Electronic heart diagram. Electronic heart. Immediately soldered handkerchief

An electronic LED heart on a microcontroller can be a great gift for a girl on Valentine's Day, March 8, or for her birthday, if you know how to solder, of course. It will make a good gift, besides made with your own hands. In order to create such a trinket, we need:

1) ATmega88 microcontroller
2) 22 red SMD LEDs (better to take with a margin)
3) 22 620 ohm SMD resistors (similar)
4) 1 SMD resistor 10kΩ
5) 1 SMD capacitor 0.1uF
6) 2 SMD jumpers
7) Fiberglass
8) Programmer for AVR
9) Photoresist PV-ShchV
10) Soda ash
11) Caustic soda
12) Ferric chloride

We draw a diagram (click on the picture to enlarge). According to this scheme, then we will breed the board. From the schematic editor, then unload the NET-list (list of circuits) and the archive library of the components used.

We split the fee. All components for surface mounting.

We glue the film photoresist on the board, the main thing is to prevent the formation of air bubbles. We wrap the board with paper and pass it 2 times through the laminator, so that the photoresist sticks to the board better.

We print a photomask. We will put the photomask with toner to the board, so we print in a mirror way.

We put the photomask on the board, press it with glass from above. Turn on the UV lamp for 3 minutes. The science says you should use plexiglass, but with glass taken off a bookshelf, it works just fine.

After exposure, remove the top protective film and prepare the developer. To do this, we take ordinary water, passed through a filter or boiled, to reduce its hardness. With ordinary tap water, as a rule, there are problems. We also need soda ash (NaCO3). The concentration of the solution is a teaspoon per 100 ml of water. We show payment. The pattern that was exposed to ultraviolet remains on the board, everything else dissolves.

Developed board ready for etching:

We prepare an etching solution of ferric chloride. To do this, we need, surprisingly, ferric chloride and water (now you can directly from the tap). We breed 1 to 3. We poison the board. We are careful, since ferric chloride is poorly washed from hands and furniture and is extremely difficult to wash from clothes.

We make a solution for removing photoresist. We take water (again from the tap) and caustic soda, the concentration is already familiar to us - a teaspoon per 100 ml of water. We remove the photoresist, do not forget to use rubber gloves, since the solution is quite caustic, after which we wash the board in water, tin and solder the parts according to the drawing.

And we start coding. For programming and firmware, the WinAVR package is enough for us. It is not a sin to spend the whole evening and night on programming - a very interesting toy, you can pervert as much as your imagination is enough. We sat up until 4 am. After all the above procedures, batteries and a reed switch were soldered to the board, then the board was placed in a box with a magnet on the lid, which, when the box is closed, opens the reed switch with a constant magnetic field.

And now a video illustrating the operation of the LED heart:

I do not pretend to be a novelty of the idea or execution, but it may be useful to someone. Actually, this device was made by my wife for her wedding anniversary, although you can quite easily reorient it for other holidays.

The electrical circuit diagram was found on the Internet and was safely lost there. Therefore, it will not be. Yes, in principle, it by itself is not needed. this device is a logical continuation of the first attempts to light LEDs. The idea itself was to please my wife and prove to her that it was not in vain that I was sitting in the evenings with a soldering iron.

Printing in Sprint Layout

As you can see from the printed circuit board, there is nothing special about it:

• Atmega8-tqfp32
• Rezuk for 100k
• Conder at 0.1uF
• 22 smd leds
• 22 smd summaries

Regarding LEDs and resistors, perhaps choose them so as not to exceed the threshold voltage value of + 5V. I took super bright ones at 3V, the current was 20 mA, respectively, the cutters were 120 ohms each.
In order not to think too much, there are a bunch of online calculators.

There is no connector for ISP programming in the usual sense. Stupid wiring. Yes, by the way, everything is signed there for convenience. The process of “LUTing” the board and explaining the technology, I think it’s not advisable to give here, because. who knows and knows will understand, and who does not Google to help him.

