Microphone amplifier circuits with regulation. High quality amplifier without feedback: The End Millennium. Using DC output voltage protection

The End Millenium is a high-end power amplifier with a power range from 99 to 300 watts (at 8 ohms). The use of high-quality class A/B amplifiers is achieved by a number of circuit solutions. First of all, attention is drawn to the absence of any chains feedback, because Even if it corrects the error of the signal received at the input, after it it is no longer reversible. A simple circuit design, together with high quality components, ensures a short signal path from input to output. The use of high-tech components can be noted by the use of polypropylene capacitors, multi-emitter bipolar transistors and miniature resistors on a glass substrate.

The higher frequencies of the range are easily reproduced by the ultra-fast amplifier (linearity up to > 500,000Hz), and the use of a four-stage tunnel at the output gives the signature fast transmission of low frequencies. The overall scene is well detailed and transparent.

Schematic diagram of the The End Millennium amplifier:

The circuit diagram shows how simply the amplifier idea is implemented. Absence of feedback circuits (100% without feedback), absence of capacitors and other components that introduce distortion into the signal in the signal circuits. The frequency response is linear from DC to the maximum high-frequency signal - 500,000 Hz. This may be the fastest amplifier you've ever heard! Any part of the musical accompaniment, from the deepest bass to the smallest transitions, is transmitted by the amplifier with ease.

The amplifier board also contains additional features such as DC protection and output short circuit protection. The protection monitors the occurrence of any overload at the output and turns off the amplifier for a few seconds. No current or signal limitations are used. If an error is detected, the device automatically turns off and waits for the situation to normalize. It will then turn on and continue playing. This system is so effective that it allows the output to short circuit for several days!

Thanks to the new amplifier topology, which, in fact, in some aspects destroys generally accepted principles, it has become possible to build an amplifier with a well-controlled sound picture, a moving stage with a high degree of detail, very affordable price. The low cost is achieved mainly by the fact that you do the assembly yourself.

The four-stage tunnel output stage allows you to accurately transmit the amplified signal from the source to the sound head membrane. Not only start the movement of the membrane, but also stop it in a microsecond.

100% no OS = 100% musicality

The soft, almost intimate sound is mainly due to the amplifier's circuit design, which does not contain the usual feedback circuit in such cases. This design principle is usually called 100% no feedback and is also used in the designs of other brands of high-end amplifiers (usually very expensive).

In conventional amplifiers (with a feedback circuit), the typical approach is to use circuits with large gains (Kus up to 100,000) and a high degree of signal distortion in order to achieve the required voltage gain. By comparing the shape of the output signal with respect to the input signal, it is possible to correct the transmission error and thus reduce the measured harmonic distortion. However, such an error cannot be corrected until it is detected and reproduced by the sound head, which is also connected to the distorted signal. This can be compared to an attempt to suppress waves in a swimming pool by creating the same waves in antiphase. Not practical, and the waves are too low in frequency to match the time required to reach the corrective waves on the other side of the pool.

Another problem arises when you try to linearize a signal that has been amplified by a nonlinear (signal-distorting) element. Inevitable modulation occurs, previously called intermodulation signal distortion. This annoying misunderstanding can be described as if two vocalists are singing at the same time, and you hear a third, inharmonious, annoying tone. At best, this can be eliminated by losing the frequency range, but it is still a loss. Another way to hear intermodulation distortion in a conventional amplifier is by turning the signal volume up or down.

Millennium reproduces the signal regardless of the volume level and dynamic range. It uses a completely different principle for correcting distortion. In non-OS circuits, it is impossible to get rid of distortions once they have already occurred, so all measures are taken to prevent their occurrence. Ultra linear semiconductors, high stability resistors, no capacitors and looped PCB traces for all audio signal circuits. All components used in the design are top class recognized market leaders from manufacturers, which can also only be found in high-quality amplifiers in an exorbitant price range.

The result is a circuit that is not overloaded with complexity and clear sound without modulation, but with good detail and musical dynamics.

The English-made Z-transistor is a bipolar vertical transistor created using technology usually applied to the production of MOSFET transistors. However, it has a significantly lower junction resistance (Re or Rs) than a FET or MOSFET and therefore introduces less distortion into the signal.

Low junction capacitance (6 pF) and very low noise figure are also advantages.

Millennium High Voltage Circuits

Initially, the Millennium was conceived as an amplifier with a power of 120 Watts into a load of 8 Ohms or 240 Watts into a load of 4 Ohms with a transformer supply of 33-0-33 Volts. But by adding additional output stage modules you can use it for more high power or lower load resistances (down to 1 Ohm). When powering the amplifier 40-0-40: one additional module provides 180 Watts into an 8 Ohm load, two modules provide 350 Watts into a 4 Ohm load. With a power supply of 50-0-50 Volts: three modules - 250 Watts at 8 Ohms, 500 Watts at 4 Ohms.

The add-on module parts are placed on a separate board, which also contains emitter resistors and associated bypass capacitors to ensure stage stability.

An increase in output power is also possible by reducing the load resistance when powered at 33-0-33 Volts, more than 800 Watts at a load of 1 Ohm.

