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!

As me, most people start of creating music in the comfort of their own home, since most of us do not have the financials to build or rent a studio.

Over the years, I have gathered a few tricks that I share with you, here and now.

To understand how to create an (as) optimal (as you can get at home) environment you need to know what happens to the sound in a room. And not as your ear hears it, but as the microphone registers it.

As you might have already read in the “What is Sound?”-section of Guide of Sound, you know that each soundwave has a (or rather many) frequency. The frequency is measured as the length it takes for a cycle to complete (or the distance between two wave tops). The reason for bringing this up again is that should you, in your room, have two parallel surfaces, say two walls or the floor and the roof, there is a fixed distance between those surfaces. As your sound travels through the room, it will bounce back and forth on all hard surfaces, and if you are unlucky, your sound source emits a frequency that is an exact match (or double, or triple, or quadruple etc) of the length between two parallel surfaces in your room. This cause a “standing wave” and it boosts the amplitude of this particular frequency, making it louder than every other frequency your sound source have emitted. So you have to watch out for standing waves (in reality, it is hard to create them at home, but you should be aware of them).

What is far more common at home is that you have hard surfaces here and there, like the walls, a painting with cover glass, bookshelves, a roof and a floor, windows, furniture etc.

On every flat and hard surface the sound wave have the opportunity (and it will use it) to bounce off in another direction. This is what causes the sound of the room giving the room its character. In most cases (far from all cases) the sound of the room is unwanted, so you have to listen to how your room sounds through a microphone before you start making changes.

Put the mic where you intend to use it to capture your sound source. Listen in headphones while you stand in front of the mic clapping or snapping your fingers or create a noise with your mouth (I will not assume you can sing).

You will probably hear that even if you stand rather close to the mic, you will have (unwanted) support from the room itself, giving your sound source its own colour and ambience.

In cases where the ambience is indeed unwanted, there are a number of things you can do to reduce it. Be creative, use your imagination and stuff you have at home.

Here are a bunch of stuff you can try:

• Break parallel surfaces. Use what you’ve got, if you have a bookshelf with books (or movies or CD’s), make sure that they are standing irregularly (yeah, I know, it looks terrible, but it will help to break the sound waves from bouncing around in your room).
• Put pillows or a mattress on the floor, cover things with blankets and other soft materials.
• Cover windows (no not on your computer screen) with curtains or blankets.
• Open your wardrobe and put the mic in front of it, so you actually sing(?) straight into your closet.
• Use your winter clothing and put it all over the room.
• Turn the TV or computer screen so its flat hard surface does not face another flat and hard surface.
• Towels, if you have read the Hitchhiker’s Guide to the Galaxy, you know you should never be without a towel, spread them around together with your clothes, defy your mother and build small piles with clothes here and there on the floor and on the tables. Does not matter if they are clean or not.
• Turn a bookshelf and pad the back with your mattress and pillows and blankets, creating a small, isolated (well padded) space to put your mic to minimize the room ambiance.

In other words, use what you’ve got. All soft materials will help to break the bouncing of the sound creating less ambience, giving you a more dry signal from your sound source.

Don’t forget to listen to the mic in your headphones as you keep making the changes in your room, so you know how it affects the character of the room!

I often get the question: Which microphone is the best, or What microphone should I use, I want just one that is good for everything!

My reply is very often: How long is a piece of string?

The most common answer to that is: It depends!

And there you go! There is your answer!

It depends on what you are after!

Many seek the perfect and clear microphone sound, only to realize that it wasn’t what they were looking for. Many microphones that have been used throughout recording history that has been widely used is because they have a certain character, giving a certain sound. So in order to help you in your choice of microphone I will go through the simplified basics of different types of microphones.

Please note that this is the simplified basics, if you want to go in depth on microphones you can do so for several hours. Which is true for just about anything… 🙂

First of all – a microphone contains a membrane that registers the vibrations (or movements) in the air. It is the equivalent of our ear. So, a microphone is the ear of the audio recording chain.

Dynamic vs. Condenser Microphones

In very general terms.

*Dynamic mic’s does not need a power source, the membrane react to the movements in the air and induces a current that can be recorded.

This makes the membrane heavier to move and you often need more energy in the sound to be able to get it registered by the membrane.

*Condenser mic’s does need a power source, either a battery inside the mic itself (usually 9V battery), or phantom powered by the equipment it is connected to (usually 48V).