Immediately soldered handkerchief.
And of course, as in the Russian joke about the plane, “And now it’s rough to process with a file.” As for the code, there will only be a firmware file without source code, because this is a standard footswitch.

Yes, I almost forgot the video. For the quality I apologize for what was at hand.

Perhaps that's all. Project file:
About Fuse, we leave the factory. Happy repetition everyone.

The contour of the heart is formed by four garlands of four LEDs in each and one blinking LED, which plays the role of a "diamond" (Fig. 1). The LED garlands are controlled by a field effect transistor VT1, which, in turn, is controlled by a flashing LED HL1, it happens like this. The first (HL2, HL6, HL10, HL14) and the second (HL3, HL7, HL11, HL15) strings of LEDs are connected in parallel and connected to the battery through a current-limiting resistor R4 and a channel of the field-effect transistor VT1. Two other garlands - the third (HL4, HL8, HL12, HL16) and the fourth (HL5, HL9, HL13, HL17) - are connected to the battery through the same resistor and an additional VDI DIODE.

When the field effect transistor VT1 is closed, the third and fourth garlands light up. When the field effect transistor is open, only the first and second will shine, and the third and fourth will go out. This is explained by the fact that the drain-source voltage of an open field-effect transistor (tens of mV) is significantly less than the voltage on the open diode VD1 (0.6..0.7 V), so the voltage on the first and second garlands will not be enough for the third and fourth to glow. The blinking LED HL1 is connected to the power supply battery through a resistive circuit R1-R3, and when it is off, a small current flows through it, so the voltage at the gate of the field-effect transistor VT1 is not enough to open it. When the LED HL1 (“diamond”) flashes, the current through it will increase sharply, the voltage at the gate of the field-effect transistor will increase and it will open. Therefore, in time with the blinking NI LED, the flash frequency of which is 1 ... 2 Hz, the first and second garlands light up, and the third and fourth go out. All LEDs are placed on the board in such a way that when they are switched, the effect of a running fire is realized.

All parts, except for the battery, are mounted on a printed circuit board made of one-sided foil fiberglass with a thickness of 1 ... 1.5 mm, the drawing of which is shown in Fig. 2. Resistors C2-23 are used, we will replace the flashing LED L-56BID with L-5013LRD-B. If you use a two-color flashing LED, for example, L-5013SBW-B or BK5RB6SSC 5mm, then red and blue flashes will alternate. Instead of AL307BM LEDs, you can use L-5013SRT, KIPD21A-K or similar, necessarily red glow.

The appearance of the mounted board of the heart is shown in fig. 3. The device does not need adjustment.

At the same time, it does not matter at all what it is - an expensive "iron", or - in general, "virtual", in a computer. The main thing is that he is. You can't do without a mixing console - neither in the studio, nor at the concert venue, nor in the theater - anywhere.

In many ways, the mixing console is similar to Lake Baikal, forgive Greenpeace for such a comparison! Just like Baikal, many “rivers” and “brooks” - sound signals - from microphones, electronic musical instruments, reverberators, and so on, flow into it, and only one “river” flows out - the total sound signal.

Sound signals entering the remote control are amplified, attenuated, processed by various equalizers, compressors and other things (sugar and salt - to taste!), Mixed - and you're done! Hmmm ... That - the lake, then - the kitchen. So sleep is not long! But, we didn't invent it.

One of the English names for a mixing console is Mixing Board, which means “mixing board”. This name was born a long time ago, at the dawn of the development and formation of radio electronics, when the consoles did not yet have all the modern delights - no equalizers, no subgroups, not even the slightest automation - nothing! Longing, in a word... On the other hand, a modern mixing console is often such a complex device that even the most sophisticated professional will not always be able to figure it out right away.

There are a great many consoles - concert, studio, theater, etc. etc. However, despite their great diversity, there are many common features in the designs of all consoles. Any console contains, at a minimum, input cells and a master section. But this is not always enough, especially when working with a large number of signal sources. Therefore, as the working conditions become more complicated, many additional devices were invented - such as subgroups, “auxes” (AUX), breaks (INSERT), for multi-channel recording - special cells (IN-LINE), and much more.