To avoid loss of quality, it is not recommended to use additional modules at the output of devices that will be designed to reproduce high-frequency and mid-range frequencies. A parallel module will inevitably have differences in the characteristics of the transistors, which will lead to the appearance of higher harmonics in the signal, manifesting itself as an aggressive sound at high signal volumes. A solution may be to use separate outputs for the LF and MF/HF channels. Although this will require speakers with separate channels, most modern loudspeakers have this option. In this case, one output channel will be loaded onto the mid/high frequency section, and a number of additional modules will be loaded onto a more powerful bass output, where higher harmonics will be cut off by the speaker input filter.

Separate output connectors are standard on our 180 Watt and larger sets. (Except for balanced input versions where parallel output stages are not used in any case)

Additional output module board with emitter resistors and blocking capacitors - up to three boards at a time. They are connected to the main board with power and input/output signal wires.

“The End” is the most successful audio design in Scandinavia!

Any Scandinavian radio amateur knows the previous version 3.1 design. More than 3,600 of these DIY kits were sold between 1995 and 1999 until the Millennium. Almost all of them now work in hundreds of different audio systems, confirming the unusually high quality of reproduction.

In the Millennium version it is improved in all aspects:

Four-stage output tunnel bass pumping

Glass backed resistors for better linearity and uniformity

Signal amplification with specially designed Z-transistors with very low Re and output capacitance (Сс=6 pF).

Low signal distortion due to ultra linear core topology.

High frequency detailing through the use of 4.7 mF blocking capacitors with a polypropylene separator on the power buses.

All PCB traces related to the audio signal have rounded vias. This prevents the occurrence standing waves and contributes to more accurate and correct reproduction.

In addition, several additional functions have been added to the compact board, made of high-quality Fr4 fiberglass. The switchable protection function will react to the appearance of a constant voltage at the output of 5 mV, and effective protection will protect your amplifier from short circuits even under extreme overloads.

The bias system, subject to temperature conditions for supply voltages of +/- 100 Volts, ensures long-term operation in any application. Millennium is also stable when powered down to +/- 10 Volts.

Nutritional Considerations

Amplifier power supply is very critical for playback quality!

If you are planning to build the perfect power supply for an amplifier, the most attractive would be to use a bank of (Swedish) RIFA capacitors from 100,000 uF each. Add blocking inductors to reduce charging currents and you have the best power supply for an audio system.

However, the price and size of the installation with this approach make it less attractive. It's too expensive and will take up about the same amount of space as a small refrigerator. Therefore, we have developed a "Super-Duper" Power Supply that is more rationally designed than the bulky but simple solution from RIFA.

120,000 µF of American low-impedance capacitors from ChemiCon are allocated to separately power the high-power and sensitive signal stages, so any power dips caused by overloading the power stages will not affect the input and driver circuits.

In addition, a set of polycarbonate capacitors helps reduce high-frequency noise from the rectifier.

These two 4.7 mF capacitors are marked on the board, but are now mounted on the amplifier board rather than the power supply.

AUX output is used to power the voltage amplifier and drivers.

A capacity reserve of 120,000 uF ensures complete stability and sufficient power to power even critical loads. The ChemiCon brand was formerly known as Sprague.

Complete The End Millenium Amplifier Circuit

Scale Not 1:1

Board size: 107x54mm

Photo of the amplifier board

"Hatsink placed here" - Radiator installation location

"BIAS Testpoint" - Bias setting test point

Assembly instructions

Assembly of the Millennium is not difficult and does not take much time.

Start by emptying all the parts from the bag onto the table.

Heat up the soldering iron.

Start by installing low-profile components such as resistors and trimmers. Check the numbering of elements on the diagram with that written on the board itself and compare with the color code printed in the table on the previous page. If you are sure that everything is installed correctly, proceed to soldering. After this, install capacitors, first small ones, then larger ones. Solder it.

Two 470 mF electrolytes are installed with reverse side, do not confuse the polarity; the strip indicating minus on both faces the near edge of the board.

Install them onto the board before cutting the leads and soldering them.

Now install the T9 and drivers (be careful, they are installed on their own side) as high as the length of the leads allows. They must be at the correct angle to the board.

After this, screw the drivers onto the heatsink using short 3mm screws and small spacers. No grease is allowed on them and they must fit tightly to the gasket without an air gap. The picture shows that 4m7 capacitors are also already installed, but it will be a little easier if you wait.

Place a thermal pad over the output transistor mounting location and install cardboard washers under the mounting screws. The use of lubricants is not permitted!

Secure each Sanken to the CORRECT place on the board, with the metal backing to the spacer. Make sure that there are no foreign matter (chips, dirt) under the gasket. Use gasket and screws bigger size. Tighten the screws as tightly as possible without tearing them off.

Then solder them to the board and trim the leads.

Now install the 4.7 mF capacitors on the back of the board. Solder the input and output conductors as shown in the pictures.

ATTENTION!

If you use a "Super-Duper" power supply with separate transformers for the input stages and driver (recommended), do not forget to cut the conductors into printed circuit board between + and Aux+, as well as - and Aux-

Connecting input connectors (unbalanced and balanced, respectively)

Connecting additional modules to the main board

Settings

Connect a multimeter (mV) between two test points on the board, see page 10.

Apply power to the amplifier, DO NOT connect the load yet.

Set the bias adjustment trimmer resistor (501) to 10 mV if you use an amplifier with an 8 Ohm load or 20 mV at 4 Ohms.

