically unable to determine the direction of the sound source. In speakers with a subwoofer, the “sub-bass” speaker is placed in a separate acoustic design, this is the subwoofer as such. The subwoofer is placed, in principle, as conveniently as possible, and the stereo effect is provided by separate MF-HF channels with their own small-sized speakers, for the acoustic design of which there are no particularly serious requirements. Experts agree that it is better to listen to stereo with full channel separation, but subwoofer systems significantly save money or labor on the bass path and make it easier to place acoustics in small rooms, which is why they are popular among consumers with normal hearing and not particularly demanding ones.

The “leakage” of mid-high frequencies into the subwoofer, and from it into the air, greatly spoils the stereo, but if you sharply “cut off” the sub-bass, which, by the way, is very difficult and expensive, then a very unpleasant sound jumping effect will arise. Therefore, channels in subwoofer systems are filtered twice. At the input, electric filters highlight midrange-high frequencies with bass “tails” that do not overload the midrange-high frequency path, but provide a smooth transition to sub-bass. Bass with midrange “tails” are combined and fed to a separate UMZCH for the subwoofer. The midrange is additionally filtered so that the stereo does not deteriorate; in the subwoofer it is already acoustic: a sub-bass speaker is placed, for example, in the partition between the resonator chambers of the subwoofer, which do not let the midrange out, see on the right in Fig.

A UMZCH for a subwoofer is subject to a number of specific requirements, of which “dummies” consider the most important to be as high a power as possible. This is completely wrong, if, say, the calculation of the acoustics for the room gave a peak power W for one speaker, then the power of the subwoofer needs 0.8 (2W) or 1.6W. For example, if S-30 speakers are suitable for the room, then a subwoofer needs 1.6x30 = 48 W.

It is much more important to ensure the absence of phase and transient distortions: if they occur, there will definitely be a jump in the sound. As for SOI, it is permissible up to 1%. Intrinsic bass distortion of this level is not audible (see curves of equal volume), and the “tails” of their spectrum in the best audible midrange region will not come out of the subwoofer.

To avoid phase and transient distortions, the amplifier for the subwoofer is built according to the so-called. bridge circuit: the outputs of 2 identical UMZCHs are switched on back-to-back through a speaker; signals to the inputs are supplied in antiphase. The absence of phase and transient distortions in the bridge circuit is due to the complete electrical symmetry of the output signal paths. The identity of the amplifiers forming the arms of the bridge is ensured by the use of paired UMZCHs on ICs, made on the same chip; This is perhaps the only case when an amplifier on microcircuits is better than a discrete one.

Note: The power of a bridge UMZCH does not double, as some people think, it is determined by the supply voltage.

An example of a bridge UMZCH circuit for a subwoofer in a room up to 20 sq. m (without input filters) on the TDA2030 IC is given in Fig. left. Additional midrange filtering is carried out by circuits R5C3 and R’5C’3. Radiator area TDA2030 – from 400 sq. see. Bridged UMZCHs with an open output have an unpleasant feature: when the bridge is unbalanced, a constant component appears in the load current, which can damage the speaker, and the sub-bass protection circuits often fail, turning off the speaker when not needed. Therefore, it is better to protect the expensive oak bass head with non-polar batteries of electrolytic capacitors (highlighted in color, and the diagram of one battery is given in the inset.

A little about acoustics

The acoustic design of a subwoofer is a special topic, but since a drawing is given here, explanations are also needed. Case material – MDF 24 mm. The resonator tubes are made of fairly durable, non-ringing plastic, for example, polyethylene. The internal diameter of the pipes is 60 mm, the protrusions inward are 113 mm in the large chamber and 61 in the small chamber. For a specific loudspeaker head, the subwoofer will have to be reconfigured for the best bass and, at the same time, the least impact on the stereo effect. To tune the pipes, they take a pipe that is obviously longer and, by pushing it in and out, achieve the required sound. The protrusions of the pipes outward do not affect the sound; they are then cut off. The pipe settings are interdependent, so you will have to tinker.

Headphone Amplifier

A headphone amplifier is most often made by hand for two reasons. The first is for listening “on the go”, i.e. outside the home, when the power of the audio output of the player or smartphone is not enough to drive “buttons” or “burdocks”. The second is for high-end home headphones. A Hi-Fi UMZCH for an ordinary living room is needed with dynamics of up to 70-75 dB, but the dynamic range of the best modern stereo headphones exceeds 100 dB. An amplifier with such dynamics costs more than some cars, and its power will be from 200 W per channel, which is too much for an ordinary apartment: listening at a power that is much lower than the rated power spoils the sound, see above. Therefore, it makes sense to make a low-power, but with good dynamics, a separate amplifier specifically for headphones: the prices for household UMZCHs with such an additional weight are clearly absurdly inflated.

The circuit of the simplest headphone amplifier using transistors is given in pos. 1 pic. The sound is only for Chinese “buttons”, it works in class B. It is also no different in terms of efficiency - 13 mm lithium batteries last for 3-4 hours at full volume. At pos. 2 – TDA’s classic for on-the-go headphones. The sound, however, is quite decent, up to average Hi-Fi depending on the track digitization parameters. There are countless amateur improvements to the TDA7050 harness, but no one has yet achieved the transition of sound to the next level of class: the “microphone” itself does not allow it. TDA7057 (item 3) is simply more functional; you can connect the volume control to a regular, not dual, potentiometer.

The UMZCH for headphones on the TDA7350 (item 4) is designed to drive good individual acoustics. It is on this IC that headphone amplifiers in most middle and high-class household UMZCHs are assembled. The UMZCH for headphones on KA2206B (item 5) is already considered professional: its maximum power of 2.3 W is enough to drive such serious isodynamic “mugs” as TDS-7 and TDS-15.

Amplifiers whose main purpose is to amplify the signal by power are called power amplifiers. As a rule, such amplifiers drive a low-impedance load, such as a loudspeaker.

3-18 V (nominal - 6 V). The maximum current consumption is 1.5 A with a quiescent current of 7 mA (at 6 V) and 12 mA (at 18 V). Voltage gain 36.5 dB. at -1 dB 20 Hz - 300 kHz. Rated output power at 10% THD

temporarily turn off the sound. You can double the output power of the TDA7233D when you turn them on according to the circuit shown in Fig. 31.42. C7 prevents self-excitation of the device in the area

high frequencies. R3 is selected until an equal amplitude of the output signals is obtained at the outputs of the microcircuits.

Rice. 31.43. KR174UNZ 7

KR174UN31 is intended for use as output low-power household electronic devices.

When the supply voltage changes from

2.1 to 6.6 V with an average current consumption of 7 mA (without input signal), the voltage gain of the microcircuit varies from 18 to 24 dB.

The coefficient of nonlinear distortion at an output power of up to 100 mW is no more than 0.015%, the output noise voltage does not exceed 100 μV. The input of the microcircuit is 35-50 kOhm. load - not lower than 8 Ohms. Operating frequency range - 20 Hz - 30 kHz, limit - 10 Hz - 100 kHz. The maximum input signal voltage is up to 0.25-0.5 V.

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