The compressor is a most usable and interresting thing. With this marvellous thing, you can affect the dynamic of the soundsignal.

This may sound a bit wierd at first, but once we take a deeper dive, it will hopefully make more sense.

Just a quick background. A soundsignal is measured in deciBels (dB). There are a lot of variations to different dB scales, and I will not go into depth with those for now, so let’s just generally agree on that dB is the scale that measures the energy of the soundsignal. The greater engergy, the louder is the sound and the larger the dB value will be. And the opposite, the smaller energy, the quieter the sound is, and the smaller the dB value will be. So loud = high dB. Quiet = low dB.

If you look at a recording of a human voice, either spoken or sung, you will see that the soundwave varies a lot from very high peaks to very small peaks.

When you listen to such a file, you will eventually see the correlation between louder parts and higher peaks, and quieter sections with lower peaks.
Keep this in mind.
There are two more steps to take to fully understand the compressor. The first is the Threashold and the second is the Ratio. (Then there is the attack and release, but let’s save those for later.)

The threashold is a value in dB where you want the compressor to start working. As soon as the soundsignal gets stronger (louder) than this value, the compressor kicks in with the force that you set on the Ratio knob.

To set it in perspective.

If your soundsignal is a song and you sing quietly in the verse, your song may stay below the threashold, preventing the compressor from doing anything with the soundsignal. And then you come to the refrain where you start to sing a little stronger, and the soundsignal passes over the threashold value, and as soon as you do, the compressor starts to work.

As I wrote before, the work the compressor does is set by the Ratio. For example the ratio can be 2:1, which means that counting from the threashold value, every 2 dB that the signal gets stronger will only result in 1 dB increase of the signal. 4:1 means that counting from the threashold value, every 4 dB that the signal gets stronger will only result in 1 dB increase of the signal. (Setting the ration to infinit:1 (infinit being the symbol 8 laying down) the signal will never get any stronger than the threashold value, and then you will have a limiter.)

The attack and release values (if available) determins how fast the compressor reacts when the signal crosses the threashold, and for how long after the signal has crossed the threashold the compressor should keep working. The benefit of those functions is to get a smoother compression or a harder compression, depending on how you set the values. With a fast attack and a fast release, you can get a pumping sound (which can be good for i.e. drums or bass), while a slow attack and slow release will get a smoother compression. But that is not entierly true. All depends on the soundsource, so you need to experiment to gain knowledge on how to best make use of the compressor.

Well, that is in short terms what the compressor does, but nothing about what you can use it to, or why it is good to use it.

The recorded signal is most often very dynamic, meaning it has it’s loud (strong) parts and it’s more scilent (quiet) parts. If you want to make the signal stronger (louder) you will boost the entire signal, and the limit of how much you can boost it is the strongest peak of the signal. Sometimes you want to boost the signal more. So the compressor will “push down” the strongest peaks, giving you more room to boost your signal, since the strongest peaks now have a lower dB value than before the compressor was applied. The downside is that you have less dynamic in your sound signal. And also, when you boost your sound signal, you also boost the background noise, both the background where your recorded, and the noise that all electronic equipment emits.

There is a Golden rule that I learned from a man much wiser than me:
Use the compressor a little when recording, and than a little when you are mixing. This to prevent the sound signal from being overcompressed in one go.

This I have developed to a thing of my own. When I record, I run the microphone through a Mic Preamp with compression included. (The one I use is a dBx 286a, as to where the more modern version would be the dBx 286s.) The Mic Preamp is connected to a mixer, and then I run one of the Aux sends through another compressor that I compress even harder. And then I blend in the output signal from the secondary compressor with the signal from the mic preamp. This way I boost the lower parts with the highly compressed signal, and keep the dynamic (with only a little compressionrate on the mic preamp) from the original signal.

This can also be done within your recording/mixing software. The way you do that is that you clone the track that you want to apply this technique to, make sure not to move the soundfile sideways. Should you do this, the effect will be totally different (and can be awesome in a different way).
The “original” track, apply a little compression to this, and then apply a heavier compression to the clone track. Then adjust the volumes to find a balance between the two. Typically the original track is stronger (volume wise) than the cloned track. But as I said earlier, every sound signal is uniqe so you need to find the balance between the two tracks.

This is a small chapter regarding the equalizer. For some people, equalizer is something on their sound system, two knobs, one for bass and one for treble. For others (like me) this is a very simple thing that quickly get very complex in it’s simple nature.

The equalizer comes in many different forms, from simple two, three of four knobs on a mixerboard channel to it’s own stack with multiple bands for each channel. But no matter how it looks, the usage is the same.

The whole point with an equalizer is to cut or boost the audiosignal in a specific Hertz (Hz) range. This means that if you use an Equalizer knob marked 3kHz (= 3000 Hz), you will affect the audio signal in (and around) 3kHz. If you turn it up (+) you will boost the signal which will make that part of the signal louder. If you turn it down (-) you will cut the signal which will make that part of the signal more quiet.