Connect a multimeter to the output terminals of the amplifier. Set the DC component adjustment trimmer resistor (103) as close to zero as possible. Deviations of +/- 50 mV are within tolerance when using any speakers.

Check the bias voltage again; it may need to be adjusted. The parameter deviation +/- 20% of the value is within the tolerance.

Repeat the procedures for the other channel. If voltages differ from those listed, please contact LC Audio before proceeding.

Connect your speakers to your amplifier and start playing! You must understand that it takes 1-2 weeks of running in the amplifier to enter operating mode.

Using DC output voltage protection

The Millennium has built-in DC output voltage protection, which you can use at your discretion. You can disable it or exclude it from the scheme altogether if you wish. Some recommendations on this matter:

Some experts are inclined to believe that the protection circuit affects the transmission of low frequencies. And in some cases they are right. The bass becomes softer and more diffuse. This happens because the protection in some amplifiers operates at input filter cutoff frequencies much higher than necessary, say 10-20Hz.

Millennium protection, thanks to our efforts, does not affect the bass section, because the filter cutoff frequency is below 0.5 Hz and a second order filter is installed instead of the usual first order for such cases. This means that the filter cutoff characteristic is steeper, and there is practically no effect on the audio signal (at 20 Hz the filter effect is close to zero)

Filter capacitors C12 and C14 are made in plastic cases and with non-magnetic leads, so that if the entire frequency range of the signal passes through them, they will withstand any, the most demanding audio test. However, a signal above 0.5 Hz does not pass through them.

It is necessary to use a protection system if you use electrostatic speakers, since their DC resistance is close to zero.

You may NOT need to use the protection system if you are using conventional dynamic systems, as some will allow up to 200mV of constant input voltage without damage.

*The name of the topic on the forum must correspond to the form: Article title [article discussion]

It is no secret that knowledge (in the broad sense) is a subjective image of reality. In a narrower sense, knowledge is interpreted as the possession of a certain objective(verified) information that allows you to solve a specific problem.
How objective your image of reality?
Try to analyze how much of your knowledge has been gained true by, i.e. either from your direct experience or as a result of your thinking based on fundamental truths and scientifically based concepts.
This will be the immutable thing that you can rely on when choosing equipment. The remaining approximately 80-99% of all other people’s para-quasi-anti-false-pseudo-as-if knowledge obtained from fabricated articles, abundantly supplied with stunningly beautiful pictures, six-figure price tags and extremely subjective utterances of experts - singles, I suggest you immediately forget.
But remember forever that Scientific explanations are aimed at consciousness. And advertising of all sorts of expensive audiophile things affects the subconscious. It works much more effectively; it is difficult for a person to go against his faith. In general, people, take care of your head!
In fact, almost everything we think his knowledge is gleaned from what came to hand or directly into the ears from the ether. From a young age, in the most primitive way, we become victims of marketing, the flock of professional and well-paid “gurus”. We were told a lot about the intricacies of the sound of this or that cable, about the various influences of interference from the network, about errors when reading laser disks, jitter......about a great variety of processes that must influence the sound.

We now know exactly what should influence! But what are these influences? in numerical terms, and most importantly, can we hear it?! Somehow we were not informed about this.
Let me remind you that influences with similar results add up as the root of the sum of squares. 5% and 1% will not give 6%, but only 5.099%. In other words, when analyzing any influences, you need to know at least order their smallness. Otherwise, we are simply doomed to be Don Quixotes! Horror stories and windmills Adepts Secret Knowledge we came up with a lot...

I am not against esotericism and even some superstitions, because (like all of us in this world) I do not have a comprehensive picture! On the contrary, I try to find a rational grain in everything; however, there are some things I know very well.

So, Horror Stories, sorry, our typical misconceptions

Fallacy of Delusions, №000
About the “deadness” and “boringness” of uncolored sound
There is a common belief that precision equipment quickly becomes boring with its monotonous and idealized sound.
This would certainly be the case if recording studios always produced equally “sterile” and “standard” sound. Certainly, there is no standard sound! All musicians, without exception, strive to give the sound “their own,” preferably an easily recognizable style and coloring; many of them use only their favorite gadgets, worn to holes, the position of the knobs on which is kept in the strictest confidence and is not shown even to their wives! Sound engineers are not far behind them, because no one wants to be an inconspicuous robot.
But alas, there are always those who want to claim that all the efforts of the above people are a waste of time without their wonderful “warm” sound! It is not clear why they decided that the sound was initially “cold”.
Really, you shouldn’t exchange the great variety and individuality of possible sounds for a single sound, even if it’s pleasant to the ear!

Misconception #00
About the "flaws" of sound engineering
It is often written that a high resolution equipment allows you to hear a lot of what you hear not worth it, for example, flaws in sound engineering or the creaking of chairs in a concert hall; and that instead of music it turns out to be an anatomy lesson.
As they say, if you are afraid of wolves, do not go into the forest... From my own experience, I can say that I am not very pleased to hear the shortcomings of a recording, but not to hear its advantages is doubly unpleasant!!!
Advantages but very different things happen, for example, in some moments I really enjoy the strong distortions and other features from the same Alana Parsonsa, although some would call them disgusting. And his remastered 24-bit recordings are actually something, these features form a wonderful sound canvas and begin to live their own lives. And it is especially important that the chips reach your ears “as is”, because the colored ones also in yours equipment they have a chance to become just garbage.
What sounds like garbage on low-quality equipment often turns out to be very lively, stylish and unusual sound events. And it is useless to argue whether these are really flaws or whether they were specially written this way for beauty.
Well, if we get tired of all this, we can always listen to MP3 bitrate 64 or net radio, we certainly won’t hear any mistakes from the sound engineer, everything is clear, we can distinguish zero from one!

