How to create your own sound production - or review of a budget home studio

I think that everyone, at least in the slightest degree of musician (at heart or in fact), thought about building his studio, and even with his own hands! But often, unfortunately, there are a lot of problems on the way to creating your own "musical corner", both financial and "interrogative", such as: ... what is better to take? event budget? Where? how to install? what software to use? ...? and many more different why and why.
In a series of my articles, I will try to uncover the theme of organizing my audio production as widely and deeper as possible, albeit somewhere technically incorrect, but in an accessible and popular language using the example of my own studio and personal experience. The basis of my story I put such a meaning - “Cheap and angry!” - I think that such a factor as price-quality ratio is most relevant for this kind of occupation, so if you have “anlim” money, then you are not here (but read I still recommend).

Chapter 1. Start
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Section 1. Workstation

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Nuss, let's start, and let's start, perhaps, from the most basic, albeit somewhere quite banal and mundane, but believe me, this is the main element of the modern sound industry - your computer.
- I have a mega cool comp for $ 1hhhhhh $
- Wait, here please for more details! What is the axis? Audio? What settings are you using? etc.
Everything in order:
1.1. Axis
It is the operating system, namely, the algorithms of its sound subsystem create the main weather! What is the difference you ask? I will try to explain intelligibly. Imagine a layered cake consisting of different-sized, written by different offices, often “crooked” code - this is about Microsoft, well, don't worry, avid Windows users, but that’s it. Let's explain. As far as I know, this is exactly how sound “lives” in the modern axis. For comparison, take Windows, Mac OS X and Linux.

Windows - I don’t know how anyone but personally I can hear the difference by ear. I have not been using Microsoft products for 5 years, and at the same time I don’t feel deprived ’or offended’ because of shortages, flaws in software or something else, rather the opposite.
Mac OS X - Everything is fine. Available, understandable, stable, shaft professional software, but expensive. In my case, I use only OS, on a non-original gland. This way I get a workstation with all the benefits but no problems with anything.
Linux is free, stable, good, but there is no software - I have this axis just writing sound.

1.2. Audio interface

The modern market of audio devices is replete with diversity. It is important to understand that if the conversation is about high quality, then 2-cent chips onboard sound, all kinds of creatives and other stock and home interfaces are automatically dropped.
The main differences:
1) a good interface is always built on a serious chip no worse than ENVY 24
2) it is always equipped with large TRS or XLR connectors
3) its cost starts from $ 200
4) be sure to support 24 Bit 192 KHz
5) powerful DSP *
6) availability of TOS, ADAT, SPDIF, AES-EBU - to taste and necessity.
If all these characteristics are enclosed in a PCI or PCIe card, it is important to understand that the laws of wave physics work everywhere and for everyone in the same way, so do not be surprised at the appearance of extraneous noise from video cards, coolers and other stuffing on your PC. Here, only the shielding of the device, the soldering / doping of the power supply tanks and other tambourines can help you. I once had to face this, the problem was solved, but it took a lot of effort, health and time.
Everything is really simple, absolutely all studios use external Fire-Wire or USB interfaces, probably not just and that makes sense. In my case, I use ESI Maya 44 interface with optics on board ESI MI / ODI / O for recording PCI - it is not a problem for interference. For the workstation, I took the Fire-Wire interface with powerful DSP * and “pleasant appearance” from TC Electronic Desktop Konnekt 6.
* DSP - it is he who is responsible for processing the streams in the audio interface. The more powerful the DSP, the more tracks and plug-ins it can handle.

1.3. Audio settings
Don't get excited, IEE 32 Bit Floatig Point 192 KHz is certainly cool, but why? Of course, I don’t suggest using 16 Bit 44.100, although I know a couple of “cool” studios that work as such, using 24 Bit 96 KHz will suffice here. In any case, below -150 Db you will hear absolutely nothing, and even the character of amplitude-frequency distortions, even at 48 KHz, tend to zero at frequencies up to 21KHz, when the human ear is able to subdue up to 20KHz and not always. Here, settings in the range from 48 to 96 KHz in 24Bit will be acceptable.