An example structure of a mixing console is shown in the figure below.

Input cells Subgroups Master section

Input cells

Input cells, as the name implies, receive input signals from microphones and other sources. Here, preliminary amplification of signals is carried out, their processing - frequency, dynamic, as well as some other types, and distribution to further devices. In the very general view an example input cell structure is shown in the following figure:

1.Entrance section.
2.Processing block.
3. Signal distribution block.

The signal from the source is fed to the input section, where the signal is selected, its normalization is brought to the level necessary for the normal functioning of further circuits, and preliminary filtering.

The input section typically has the following elements: a MIC/LINE input selector, GAIN control(s), a PHASE shifter (sometimes just an icon), and filter(s). Sometimes there is a PAD button for stepwise attenuation of the input signal of the microphone input - usually by 20 or 30dB. The signal level is adjusted by the GAIN knob of the input amplifier, and the term amplifier is somewhat arbitrary, since both amplification and attenuation of signals can be carried out here.

In professional equipment, as a rule, there are two separate inputs - a balanced MIC for a microphone and a line LIN - for signals with high levels.

Line input - most often unbalanced, but in a very serious technique - it can also be symmetrical.

One remark must be made here. In relatively cheap equipment, sometimes you can suddenly see, frankly, unexpectedly, a balanced line input. About free cheese - remember? So here, it would be nice to ask - why is it all of a sudden such generosity? If someone believes in the altruism of the manufacturer - forget it! Everything is much simpler - and worse. This is purely a publicity stunt, nothing more. Although the entrance is indeed symmetrical, this is true. But not all...

Remember the famous saying - "Always tell the truth, the whole truth, and nothing but the truth. But never tell the whole truth!" Here is just a similar situation.

Everything is very simple: the signal from this input is first attenuated, sometimes quite strongly, several dozen times, and then fed to the input ... yes, you guessed it right - a microphone amplifier! The task of one action - will the sound improve after such a transformation? Decide for yourself...

A good indicator of this trick is the presence of only one input sensitivity knob - instead of two separate ones, as well as the absence of an input select button.

After pre-amplification, there may be two not entirely obvious devices in the signal circuit - a phase shifter and a filter (s). Strictly speaking, it is more accurate to call the first phase inverter, since nothing in it “rotates”, but the signal phase is simply inverted by 180 degrees, but - apparently, “it’s so beautiful”. It is necessary for phasing microphones, and sometimes for other purposes. The signal can then be applied to filters to limit its bandwidth and remove unwanted components. In expensive (alas!) professional consoles, you can sometimes find a complete set of them, both for cutting low frequencies (LO-CUT) and for cutting high frequencies (HI-CUT), and even with tunable cutoff frequencies! But most often, alas, the simplest “one-button” filter is used, which, as a rule, cuts only low-frequency components below 80 or 100 Hz. This filter is sometimes referred to as the “step noise filter” because it serves primarily to reduce the "stomp" from footsteps transmitted from the supporting structures of the stage to the microphone through its stand.

Further, the signal after the input section is fed to the processing unit. This block includes various tone control circuits (equalizer), as well as inserts (INSERT) for inclusion in the signal path of external devices - compressors, flangers, etc.

These nests are usually paired. One socket - “Send” (“send”, “output”) is used to send a signal to an external device, the other - “Return” (“return”, “input”) to return the processed signal to the cell. In some models of inexpensive consoles, there are also combined jacks, on “stereo jacks”. This saves space on the back of the remote, but is much less convenient. By the way - in good consoles, INSERT jacks are mandatory in all its sections - in cells, and in subgroups, and in the master section.

Of course, strictly speaking, these nests (“breaks” - INSERT) - are not included in any blocks, because “physically” - are located between different nodes of the cell, but it is advisable, when considering the structure of the console, to consider their purpose here, based on their functional role. In expensive professional consoles, there are usually two INSERT jacks - one before the EQ, and one after. What are two for? Well, first of all, more is not less. (Joke!) And secondly, many processing devices “behave” differently, being included in a “clean signal”, or in an already “timed” one. Accordingly, the results obtained will be different.