Misconception #3.1
I repeat, there are no amplifiers without feedback at all; for example, in the emitter (source, cathode) follower circuit, in which 99.5% of all output stages are assembled, there is 100% local current feedback. Simply put, local feedback is an integral property of any amplifier stage, and talking about its harmfulness is simply stupid.

It's time to figure out how the general OS differs from the local one.
1. In both cases, part of the voltage (current) from the output of the amplifier is supplied in antiphase to its input.

2. In both cases, similar circuit solutions are used, usually the only difference is in the resistor values, which determine the depth of the local OS.

3. The local OS linearizes the amplification stage, but only up to a certain limit, about 0.05 – 0.2% total harmonic distortion. Limitations are imposed physical properties active elements. General environmental protection is free from this fundamental limitation.

4. The phase shift in a circuit without OOOS is completely harmless, since it cannot exceed 90 degrees for each stage, and the stability condition is satisfied automatically. In a circuit with an OOOS, consisting of several stages, this phase shift “accumulates”, and this is the only limitation on the depth of the OOOS. .

And, if you believe the esotericists, the sound is “killed” only by the general operating system, but not by the local one, which makes it possible to localize the problem precisely in the phase shift.
It is interesting that the phase shift in an amplifier is a virtual concept in a sense and for audio frequencies is in no way related to the delay in signal propagation in time, from which In fact The quality of the LLC's work depends very much. Latency equivalent to a 90 degree phase shift at 20kHz – approx. 12 µsec, and no, not even the slowest amplifier has such a delay. For comparison, in ES6.2 the delay from input to output is 60 nsec, i.e. 200 times less. Accordingly, the general environmental protection system in it works in exactly the same way as any local one.

So, the general OOS does not differ in any fundamental way from the local one, with the exception of the number of cascades covered, and the phase shift, which “accumulates”. The difference completely disappears, if you build an amplifier so that the phase shift from input to output in the audio frequency band is small.

But let's return to the quality of amplifiers without OOS.
With input stage everything is fine, the nonlinearities it introduces are small, since the amplitude of the input and output signals is small.
With voltage amplification stage everything is not so great anymore, its gain is usually quite high, and the output amplitude is comparable to the supply voltage, and nonlinear capacitances and the nonlinear dependence of gain and output resistance on voltage are fully affected. The distortions introduced by this cascade are 0.05 – 0.5%, and contrary to popular belief, they do not depend very much on the amplifier architecture.
Fully (supposedly) balanced amplifiers perform almost as well as any other.This happens for the reason that the main contribution is made by only two transistors (in the diagram below Q4 and Q7), but in good amplifiers they Always two, regardless of whether the amplifier is "balanced" or not. In addition, completely complementary transistors simply do not exist; the capacitance and curvature of transistors of different structures differ significantly due to technological reasons.
The figure below shows the results of modeling a “symmetrical” and once sensational amplifier without OOS “ The end Millennium » , the diagram is taken from here, simple and beautiful.

From the simulation results it is easy to see that the distortion of the End Millennium amplifier without load ( and even without an output stage!!!) approximately 0.07% THD and 0.1% IMD. As a trick, a cascade, even a carefully tuned one, will add (as will be shown below) about the same amount, but the trick is that as a result of multiplying the distortion spectra, the final spectrum will contain a lot of harmonics and intermodulations of a high order. Apparently, this same garbage is declared to be of “unique” quality.
It is unclear what 0.0017% THD the authors claimed. Quite a bold statement even for a good amplifier with OOOS. The error is almost 50 times, however! But, thanks to the authors, now we know which numbers they consider “reference”.

Output stage. The best and carefully built(including in class "A") has an output impedance of 0.05 - 0.2 Ohm and distortion on a large signal of the order of 0.05 - 0.2%, and up to 0.4% on a medium-small signal (). The resulting distortion (especially on a large and complex signal, where it will vary chaotically with frequency, since the load impedance is not constant and is not very similar to a resistor) can be up to 0.5%. This “accuracy” can be checked by any Chinese tester!

So, what can you count on when you become the owner of an amplifier with the proud inscription “amplifier without negative feedback”?

Misconception #4
About the need for long-term “warm-up” of equipment
I don’t see any practical point in long-term (more than half an hour) warming up devices that do not contain moving parts or parts with a very high heat capacity. Well, I don’t believe in the possibility of hyperfine states of matter in an ordinary transistor or capacitor!
The human hearing aid is another matter! It can and should be warmed up over the years, especially when it begins to hear new synthetic sounds. It takes time to convince yourself that something is good.
In addition, if a product “warms up” for a week, that is, there is a rapid drift of parameters, then in a month it can “grow old”, and in two months it can die.