1.4. Formats
Here I will be brief, WAVE, AIFF, RIF. Everything else is not compatible with the concept of "quality"

1.5. Soft
As they say "the taste and color of the markers are different." And in fact, what difference does a sequencer or software sampler use? Not what! This statement is true for Mac OS X and Linux users, and as for Windows users, in their case, each software has its own audio engine, there is a difference. As for the rest, who is more comfortable in what, that is what he does.
I always could not understand how the software, originally positioned as a software synthesizer for children, can claim a place in the line of professional software? How can we expect a "smooth" sound from all sorts of FL Studio and other "samograek"? There is a number of professional software that has proven itself in the world of sound, such as Cubase, Nuendo, Reason, Samplitude, etc. In this question you will be helped by an overview of major studios running Windows. In my case, I use the full package version of Logic Studio 9, Propellerhead Reason 5 - MAC OS X and Ardour - Linux.
Logic Studio 9

Logic Studio 9

Propellerhead Reason 5

Ardour

Section 2. Acoustic Systems

Armed with the right computer with a good audio interface, it becomes a question of choosing acoustics. You should not waste your time on all kinds of household and 5.1 systems, this is all complete nonsense. None of the household speakers I know is capable of giving such characteristics as studio sound. Studio sound is always bad and not familiar to the average person, but the problem here is not the speaker, but the fact who says that.
We are accustomed to listening to household acoustics, bad or good, more like or less, but it always deceives us, because it is not able to show us the fullness and character of the sound spectrum. The main difference between studio speakers and household speakers is how much equal frequency response (Amplitude Frequency Characteristic) they have, the more equal it is, the less dips and peaks in it are the better. In this section, I consider it necessary to highlight such characteristics as the frequency range, frequency response, sound pressure, resonant frequencies and field type.

2.1. frequency range
As mentioned above, the human ear is able to detect sound vibrations from 20 Hz to 20 KHz, this is a scientifically proven fact, which I think is not disputed. Why is the studio sound such a wide range you ask? Everything is very simple, any dynamic head has a zone of decline, in other words, the limit of its capabilities.

Accordingly, the farther from the cherished 20 Hz - 20 KHz, the smoother the frequency response we get in the useful range. So you shouldn’t look askance at such characteristics as 15 Hz to 24 KHz, this is not a typo or home speakers of aliens, these are quite normal characteristics of professional sound.

2.2. Frequency response
AFC (Amplitude Frequency Characteristic) - the holy groan of AC builders. The best minds of the largest manufacturers break the brain in an attempt to achieve perfect graphics. And it depends not only on which dynamic head was used, but also on how many of them, what material they are made of, their diameter, the volume of the system in which they are built in, which amplifier was used, what type of filter, etc. All these technical details that at this stage should not disturb us, now we consider the main aspects affecting the frequency response of monitors.

1) carefully read the frequency range of the speakers, it should not be worse than 45 Hz - 21.5 KHz
size 2. There are no small and light monitors! Why? More on that later.
3) the diameter of the dynamic head. It is directly proportional to sound pressure.

2.3. Sound pressure
Did you know that the volume consists not only of the digit with the letter “W” at the end? Not a few important factor, rather a very important one, which depends directly on the area of ​​the diffuser, the dynamic head, the volume of the box and its design, is the sound pressure. It is measured in a unit of signal level per volume, ie, in decibels per cubic meter Db / m³.
A good monitor always uses a dynamic head not less than 6 "with a large magnet (which indicates high power), a dense tree or a chipboard with a thickness not less than 6 mm. That is why the monitor cannot be small and light. The combination of all these factors forms sound pressure, so for example, if you send a signal from an amplifier with a power of 100W to speakers with a diameter of 5 "and 15", the difference in the volume of their sound will be approximately directly proportional. But there is a limit for this dependence, for example, the same power is simply not x Amum for Diffusers diameter of 2 m, everywhere has a golden mean.