For example, the property of strong compression is known to “eat up” timbres, as it were. That is, if you strongly “wind up” the timbre of the signal, and then apply it to the compressor, then all your “cheating” can “fall like a death of the brave”. To avoid this, it is better to turn on the compressor before the equalizer. From the same jacks, you can remove individual channel signals for feeding to - for example - a second console (monitor, video, etc.), so that independent tone control can be carried out there.

It is advisable to use the INSERT jacks after the equalizer, for example, to connect devices with a limited dynamic range - a flanger, etc., so as not to “steer” the equalizer along with the useful signal and processing noise. In many cases, it is also useful to apply an already equalized signal to the processing included in the insert - for example, to a noise gate, to an exciter, etc. etc. Of course, all of the above is “not” the ultimate truth. The author is not the Lord God, and not even Bill Gates (according to a well-known anecdote...). These cases are provided as examples only, to demonstrate the need to have two breakpoints in each cell. However, in most inexpensive consoles INSERT - alas! - just one, after the equalizer! Keep this in mind when using it.

The equalizers in the cells are very diverse - from the simplest bass and treble, with “shelf” controls, to the most complex fully parametric four-band ones. The latter, as a rule, on the extreme LF and HF regulators have the ability to switch the “bell / shelf” regulation characteristic. In a parametric equalizer for each band, all parameters are independently set (hence the name - “parametric”): the center frequency of regulation fо, the width of the regulation bandwidth and the amount of rise / cut of the frequency response, and in equalizers of the “shelf” type - only the amount of rise / blockage of the frequency response at the edges of the range, the remaining parameters are determined by its circuitry, and their change by the sound engineer is impossible. Name - corresponds to the type of frequency response. For a regulator of the “bell” type (from the English word BELL - “bell”), the frequency response has a really “bell-shaped” shape, with a maximum depth of regulation at the main frequency of its tuning, and gradually decreasing as it moves away from it. The regulator of the “shelf” type (from the English word SHELF - “shelf”) does not have a pronounced tuning frequency, its frequency response has a maximum depth of regulation at the edges of the sound range, and gradually decreases towards its middle. Sometimes, however, in (and what can you do? Again!) Expensive consoles have the ability to adjust the frequency for the “shelf” control, but this is a completely different adjustment: the frequency changes, HIGHER than which for the LF controller, (or BELOW - for the HF -regulator), the characteristic becomes smoothly falling. Below this frequency - in the first case, and above it - in the second, all frequencies rise or fall in the same way.

So, the signal amplified, corrected - and went to the distribution block. It is this part of the cell that is distinguished by the maximum variety of designs, and often causes the greatest difficulties, although in terms of design it is the simplest part, “a set of buttons and knobs”. With the buttons you choose where the signal will be sent next, and with the knobs (if any) you set the level of this signal.

This part is called “Routing” in the literature and sometimes on the consoles themselves. The signals coming from the cells to subsequent circuits are taken from two points in the circuit: some of the signals are taken before the cell fader (PRE - Fader), and some - after it (POST - Fader).

As a rule, all signals that go further to the main mix and processing are removed after the fader, and those signals that go to the main mix and subgroups are removed after the pan control. The signals taken BEFORE the fader are, as a rule, only those that go to the monitors - stage or studio.

Why exactly? Yes, it's very simple - so that the balance of monitors does not depend in any way on possible change balance in the hall or in the main mix! Once you have built it - and you no longer think, you are doing your main business.

In the most general form, the following controls serve to distribute signals: a panoramic control “PAN”, feed buttons - to the main output (“MIX”), to subgroups (“SUB”, or “GROUP”), to a multi-channel tape recorder - “ODD ” and “EVEN”, (“Even” and “odd”), as a rule - with numbers from “1” to “24”. By the way, at the same time, on the panorama regulator, too, there are inscriptions not “L” and “R”, but “ODD” and “EVEN”. True, this, as a rule, is only on the “In-Line” consoles, but about them - later. The essence of the matter does not change, however.