Misconception #5
About the “unimportance” of harmonic distortion.
Harmonic distortion has always been considered one of the main characteristics of the sound amplification path. But, like everything in this world, their correct understanding has its own subtleties. One subtlety - with numerically equal Kg, amplifiers can sound completely different due to the different spectral composition of the harmonics. The second subtlety is the unevenness of Kg at different frequencies. Below shows that It is incorrect to talk about distortions by considering only harmonic distortions, without regard to intermodulation ones.
The fact is that the same nonlinearities in the amplifier path that give rise to harmonics absolutely inevitably give rise to intermodulation. And this is not a subject for discussion, it is a mathematically proven fact. In fact, harmonic distortion is just special case intermodulation, when one of the test frequencies is missing. Intermodulation of high-frequency components also affects mid-frequencies, the zone of greatest hearing sensitivity, and Not masked by HF components. The hearing threshold at mid frequencies is around 0 dB, and it is important to keep intermodulation below this threshold. First-order intermodulations, at best, are equal to harmonics in amplitude, hence the clear requirement: the level of harmonic distortion at high frequencies of the entire path (this is especially difficult to achieve in a PA) should not exceed the audibility threshold at medium frequencies. Thus, for a sound pressure of, for example, 96 dB, the level of harmonic distortion at HF should not be more than 0.0016%. An amplifier with such low HF distortion demonstrates an unusually subtle, airy, weightless sound.
This, as they say, is an argument Behind little distortion.
Argument Against the fact that supposedly the distortions are quieter than the background noise of the room and are not audible.
The assumption that distortion below the noise level will not be noticed is, in my opinion, an unforgivable and incorrect simplification. For example, we can perfectly hear the quiet singing of birds outside the window, but if we take a microphone, record it, weigh it using an equalizer along the hearing sensitivity curve and try to find the signal peaks corresponding to the singing in the resulting noise picture of the room that is adequate from the point of view of hearing, then we won't see anything! This happened because the measured level of the noise track carries information about the integral value of the signal, roughly speaking it is the root of the sum of the squares of all frequencies, each of which is significantly smaller in amplitude. We would easily see it on a spectrogram, because birdsong is a narrow-band signal that exceeds noise in the observed frequency interval.
There are at least two more features of human hearing, which should not be ignored and “simplified”, and which helped us to hear the birds singing against the background of the rumbling of the refrigerator and the snoring of our flatmate. This is direction selectivity and the ability to “accumulate” information about a repeating signal that is sufficiently long in time. According to some researchers ( Stereophony . - Kovalgin Yu.A.), the first of them is 12-15 dB (!), information on the second, unfortunately, could not be found. I don’t want to overestimate it, just as I don’t want to ignore it, so let’s take something average, for example 6 dB.
The total is approximately 20 dB.
As a result, if we listen to music in a quiet room (20-30 dBA), we arrive at approximately the same numbers: intermodulation and harmonic distortion of the amplification path throughout the entire frequency band should be less than the audibility threshold, about 0.003% and 0.002%, respectively. Naturally, it is preferable to have a reserve, just to be sure.

DIY microphone amplifiers.

Amplifier for computer microphone with phantom power.

I installed a program like Skype on my computer. But here’s one problem: you need to keep the microphone close to your mouth so that the interlocutor can hear you well. I decided that the microphone sensitivity was not enough. And I decided to make an amplifier amplifier.

An Internet search yielded dozens of amplifier circuits. But they all required a separate power source. I wanted to make an amplifier without an additional source, with power from the sound card itself. So that there is no need to change batteries or pull additional wires.
Before you fight the enemy, you need to know him by sight. Therefore, I dug up information on the Internet about the microphone design: https://oldoctober.com/ru/microphone. The article tells how to make a computer microphone with your own hands. At the same time, I borrowed the idea itself: there is no need to break a ready-made device for my experiments if you can do it yourself. A brief retelling of the article comes down to the fact that a computer microphone is an electret capsule. An electret capsule is, from an electrical point of view, field-effect transistor open source. This transistor is powered from the sound card through a resistor, which is also a signal current-to-voltage converter. Two clarifications to the article. Firstly, there is no resistor in the capsule in the drain circuit, I saw it myself when I took it apart. Secondly, the connection between the resistor and capacitor is made in the cable, not in the sound card. That is, one pin is used to power the microphone, and the second is used to receive a signal. That is, it turns out something like this:

Here the left part of the figure is an electret capsule (microphone), the right is a computer sound card.
Many sources write that the microphone is powered from a voltage of 5V. This is not true. In my sound card this voltage was 2.65V. When the microphone power output was shorted to ground, the current was about 1.5 mA. That is, the resistor has a resistance of about 1.7 kOhm. It was from such a source that the amplifier was required to be powered.
As a result of experiments with microcap, this scheme was born.

The capsule is powered through resistors R1 and R2. To prevent negative feedback at signal frequencies, capacitor C1 is used. The capsule is supplied with a supply voltage equal to the voltage drop across p-n junction. The signal from the capsule is isolated at resistor R1 and fed to the base of transistor VT1 for amplification. The transistor is connected according to a common emitter circuit with a load on resistors R2 and a resistor in the sound card. Negative DC feedback through R1, R2 ensures a relatively constant current through the transistor.

The entire structure was assembled by surface mounting directly on the microphone capsule. Compared to a microphone without an amplifier, the signal increased approximately 10 times (22 dB).

The entire structure was first wrapped with paper for insulation, and then with foil for shielding. The foil has contact with the capsule body.

Single-wire powered microphone amplifier.