In general all Equalizers affect the specified value (Hz) and a range above and below that value. And if there are two knobs, they are usually for treble (high frequensies) and bass (low frequensies).
Then there can be three, and then it is usually for mid-range usage. Every now and then you’ll find four knobs, and the fourth would control the value of the mid-range, changing if from darker to brighter area of control.

For my personal tast, I prefer to cut out content rather than boosting content. But that is my personal preferance. I usually cut out unwanted frequensies from the signal.

A good thing can be to cut out low frequensies from a recorded signal (or even better, use it while recording, so those frequensies is not polluting your recording in the first place) like buzz, traffic noice etc. And on the opposite end of the scale there is the high frequensies from ventilations or draft in windows, that you can filter out using the Equalizer.

Should you have a multiband equalizer each controller will affect a smaller range, and as so, you can isolate only the frequensies that you want to use from each specific signal. But this is rather advanced stuff and nothing I recommend doing from the start.

I hope this small section has given you a better understanding of what the Equalizer does, and I urge you to play with it. It is a powerful tool, but it can also easily and quick turn your signal to a terrible signal. Play around and listen to what your equalizer does to your signal, try to use it in a wise way.

I dedicate this session to record vocals for the simple reason that it is one of the most common questions I get. How do I record the perfect vocalsound at home in my appartment, using only minimal budget equipment. Well, you don’t! But I hope you can prove me wrong! It is very few places (professional studios, or semiprofessional or even home setups) that can produce the sound of Celine Dion or Bon Jovi. But on the other hand, don’t try to mimic the sound of others, instead, try to focus on finding your unique sound.

First, check the Microphone part of Guide of Sound to get input on what microphone to choose, if you are lucky enough that you have several microphones to select from. If not, just go with the microphone you have.

Secondly, check the Recording part of Guide of Sound to make sure you understand the challanges with levels at different points.

If we start of with the distance between the microphone and your (the singer’s) mouth. I usually recommend using the “Thumb rule”. By this I mean about a thumb’s distance (5-7 cm) between the microphone and mouth. For some singers, this distance is to short, as it causes the microphone to distort when they sing too strong. and then you will need to adapt that distance and move the microphone further away.
The oposit is very rare, but, if the singer sings very quiet and you are aiming for “a sexy proximity” in the voice, you can decrease that distance to a minimum. What most often will happen is that the voice will get recorded with a little more base (depending on the microphone, of course).

Basicly, there is not much more to is. But I can always give you a few more tips on the way.

The first would be to use puff protection, you know the thing that looks like officer Nordbergs (from Naked Gun 33 1/3) hair in the 70’s scene. There are also other things looking like old socks or a ring with a nylonsock in it, well, you know the drill.
These are used to scatter the heavy airpuffs our mouth are making when forming the sounds B, P, T, F, S and so on. To understand this better, put your palmed hand infront of your mouth and read the entire alphabeth from A-Z. Every now and then you will actually feel the air comming from the mouth. This is what is going straight into the microphone membrane causing a “puff” or distortion. It does not sound good when you get that, so puff protection is important.

Three other things that are very important is:
1. Use a microphone stand. Otherwise the singer may cause “handmovementnoice” on the microphone (it will pollute your recording with unwanted sound) or make the cable move (it may also pollute your recording with unwanted sound).
2. Make sure that the singer does not hold or physically touch the microphonestand, the cable or the microphone through out the take. It may pollute your recording with unwanted sound.
3. Make sure the singer does not stomp to the beat, or make little dancemoves or something similar. Stomps or steps on the floor might be caught by the microphonestand an may pollute your recording. And movement with some types of clothes fabric will also create a clear hearable noise that may polute your recording. (With this not saying that all singers should record their voices naked, but sometimes it would help.)

Some rules for the singer:
1. Be quiet (completely quiet and still) a few seconds before the recording starts and after the recording is done.
2. Be mindfull of your breathing. It is widly known that everybody needs to breath. While focusing (and maybe have headphones on) it is easy to forget your own breathing, when recording, try not to forget your breathing. Try to breath extra quiet and controlled. Especially in between your song sessions.
3. When you sing you will eventually run out of air, it is fully naturall. If possible, turn your head away from the microphone when you take your breath, and then turn your head back to the microphone. This will reduce the impact your breath has in the recording, making it easier to edit out in a nice way.

My final piece of advice:
Record as much as possible in one session. As soon as you leave the room to do something else, the mindset of the singer and possible the entire sound properties of the room might change (since you actually might have moved stuff around and/or changing the temperature of the room) and you will end up with two different pieces of takes that might not be fully compatible with eachother.

Now I will talk about recording in general. Even if I know there is plenty of analog equipment out there, still recording on tape, I will focus mainly on digital recording. As for the basic principals, it does not really matter if you record analog on a tape or digital on a harddrive or somekind of flash memory card. You still have your sound source, something that captures it, creating an output signal (or if you are using the line output of your sound source), something that recieves the signal and eventually turning into a stored reprecentation of the sound source that can be played back at your choise.