There is one subtlety in the design of this regulator that is often forgotten. The fact is that there are two ways of panning - with constant voltage and with constant power. With the first method, the signal in the middle position of the PAN control is attenuated by 6dB. This is very good for sound recording, in terms of mono compatibility, but with “live” sound reinforcement, problems arise, because. the signal in the center “failed” in power by 3 dB. In the second method, the signal in the middle position of the PAN control is attenuated by 3dB. For sound amplification - great, no dips in the center, but when trying to record on such a remote - problems with mono compatibility, because. in this case, the signals in the center (in the "MONO" mode) increase in level by 3 dB. As a half measure, some consoles use the “arithmetic mean” - signal attenuation in the center by 4.5 dB.

Another node, which is also structurally and by location included in this part of the cell, is the control and listening node. (Buttons PFL, AFL, CUE, SIP, SOLO.) With these buttons you choose how the signal will be monitored at a given point on the console. By the way, this applies to the entire console, not just the input cell. There is often confusion with these buttons, because. they all perform similar but slightly different functions.

PFL stands for “Pre fader listen”, pressing this button takes the signal to be monitored before the volume control. This makes it possible to pre-monitor the signal in the still “closed” cell, before feeding it further to subsequent console circuits. In this case, as a rule, the corresponding indicators of the master section indicate the signal level at a given point, which allows you to accurately adjust it - to avoid overloads.

AFL stands for "After fader listen", listening after the fader. When this button is pressed, the signal for control is taken after the volume control, which allows you to control the actual signal level in this place tract.

SIP is "SOLO - IN - PLACE", literally - "solo - in - place". When this button is used, the control signal is taken after the volume control and after the panoramic control, which allows you to listen to the signal not only taking into account its level, but also to control its position in the stereo panorama.

The purpose of other listening buttons (CUE, SOLO and some other, rarely encountered names) is not standardized, and various manufacturers can use them to perform a variety of functions - both PFL and AFL, SIP, etc.

Sometimes - for convenience of work and saving space - instead of many different buttons, only one is placed, then this is most often the CUE or SOLO button, and the function currently performed by it (PFL, AFL, SIP, etc.) is selected by the control mode switch to the master -sections.

In cheap remotes - most often, regardless of the name of the button, only the PFL mode is used.

Another interesting control is the MUTE button. In terms of its functions, it is similar to the ON button of the cell, only it works, as it were, “on the contrary” - when it is pressed, the cell signal is turned off. Sometimes, however, this button - labeled MUTE - is actually the button for turning on the cell, only standing “upside down”. In some consoles, when MUTE is activated, the entire signal of the cell is turned off, and in some, only that part of it that enters the subsequent circuits after the fader (POST FADER). What is it for? Yes, and, in fact, why the whole MUTE at all?

Imagine that a large collective concert is being voiced, with a large number of performers. At the same time, the number of simultaneously used microphones can be different, from “all at once” - up to one, for an artist of a conversational genre or a presenter. It is better to turn off unused microphones at this time so as not to catch any extraneous sounds, or simply so as not to hiss. Doing it manually, one at a time, is long and inconvenient. It is much better to be able to pre-program which microphones are not used in which room - and mute them all at once, by pressing one button. The monitor lines going to PRE FADER remain functional. As a rule, they do not add much noise in the hall. Other applications of MUTE are possible, of course. But this is already at your discretion. Often the MUTE function has MIDI automation, more on that in a moment.

To send a signal to additional processing devices (common to all signals in the console), the “AUX” controls are used to individually control the levels of signals sent to effects devices (for example, to a reverb), and the “PRE / POST” buttons, which allow you to choose where the signal will be sent, before or after the fader.

Here it is necessary to make a small digression. The fact is that the full name of these busbars and their corresponding outputs is “Auxiliary Sends” (“Additional sends”). Over time, this name “halved” and shortened, and now you can find the names both “AUX” and “Sends”, although the former is much more common. In domestic literature, the Russian name “messages” is more common, and for the regulators themselves - “selections for messages”.