A microphone with a preamplifier located in the housing requires power wires to be connected to the device (in addition to the shielded signal wire). From a constructive point of view, this is not very convenient. The number of connecting wires can be reduced by supplying the supply voltage through the same wire through which the signal is transmitted, i.e., the center conductor of the cable. It is this method of supplying power that is used in the amplifier we bring to the attention of readers. His circuit diagram shown in the figure.

The amplifier is designed to operate from any type of electret microphone (for example, MKE-3). Power is supplied to the microphone through resistor R1. The sound signal from the microphone is supplied to the base of the transistor VT1 through the isolation capacitor C1. The required bias at the base of this transistor (about 0.5 V) is set by the voltage divider R2R3. The amplified audio frequency voltage is released at the load resistor R5 and then goes to the base of the transistor VT2, which is part of a composite emitter follower assembled on transistors VT2 and VT3. The emitter of the latter is connected to the upper contact of the XP1 connector (amplifier output), to which is connected the central conductor of the connecting shielded cable, the braid of which is connected to the common wire. Note that the presence of an emitter follower at the output of the preamplifier significantly reduces the level of interference to the microphone input.

Near the input connector of the device to which the microphone is connected, two more parts are mounted: a load resistor R6, through which power is supplied, and a separating capacitor SZ, which serves to separate the sound signal from the DC component of the supply voltage.
The circuit design used in this amplifier provides automatic installation and stabilization of its operating mode. Let's look at how this happens. After turning on the power, the voltage at the upper terminal of the XP1 connector increases to approximately 6 V. At the same time, the voltage at the base of the transistor VT1 reaches its opening threshold of 0.5 V and current begins to flow through the transistor. The voltage drop that occurs in this case across resistor R5 causes the transistor of the composite emitter follower to open. As a result, the total current of the amplifier increases, and along with it the voltage drop across resistor R6 increases, after which the mode stabilizes.

Since the current gain of the composite emitter follower (it is equal to the product of the current gain of transistors VT2 and VT3) can reach several thousand, mode stabilization is very strict. The amplifier as a whole operates like a zener diode, fixing the output voltage at 6 V regardless of the supply voltage. However, when using a power source with a different voltage, it is necessary to select the resistors of the divider R2R3 so that the voltage at the upper contact of the XP1 connector is equal to half the supply voltage. It is curious that the mode practically cannot be changed by adjusting the resistance of the load resistor R5. The voltage drop across it is always equal to the total opening voltage of the transistors of the composite emitter follower (about 1 V), and changes in its resistance only lead to a change in the current through transistor VT1. The same applies to resistor R6.

More more interesting work amplifier in AC amplification mode. The audio frequency voltage from the lower terminal of resistor R5 is transmitted by the emitter follower with very little attenuation to the upper terminal - the output of the amplifier. In this case, the current through the resistor is constant and is almost not subject to fluctuations at audio frequency. In other words, the only amplifier stage is loaded onto the current generator, i.e. to very high resistance. The input impedance of the repeater is also very high, and as a result the gain is very large. During a quiet conversation in front of a microphone, the amplitude of the output voltage can reach several volts. The R4C2 chain does not allow the alternating component of the audio frequency signal to pass to the power circuit of the microphone and voltage divider.

A single-stage amplifier is not at all prone to self-excitation, which is why the arrangement of parts on the board special significance does not, it is advisable to only place the input and output at different ends of the board.

The setup comes down to selecting resistors of the divider R2R3 until half the supply voltage is obtained at the output. It is also useful to select resistor R1, focusing on the best sound of the signal recorded from the microphone. If the input impedance of the radio device with which this amplifier is used is less than 100 kOhm, the capacitance of the capacitor SZ should be increased accordingly.

Connecting a dynamic microphone to the microphone input of a computer sound card.

The microphone input of the sound card is intended for connecting an electret microphone. The assignment of the microphone input connector pins is shown in Fig. 1. The sound signal is supplied to the sound card input through the TIP contact. Power for the electret microphone is supplied through resistor R to the RING pin. The TIP and RING pins are connected together in the microphone cable.

Rice. 1

Almost all multimedia microphones costing $2-4 are only suitable for speech recognition, telephony, etc. Although these microphones usually have high sensitivity, they have high level nonlinear distortions, insufficient overload capacity, as well as a circular radiation pattern (that is, they perceive signals equally well from any side). Therefore, to record vocals at home, it is necessary to use a highly directional dynamic microphone, which allows you to minimize extraneous noise from the system unit fan and other sources.

A dynamic microphone can be connected directly to the microphone input of the sound card. The signal wire of the microphone cable must be soldered to the TIP pin, the shield to the GND pin, and the RING pin must be left free. If the microphone has two signal contacts - HOT and COLD, then connect the HOT contact to the TIP contact, and connect the COLD contact to GND. Since the sensitivity of a dynamic microphone is low compared to an electret microphone, a sufficient recording level is obtained only when the microphone is positioned at a distance of 3-5 centimeters from the performer’s lips. This is not always acceptable, since some types of microphones will spit despite the built-in wind protection. Such microphones must be placed further from the performer, and to obtain a sufficient recording level, use a preamplifier. The circuit of a simple preamplifier powered from a microphone input connector is shown in Fig. 2.