2.4. Resonance frequency
Each frequency has its own wavelength, each object having the same or divided by 2, 4, 8 ... etc. the length will go into resonance, to put it simply, vibrate. The same rule applies to sound waves. The box of any speaker has a number of frequencies at which it will enter into resonance with sound with varying intensity, it depends only on how correctly it was calculated. Boxes made of a thin material and having a too small capacity relative to the diffuser area of ​​the dynamic head, etc. are most susceptible to this phenomenon. The most important thing you need to know when choosing a speaker is that it is a regrettable phenomenon that absolutely does not contribute to the correct sound, so before you buy monitors, you will need to stock up on a disc with a recording signal changing from 20 Hz to 20 KHz. Armed with this not tricky tool, you can easily hear it when playing it, with what intensity the speakers enter into resonance, which will be indicated by a sharp increase in loudness at one or another frequency.

2.5. Field type
All speakers are divided into three types of field of action, near field, middle and far respectively. The difference is in how far from them the listener needs to be in order to achieve the maximum effect. Distinguishing them is not so difficult, if you do not delve into all the technical subtleties and physics of sound, then we can say this: the smaller the column, the smaller its field of action.
Often studios use near and middle field monitors, but many have a complete set at their disposal. I personally use near field monitors.

2.6. Conclusion
Thus, when choosing a studio speaker you need to know:
1) the frequency range of the speakers should not be worse than 45 Hz - 21.5 KHz
2) there are no small and light monitors
3) the diameter of the diffuser must be at least 6 "
4) the box resonance should be minimal
5) The frequency response should be as smooth as possible without large peaks or dips in the claimed frequency range.
6) it is necessary to determine in advance the type of field
7) the cost of good sound can not be less than $ 600.
In my case, I use the active monitors ESI nEar 08 Experience, F & D 31 and the active subwoofer ESI SW10K

Section 3. Speaker Management and Portal Processing

It is not enough to have good acoustics, you must still be able to use it well and correctly. After reading the section on resonant frequencies, frequency range and sound pressure, I think many people have a lot of questions, such as how to fill the gap between 20 Hz - 45 Hz, what to do with the resonance of the room, how to install speakers, etc.
This section will describe the methods of the portal management of the AU and the equalization of the room.

3.1. How to install?
It is very important to know how, where and how to install your speakers correctly, because the final result of the room sounding will depend on it. I will try to explain visually.



It is very important to arrange your acoustics at an equidistant distance from the listener, walls and objects so that the center point of your speakers is at the level of your ears and sounded right in them, and not at the cat Vaska who was frightened by the roar of your acoustics and who was lost in a panic under the sofa. But as practice shows, it is not always possible due to various reasons, such as the characteristics of the room, its filling, etc. Everyone knows that sound has its own speed, so it takes a certain amount of time to overcome a certain distance. Thus, the sound coming out of the speakers passes you, is reflected from the walls and returns to your ears. We get an echo. The echo can be different, from noticeable delays to completely imperceptible to consciousness, but extremely spoiling the overall picture of the rumba sound. The way out of the situation is to buy noise absorbing material and glue all the walls to them, put basal catchers in the corners, cascade pendant and carpet on the floor on the ceiling. All this is very correct and good, but again it is not always possible and "budget." Option two, put another pair of speakers behind you (studio quality is not necessary here) and connect all this via the speaker manager or portal processor with a delay line.

3.2. Portal processor
Typically, such devices include a whole bunch of static and dynamic processing, from an equalizer with a crossover, to a limiter and delay lines. All this is useful for setting the "right" sound.
Crossover (eng. X-OVER) - a device that shares the frequency spectrum. Designed to separate the sound between speaker types, i.e. high, medium and low frequencies. It is mainly used to take a low spectrum from broadband acoustics and feed it to a subwoofer. Having read the characteristics of your speakers, you can guess that if the lower limit of the broadband ruler is for example 45 Hz, and the upper limit of the subwoofer for example is 100 Hz, then the cut-off frequency of the crossover must be 70 - 75 Hz in order to relieve both lines equally.
Equalizer (English Equalize - “align”, abbr. “EQ”) - a device that allows you to align the frequency response of the sound signal, ie adjust its level selectively, depending on the frequency. The device is quite important, but at the same time no less dangerous for the frequency response. When using it, I do not recommend being guided by the principle of “by ear”, because the auditory perception of sound waves is a rather individual and far from ideal process. In order not to get a “dumb button accordion” from an excellent speaker system, you need to use a reference microphone, an equalizer of at least 30 bands and a signal spectrum meter with a tone generator. Only by relying on the dry reality of numbers can you properly rebuild your sound. I want to note that a number of portal processors and EQ, when a microphone is connected to them (I repeat - the Reference microphone), perform this setting automatically. The principle is as follows: The tone generator generates a sound wave in the range from 20 Hz to 20 KHz to your acoustics connected via an equalizer, the spectrum meter in turn “listens” to the microphone what is happening to the sound. By changing the tone frequency in the tone generator and the level in the equalizer at a given frequency, we achieve maximum equality in the frequency response.