Basically, that's all there is to say about the input cells. Oh yes! Where are the promised "In-Line"? Now the turn has come to this.

Consoles of this structure are intended for sound recording, and therefore are less known in wide circles. As follows from the name itself (“In-Line”, literally - “in line”), the recording process itself is, as it were, “stretched into a line”. A cell of such a structure consists of TWO conventional cells connected in series, one after the other. The signal that came to the first cell (for example, a microphone one) is processed in it and goes to the input of one of the channels of the tape recorder for recording, and the signal reproduced by the tape recorder (usually the same channel) goes to the second one, where it is processed in the process information to get the final mix. Thus - no problems with switching, nothing needs to be switched - everything is in its place, and the work process is greatly accelerated and facilitated.

Naturally, in this case, in each “physical” cell - everything is in duplicate. Two EQs, two faders, etc. etc. True, this is the "ideal".

Why ideally? Because, in order to reduce the cost, many firms make some of the nodes combined. For example - one equalizer, switched back and forth, or divided in half - part in one half of the cell, part in the other. Similarly with “AUX”s, and with some other nodes. Only the microphone input is always the same...

There are also two summing stereo buses, similar to “MIX” on a conventional console. In order not to confuse them, in the “In-Line” console they have different names - as a rule, “A” and “B” on the cells, and in the master section you can then choose which signals will consist of the main mix - “A ”, “B” or both.

Because of great opportunities These remotes are much more expensive. As a rule, they have a very complex structural scheme, so there is not much point in going into subtleties here. In addition, there is a great variety of these structural schemes, and for each specific remote control a separate story is needed, in terms of volume - significantly larger than can be placed on magazine pages.

So - with the input cells like more - less sorted out. What's next? And then begins the area of ​​​​the greatest diversity in the designs of consoles - subgroups and the master section.

Subgroups

What are subgroups and why are they at all? It would seem that the sound of individual sources is already ready, with all the timbres and so on. What else is missing? Oddly enough - something that has nothing to do with sound. Namely, hands! A man is not an octopus, unfortunately... (Probably, many sound engineers will agree with this.)

Imagine - you have a large team, with many instruments. And in one of the parts of the song there is a long, loud drum solo (for example). You need to quickly turn up the volume of the ENTIRE drum kit... and there are only two hands!

This is where subgroups come into play. They carry out intermediate, before the main mix, the summation of several signals. In the case described above, it is possible to apply all drum sounds from individual cells first to one subgroup, and from it to the main master. And control the volume of ALL instruments of the percussion group with ONE knob! Conveniently? Still would! (However, for a stereo subgroup, you will have to use two subgroup cells. But it’s still more convenient!)

Similarly, when recording, you can collect any group of instruments into a subgroup, and submit them all together for recording directly from the subgroup, bypassing the main master, who is then freed up for other work.

The device of the subgroup cell has no fundamental differences from the usual input cell. As a rule, here are the same equalizers (only usually simpler), aux controls, panoramic controls, etc. Only the input part is missing (completely), and the send buttons to subgroups are excluded.

Although, of course, there are "possible options." For example, in many inexpensive consoles there are no equalizers in the subgroups at all, there are also subgroups without AUX. Sometimes - though not often - there are also stereo subgroups. In such cases, you can occasionally see a "tricky" pan control, based on MS-conversion, with two separate controls, one for the stereo width, and one for the direction. But this is very rare...

Recently, in expensive consoles - as a rule, studio ones, for sound recording, sometimes there are also so-called "virtual subgroups". What is it?

Yes, it's okay, this is not “virtual reality”, but something quite tangible. (Although the subgroups themselves, in their usual form, are completely absent!)