Reason for mainly focusing on digital recording is that it is less forgiving than analog recording. As where analog recording can peak and reach its maximum inputlevel, it more often creates a softer distortion compared to a digital distortion, given the nature of analog components versus digital components. When a digital signal reaches its maximum peak you will get a sharp, clear and hard distortion that stands out from the wanted content.

But lets start from the beginning.

Now I will talk about recording in general. Even if I know there is plenty of analog equipment… hmmm… feels like I am repeating myself, well, yes I am, I litterally started from the beginning. 

But seriously now.

No matter what sound source you have, if it is a microphone that generates the electric signal, or if you use an output of a device that generates the electrical signal, you will have something that recieves that signal, and I will look at two scenarios here.

Either you have your sound source directly connected to a soundcard in your computer, or you run the signal throuh a mixerboard. There are several other possibilities, but I will not go in to them all, since every other scenario I can think of, will be equal or close to the mixerboard scenario.

Sound source direct to sound card in computer.

This is the easiest setup, at least in terms of things that can go wrong, but it is also the most fragile and put an extra high preassure on both signal from the sound source and on the sound card itself.
In this setup, you can not affect the output signal from the microphone, only the inputlevel of the soundcard. The trick here will be to have as high input volume on the soundcard without getting distortion. Here you have to use somekind of meter, either built in OS or native to your audio recording software. The downside with this type of setup is that soundcards often have a high backgroundnoise, which also will get louder as you turn up the inputgain of the soundcard.
Should you experiance that the sound gets too noisy, or if you can identify distortions, you need to turn the gain down again. So make sure to do a couple of “dry runs”, before you actually start the recording, so you know that your levels are solid when you are recording.
Should you realize that the levels where off after you’ve made the perfect take, you can not fix it in post production and you need to do a retake with better calibrated levels. Unfortunatly, there are no shortcuts to get the perfect sound, you just have to learn the hard way.
But on the bright side. Once you’ve made a few misstakes and felt the frustration of doing them, use them and learn from the experiance.
Please note: a distorted signal might not be due to the input level of the soundcard. It might be the wrong type of microphone or simply the microphone too close to the soundsource.

The other setup I will go through here is Microphone attatched to a sound mixer who is connected to the soundcard.
This will make the chain ever more complex, but also more adjustable and potentially better sounding, than the first example.

The tricky part with this setup is that the volume of the signal is controled at multiple points, and wrong volume on one point can affect the entire chain.

If we start with the output of the microphone, it is connected to the input on a mixer channel. Now the typical mixer will have a gain knob that adjust the inputlevel on the channel. If the gain is set to low, you will have a low and possibly noisy signal through out the chain, if it is set to high, you have a high risk of unwanted distortion.
After the input gain the signal is possible affected by equaliser etc, but I will skip that for now.
Last point on the individual channel is the channel’s volume fader (or volume knob) that will control how strong the signal is when it leaves the channel and is transered to the output of the mixer.
To complex it even further, there is volume faders (or volume knobs) that adjust the output level as well.
The last step of this chain is the input level of your soundcard. Also adjustable.
So, to put it in the simplest language I can, there are (at least) four seperate points where you can adjust the volume of the signal and get it either to low (adding unwanted noise to the signal when it is normalised in post process) or to loud (causing unwanted distortions).

The golden rule of setting the “right” volumes is that there is no golden rule. Every signal is uniqe and needs to be optimized on each point on the way to get the maximum out of your signal. But looking at it step by step again.

Input gain on the channel. Set this as high as possible, without getting distortions. Some of the input gains have a red light so you very visible can see when the signal gets too loud. Should you turn this all the way down, and still have distortions, you might have the microphone to close to the sound source and it needs to be moved further away.

The second step is the channel volume fader (or knob). Aim to get this up to (but not over) the marked area on the fader (I will not go into how it is marked since there are several ways for manufacturors to mark this). In generall you can say that passing the mark will amplify the signal, and keeping under it will reduce the signal. I personaly want to avoid amplifying the signal at this stage, if it is not absolutely nessesary.

Last stop on the mixer is the main output level who is adjusted by the main output faders. Again, I prefer not to have the signal amplified at this stage, but if it nessecary on the mixer, I prefer to do it here. Keep an eye out on the meter led’s, should they get into the red area (note: not all mixers have meter, nor red areas on those meters) it might indicate that you are in danger of getting distortions.

Last step is the input gain of the sound card. Set this as high as possible, without causing distortions.

Now, I hope you have a better understanding of the complexity you (might) face when wanting to record a sound. And I am fully aware that above description might scare you and prevent you from trying. But please do! The only way you can learn is to try and experiment. You will probarbly fail a number of times. Embrace those times and try to learn from them and make it better next time.

In the end. If it sounds good you did it right, if it sounds bad, well… better luck next time!

Tips: When you record through a mixer, make sure that all other channels, including aux returns and 2-track input are muted and not part of the outgoing signal from the mixer. They can all be potential noise generators!