Delay Line (English Delay Line) - a device designed to delay electromagnetic signals for a certain period of time. As it was already said in p. 3.1 it is not always possible to arrange the front and rear speakers from the listener at the same distance, in this case the delay line comes to the rescue. As a rule, the settings for such devices are quite simple, they simply need to specify the difference in distance in units of distance (meters or feet). It is important to remember that you should “slow down” the sound in the speakers closest to the listener.
Limiter (English Limiter) - a device designed to limit the signal level. It is this device that will help you to protect speakers from overload and sudden signal jumps. In the settings of this device, you must set the maximum level of the input signal in units of measurement, for example -4 Db.
In my case, for AS management, I used Mainframe EQ: Behringer 8024, SMC: TC Electronic XO24 and JBL DSC 260 for the subwoofer.

Section 4. Routing

It is important to understand that all studio equipment must not only be conveniently located, but also conveniently connected. In this section, I will tell you how to organize audio routing correctly and conveniently using the example of my studio.



4.1. Split connectors
- What a nice set of wiring I saw yesterday in the audio shop! for only 100 rubles! Here I go tomorrow and buy it! It will now connect the guitar.
- Class. And I took yesterday connectors on the market for 15 rubles! Can you imagine? A total of 15!
This is what most of us do, cheap shoelaces and stock connectors. When I did that. Until a certain point. Until I realized that because of such junk, my health began to limp as well, which I was desperately spending on constant soldering of scattered connectors and filtering out obscure noises made as if by magic from nowhere. Here I can advise only one thing, forget about the Chinese connectors and laces. Use only proprietary connectors such as Neutrik and Klotz cables. This will save you not only time, but also money, because it is easier once to deliver normal expensive products than to constantly buy cheap counterparts. That is what will come out. Whenever possible, use digital connections of your devices, because their noise immunity is enormous! I think that it is not necessary to chew for example the advantage of a fiber-optic cable over the "noodles" - the optics "taxis" definitely.

4.2. Record
I hope you are not going to poke a guitar or microphone directly into your audio interface? That's it! This act will be at least extremely stupid, not taking into account that it’s not really possible to write anything down.
There is a preamplifier, directbox, attenuator, etc. for any instrument for sound removal. As practice shows, even specialized audio interfaces having inputs for professional microphones and instruments do not always cope with the task set correctly. There is a place for everyone, for a guitar directbox, a combo amplifier, for a microphone a preamplifier, an attenuator, a compressor, a deesser, etc. All these are integral parts of the recording process that must be connected in a certain sequence.
1) Microphone. Vocals and / or instrument:
Microphone -> Preamplifier -> Compressor -> Expander / Gate -> Equalizer -> (Harmonizer -> Effector ->) PC / Recorder
2) Guitar:
Directbox -> Combo -> Microphone -> Preamp -> Compressor -> Expander / Gate -> Equalizer -> (Harmonizer -> Effector ->) PC / Recorder
- either
Directbox -> Preamplifier -> Compressor -> Expander / Gate -> Equalizer -> (Harmonizer -> Effector ->) PC / Recorder



4.3. Reproduction
The sound should be played not only in your speakers, it should also sound in the headphones of the person who is recording at the moment, and not only the soundtrack should play, but also what he sings or plays. Not least important aspect is the microphone intercom, with which you can communicate. An example of such a routing is shown in the figure at the beginning of the section. In my case, I used a lot of rings to achieve the greatest functionality. Such a solution may seem difficult somewhere, but its possibilities are truly enormous.

In this chapter I rather superficially touched the main aspects of the studio’s organization, but in subsequent chapters I will try to reveal in more detail each of them, to consider in more depth the methods of working in a studio with a particular material. See you again!