In individual cells of such consoles, instead of variable resistors-faders, the signal level is controlled by controlled amplifiers - VCA. In this case, the faders themselves produce only the electrical control signal to control the VCA. In this case, it becomes possible to combine the VCA of several cells into one group BY CONTROL, and the control signal of one fader - control the gain of several cells at once! One of the cells is assigned as the master, and the rest as slaves. At the same time, of course, all individual adjustments are also saved, because. all control signals to the VCA of an individual cell simply add up. Sometimes this method is also called “VCA GROUP”. Similarly, the work of “virtual dynamics” is carried out, but this is already a topic for another conversation.

Since in force design features- due to the absence of some nodes on the subgroups - there is free space on the front panels of the cells, then it is very often used to accommodate various additional console nodes. So, for example, on the sub-group cells in many consoles there are various kinds of additional inputs - for returning signals from external effects devices AUX RETURN to the console, and some others.

In these cases, it turns out, as in the In-line cell: in one - two. In this case, similar building techniques are often used - the ability to switch equalizers, selections for effects (AUX), and so on. Like - about subgroups, basically, everything.

Master section

Now we have reached the most, perhaps, the most important part of the console - the master section. Why the most important? Yes, because it depends on its construction how good the whole console will sound, and how convenient it will be to work with it. It is in the master section that the maximum number of controls is concentrated, the highest quality radio elements are used, and there is a maximum indication.

It would seem that the main function of the master section is simply to sum up all the signals and “give out” the final stereo mix. In principle, this is true. But - not quite. In any console there are many nodes that do not belong to any part of it “personally”, but are common to the entire console. All these nodes are usually concentrated in the master section.

First of all, of course, this is the main adder, master fader, MASTER INSERT insert jacks and the main stereo output with a level meter. These elements are in all consoles, without exception, in which there is a main master. Another node, also present in almost all consoles, is the AUX MASTER - a place where the signals of all sends to external effects AUX are summed, with individual output level controls for each AUX line. As a rule, these outputs have one of the types of listening buttons described earlier - PFL or AFL.

Also in any master section there is a signal control unit, from the simplest to the very complex. In the simplest case, this is a button for selecting the source being listened to (the main stereo output or the PFL bus), a level meter and a control volume control (headphones). In complex consoles, as a rule, there are much more opportunities here.

Firstly, if there is one multifunctional button in the cells - CUE or SOLO, then the wizard has the ability to switch its modes - PFL, AFL, SIP, etc. Secondly, it must be possible to supply a controlled signal to an external sound control system - as a rule, from the C.ROOM sockets (control room). At the same time, in addition to the smooth level control, a stepwise decrease in the control volume is also provided, usually this is a DIMM button. The attenuation introduced to it is most often 20 or 30 dB. Thirdly, in addition to the usual control buttons on the cells or subgroups themselves, a separate block can be provided for selecting various sources for control that are not “explicitly” available - for example, stereo pairs of returns from external effects, pairwise listening to subgroups in stereo mode, control external recording devices (tape recorders), etc.

In addition, in some models of expensive remotes - to check "everything and everything" - there is a built-in sound generator. It can be either the simplest - for several fixed frequencies - or quite serious, with smooth tuning of the signal frequency over the entire audio range. In the simplest cases, the generator signal is fed to its output jack and / or to the main stereo output - MASTER OUTPUT. In more "fancy" consoles, it is possible to send a signal using internal switching to any point of the console.

Another indispensable part of a serious remote control is the TALKBACK intercom. As a rule, it is possible to connect only one microphone to it (naturally, with its volume control), and the possibility of choosing a “destination point” - that is, where exactly this signal will be sent further. This could be the main output, monitor lines, etc. etc.

Very often in the master section there is also a block for returning signals from external effect processors AUX RETURN, or sometimes - EFFECT RETURN, the essence is the same. Incoming signals here are regulated by level, by panorama, sometimes - and are subjected to frequency correction. In such cases, the presence of its own equalizer is also provided - as a rule, it is simple.

In serious remotes, each individual AUX RETURN input has its own individual path - with an equalizer, a panoramic control, a level control, etc. Sometimes there is also the possibility of a “secondary send” - from the return of one effect to the send of another, or even to its own send, for example - to control the FEEDBACK level on a reverb, delay line, flanger, etc. In small consoles, for convenience and space saving, often the FX return inputs are made stereo, with common EQs (and all) for both channels.