Rice. 2

This circuit works well for me at the following ratings: R1, R3 - 100 kOhm, R2 - 470 kOhm, C1, C2 - 47 uF, VT1 - kt3102am (can be replaced with kt368, kt312, kt315).
The circuit is based on a classic transistor cascade with a common emitter. The load of the cascade is the resistor R of the sound card (Fig. 1). The gain depends on the parameters of transistor VT1, the value of feedback resistor R2 and the value of resistor R of the sound card. Capacitor C1 is required for DC decoupling. Resistor R1 is used to eliminate clicks when connecting a microphone on the fly; if desired, you can exclude it.

Upon closer examination, it turned out that there was a constant voltage of about 2 V at the TIP contact of the microphone input of my SB LIVE 5.1. It was not possible to investigate the reason, and whether this is typical only for my copy of the sound card or for all. But it is absolutely certain that the performance of the circuit practically does not change when elements C2 and R3 are excluded.

The advantage of this scheme is its simplicity. The disadvantages include large nonlinear distortions - about 1% (1 kHz) at 1 mV at the input. Nonlinear distortion can be reduced to 0.1% using an additional 100 Ohm resistor connected between the emitter of transistor VT1 and the GND bus, while the gain is reduced from 40 dB to 30 dB. The changes are shown in Fig. 3.

Rice. 3

Higher parameters can be obtained using an external, self-powered microphone amplifier connected to the line input of the sound card. For example - assembled according to a circuit with a symmetrical input.

DIY microphone amplifier.

Probably, many of you have had the need to record sound on a computer, for example, when dubbing videos or creating clips. The use of Chinese inexpensive consumer goods is absolutely undesirable, firstly, due to the rather low sensitivity, and secondly, the quality of sound recording
it turns out *dirty*, sometimes even your own voice becomes unrecognizable.
High frequencies have a significant and unjustified rollover, and their durability leaves much to be desired.
A high-quality microphone, alas, is beyond our means!

But there is a way out! Many people have old, Soviet dynamic microphones, for example MD-52 or similar ones. And even in their absence, these copies can be bought for *mere pennies*. Do not try to connect such microphones directly to the sound card directly - the AF voltage at the output is too low. Therefore, we will use the simplest microphone amplifier, based on the widely used K538UN3 microcircuit, its cost is less than 50 rubles. But we used an old microcircuit soldered from an ancient cassette recorder. Directly, the microcircuit itself is connected according to a standard, common switching circuit, with a maximum gain. The amplifier is powered directly from the computer, the supply voltage is 12 V, although operation remains unchanged at - 5 V, in this case, power can be taken from the USB connector.

Microphone amplifier. Scheme.

Electrolytic capacitors - any, for a voltage of 16V. The capacitance value of the capacitors can be changed within small limits. The device can be assembled using a simple, hinged installation.

The amplifier does not require any adjustment and does not require shielding. But, the use of shielded cables is desirable and not too long. Sample tests showed relatively low level own noise, fairly high sensitivity and very decent sound quality, even on built-in computer sound cards such as AC97. Dynamic range is about 40 dB. To record sound on a computer, we used the Sound Forge program.

Well, and a few more diagrams for the articles in addition.

Clean sound to you!!!

For pop orchestras, school radio centers or intercoms, you often need preamplifier to a low-impedance microphone or a dynamic head used in the same role. Circuits of such amplifiers are offered by the magazine "Funkamater" (GDR).

The first, the simplest, is used when the microphone is located at a considerable distance from the main amplifier. The 7.5-12 V supply voltage is supplied to the preamplifier via an “audio” cable with a grounded braid. Transistors (V1 and V2) provide high signal amplification. Capacitor C2 eliminates self-excitation. The operating mode is set using trimming resistor R3 so that the collector V2 has “half” supply voltage. Current consumption = 1.5 mA.

The second amplifier is designed for collaboration with high quality equipment. With an increase in resistance R5 = 100 kΩ, the device gain is maximum (51 dB). Sensitivity 3-8 mV, optimal microphone impedance = 200 ohms. At the top point of R2 the voltage is = + 6 V, and at the collector V1 the voltage is approximately + 2 V.

Both amplifiers are assembled from small parts and placed in tin cases the size of a matchbox and grounded. The devices use low-power silicon transistors: V1 low-noise, for example KT312B, V2 - KT306, KT315, KT342 with any letter index. Magazine "M-K" No. 2, 1985

Non-standard microphone activation.

Placing a microphone amplifier in close proximity to the microphone reduces the shielding requirements of interconnecting wires and improves the signal-to-background ratio. However, this raises a new problem associated with powering the microphone; the built-in battery requires frequent replacement, and using an additional power cord is not always convenient.

The figure shows a diagram of a two-stage microphone amplifier whose power is supplied via a signal wire. In this case, you only need to add one resistor R4 to the main amplifier, which serves as a load for the microphone amplifier and an isolation capacitor C2.

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It is inexpensive, costs about 120 rubles.

And here is his diagram:

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Fig. 4 . Electrical circuit of a microphone amplifier.

More different microamplifiers on microcircuits

These amplifiers are used to amplify signals of low magnitude (0.2-2 mV). The input impedance of the microphone amplifier, which provides the maximum signal-to-noise ratio, is selected to be 3 times the internal impedance.

The circuit implementation of a microphone amplifier is quite simple when using an operational amplifier. The operational amplifier should be selected based on the minimum noise level applied to the input. Of the domestic operational amplifiers, the most suitable are the KM551UD2A (Uin noise = 1 μV) or K157UD2 (Uin noise = 1.6 μV). Among foreign operational amplifiers, we can recommend NE5532.

Input voltage 1 mV,
Nominal output voltage 100 mV,
Signal to noise ratio = 56 dB,
Operating frequency range Hz,
Harmonic distortion 0.05%

The operational amplifier is connected in an inverting amplifier circuit. The gain is determined by the ratio of resistors R1 / R2 and is equal to 100. When replacing the operational amplifier K157UD2 with KM551UD2A, the signal to noise ratio will increase to 60 dB.

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Figure 3 shows a diagram of a microphone amplifier with a balanced input, in which the functions of a transformer are performed by a differential amplifier based on the operational amplifier DA1.

A summator of two signals is assembled on DA2. The higher the degree of matching of resistors RЗ and R4, R6 and R7, R8 and R9, R10 and R12, R11 and R13, the higher the degree of matching of resistors RЗ and R4, R6 and R7, R8 and R9.

The microphone amplifier has the following parameters:
Nominal input voltage = 2 mV,
Nominal output voltage = 100 mV,
Signal-to-noise ratio 60 dB,
Harmonic distortion 0.5%,
Reproducible frequency range Hz,
Minimum load resistance = 10 kom.

The gain of the microphone amplifier depends on the position of switch S1.

When the switch is open, K = 50, when closed = 100.

Microphone preamp, also known as a pre-amplifier or amplifier for a microphone, is a type of amplifier whose purpose is to amplify a weak signal to a linear level (about 0.5-1.5 volts), that is, to an acceptable value at which conventional audio power amplifiers operate .

The input source of acoustic signals for a preamplifier is usually vinyl record pickups, microphones, and pickups of various musical instruments. Below are three circuits of microphone amplifiers on transistors, as well as a variant of a microphone amplifier on the 4558 chip. All of them can be easily assembled with your own hands.

Circuit of a simple microphone preamplifier using one transistor

This microphone preamplifier circuit works with both dynamic and electret microphones.

Dynamic microphones are similar in design to loudspeakers. The acoustic wave affects the membrane and the acoustic coil attached to it. At the moment the membrane oscillates, in a coil under the influence magnetic field permanent magnet, an electric current is generated.

The operation of electret microphones is based on the ability certain types materials with increased dielectric constant (electrets) change the surface charge under the influence of an acoustic wave. This type of microphone differs from dynamic microphones in its high input impedance.

When using an electret microphone, to bias the voltage on the microphone, it is necessary to set the resistance R1

Since this microphone amplifier circuit is for a dynamic microphone, when using an electrodynamic microphone, its resistance should be in the range from 200 to 600 Ohms. In this case, C1 must be set to 10 microfarads. If this happens electrolytic capacitor, then its positive terminal must be connected towards the transistor.

Power is supplied from the crown battery or from a stabilized power source. Although it is better to use a battery to eliminate noise. can be replaced with a domestic one. Electrolytic capacitors for a voltage of 16 volts. To prevent interference, connect the preamplifier to the signal source and to the amplifier input using a shielded wire. If further powerful sound amplification is needed, then you can assemble an amplifier on a microcircuit.

Microphone preamplifier with 2 transistors

The structure of any preamplifier greatly affects its noise characteristics. If we take into account the fact that the high-quality radio components used in the preamplifier circuit still lead to distortion (noise) to one degree or another, then it is obvious that the only way to get a more or less high-quality microphone amplifier is to reduce the number of radio components in the circuit. An example is the following two-stage preliminary circuit.

With this option, the number of decoupling capacitors is minimized, since the transistors are connected in a circuit with a common emitter. There is also a direct connection between the cascades. To stabilize the operating mode of the circuit when the external temperature and supply voltage change, a direct current feedback loop has been added to the circuit.

Preamplifier for electret microphone with three transistors

This is another option. The peculiarity of this microphone amplifier circuit is that power is supplied to the preamplifier circuit through the same conductor (phantom power) through which the input signal travels.

This microphone preamplifier is designed to work together with, for example, MKE-3. The supply voltage to the microphone goes through resistance R1. The audio signal from the microphone output is supplied to the VT1 base through capacitor C1. , consisting of resistances R2, R3, creates the necessary bias at the base of VT1 (approximately 0.6 V). The amplified signal from resistor R5, acting as a load, goes to the base of VT2 which is part of the emitter follower on VT2 and VT3.

Near the output connector, two additional elements are installed: load resistor R6, through which power is supplied, and separating capacitor SZ, which separates the output audio signal from the supply voltage.

Pre-microphone amplifier based on 4558 chip

The 4558 operational amplifier is manufactured by ROHM. It is characterized as a low power and low noise amplifier. This microcircuit is used in a microphone amplifier, audio amplifiers, active filters, and voltage-controlled generators. The 4558 chip has internal phase compensation, increased input voltage threshold, high gain and low noise. This op amp also has short circuit protection.

(140.5 Kb, downloads: 2,161)

This a good option for building a microphone preamplifier on a microcircuit. The microphone preamplifier circuit is characterized by high amplification quality, simplicity and does not require much wiring. This dynamic microphone amplifier also works well with electret microphones.

With error-free assembly, the circuit does not require configuration and starts working immediately. The highest current consumption is 9 mA, and at rest the current consumption is around 3 mA.