In addition to the main functional units of the remote control itself, usually in the master section there are controls common to the entire remote control, and sometimes commutations. (This refers to the recently becoming more widespread mixing matrix - MIX MATRIX.) Common controls include, for example, devices such as MUTE control and automation, switching the modes of studio IN-LINE consoles to recording or playback, etc. As a rule, the latter in good consoles should be able to be done centrally, for many - or even all - cells at once, so as not to mess around with dozens of buttons on a bunch of cells in turn.

The MUTE functions can be controlled in two ways. One way is that the console itself can be programmed with several different combinations (“scenes”) of activating these shutdowns (“plugging” the cells). Then - instantly, by pressing one button, call the desired scene. This method is often used in inexpensive consoles, while the number of memorized scenes is relatively small. The second way is to use external MIDI devices to record and then play back the desired scenes. Naturally, the number of scenes is not limited, but this method is much more expensive, and is used only in expensive, serious consoles.

As a rule, MIDI automation of mutes "does not go alone", but is usually used in those consoles where there is the possibility of automation and other functions - for example, automation of mixing. The latter is carried out either with the help of motorized faders, or by using the VCA. But that's a topic for a completely different article...

Unfortunately, the limited volume does not allow to fully cover all the issues of “console building”. Outside the scope of the article are interesting topics, as remotes for radio broadcasting and television, digital, reportage, theater, etc. However, all remotes have many common features, and if you can figure out the IN-LINE studio remote, then learning the rest of the remotes is unlikely to be special for you. complexity.

If you have any questions or suggestions - write directly to the author. Your questions will help to better understand the range of your interests and make this site more interesting and useful for you.

Mikhail Chernetsky

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The electronic heart is made in the form of two switching circuits.
This device will decorate any family celebration, celebration, christmas tree, shop window. The switching of the LEDs is controlled by an oscillator based on the 555 series universal timer chip. The device is compact and can be powered by a battery.

The switching of the LEDs is controlled by a generator made on the DA1 universal timer chip. The operating frequency of the generator is determined by the values ​​of resistors R1, R2 and capacitor C1. Transistor keys VT1, VT2, switching LEDs, prevent overloading the output stage of the DA1 chip. Diode VD1 protects the device from failure if the power supply is connected incorrectly.

Item List

R1- 20 kOhm
R2- 8.2 kOhm
R3- 1 kOhm
R4, R5- 22 Ohm
C1- 22 µF/16…50 V
VD1- 1N4148, KD522
VT1- BC547, BC548
VT2- BC327, BC557
DA1- HA17555, 555 series timer
Red LED-40pcs
A514 ( printed circuit board 72x74 mm)

Railroad crossing

This circuit flashes two red LEDs for a model railroad crossing.

Brightness control.

This circuit will dim one or more LEDs from 5% to 95%.

Alternate flashing of LEDs

Some LEDs are paired, such as red and green. This circuit causes the red/green dual color LEDs to flash alternately

Flashing Bipolar LEDs

The following circuit causes paired LEDs of different polarity to flash.

Alternately flashing red / green bipolar LEDs

Roulette

This circuit creates a spinning LED circle that spins very quickly when a finger touches the sensor wire.

When the finger is removed, the rotation slows down and stops.

Multiple LED control

The 555 timer is capable of handling up to 200mA and 12v. The following diagram shows the maximum number of white LEDs that can actually be connected to a 555, but we have limited the total current to 130mA since each LED is designed to carry about 17mA to 22mA maximum.

The voltage across the white LED drops 3.2V-3.6V, which means that only 3 LEDs can be placed in series.

3D cube

The circuit is a 3x3x3 cube consisting of 27 white LEDs. The 4020 chip is a 14 pin binary counter and we used 9 outputs.

Each output controls 3 white LEDs connected in series. The 4020 chip produces 512 different codes, before the sequence is repeated, you must build the circuit to see the effect of the 3D cube for yourself.