Since it’s already driven by power, the membrane can register smaller changes in the air, making it more sensitive that a dynamic mic (in general).
Keeping above in mind, should you want to record gentle sounds, condenser mic’s would be the natural choice, with the downside that it is more sensitive for transients (spikes) that can cause unwanted distortions. Condenser mic’s are often more clear in their sound with less colouring.

Should you want to record that are very dynamic in it’s range, rich with transients, the natural choice would be a dynamic mic. And dynamic mic’s often have their own character (all mic’s do, not only dynamic) and colouring of the sound.

To make it more complicated, I want to add two more aspects, membrane size and microphone characteristics.

The size of the membrane on the microphone is also essential as to how it registers the movements in the air. A smaller membrane often has less span to move in, making unwanted distortions more common as a result of transients in the sound.

As to where a bigger membrane often has more room for movement, making the risk for unwanted distortions due to transients less (although, that risk is always present).

Larger membrane usually gives a cleaner or cleared sound than a smaller membrane.

Microphone characteristics. This one is tricky. There are a lot of variations on how and where the microphone has it’s “optimal pickup range” in terms of direction of the sound. There are omnidirectional mics that reacts to sounds in every directions, there are directed mics that reacts to sound in a specific direction and there are variations in between. The important thing is that you know what characteristic the mic you are using have, so you can use it and place your sound source within the “optimal pickup range” of the mic.

Bear in mind (no, not beer in mind) that above is very general basics, and there are unlimited variations and examples that you can find proving above wrong by comparing different microphones. So keep reading and look for input from other sources to get a better understanding of microphones.

Last but not least, microphone usage.

General thumb rules are:

*The closer to the sound source, the greater risk for unwanted distortion, but also less of the surrounding sounds and the room itself.

*The longer from the sound source, the less risk for unwanted distortion, but also more of the surrounding sounds and the room itself.

This being said, use your creativity and experiment. Where and how do you get the sound you are looking for? Closer to the mic, or from a distance? Does it sound better if you stand in front of the mic or behind it? What happens if you mix one take in front of the mic with one take behind the mic? (Yeah, I am serious about that question!)

There is no such thing as using the mic wrong (unless you want to use it as a hammer to drive nails into the wall). Does it sound good, you are using it right. If it doesn’t sound good, change the way you are using it until it sounds goood. And by sounds good I mean sounds good in your ears!

One more quick note regarding mics. Frequency. Different mics have different frequency curves, which means that they are meant for different purposes. I.e. a large membrane mic is better at capturing low frequencies than a small membrane mic. Look at the mic’s frequency curve to match your sound source, but don’t be afraid of experimenting with mic’s. Who knows, maybe it is a good idea to record song with a mic meant for picking up kick drums? (Yeah, I am serious about that question too!)

A very simplified explanation would be that any sound source created vibrations that propagates through the air. Even if it’s not completely true, closer to reality would be saying it causes compressions and thinning of the air. But let’s stick with vibrations for now, or even better, waves (as in soundwaves).

These waves moves like a chain reaction in the air.

When the waves reach our ear, it affects the tympanic membrane in the ear that causes a chain reaction in the mechanical part of the ear, eventually creating tiny electrical signals that our brain interprets as sound.

If we look at ourselves, our primary source of sound is the glottis in the throat. As we push air through our throat we control the glottis (or vocal cords) to vibrate and those vibrations propagates through our throat, up in the scull, getting acoustic resonance support from different chambers inside our skull and throat, finally formed by the shape and movements of our mouth and lips, exciting as spoken or sung words, or what ever we are aiming at to mimic.

Should we look at technology, the equivalent of the our glottis would be a loudspeakers. Their purpose is to get the air moving and recreate the waves (the sound) we want to listen to again. And on the other side, the equivalent of our ear would be the microphone that captures (records) the waves (the sound) we want to recreate and listen to again.

Now that we know that sound is waves that are transported through the air (or any other matter, but let’s not get into that now) there are two more things we need to understand about these waves. Amplitude and frequency.

If we look at the amplitude first. This is a way of determine the energy or the force of the wave. How hard it will hit the membrane (or ear). The harder the impact, the louder we experience the sound.

The scale the amplitude is measured in is decibel. Or deciBel, dB for short.

Now, there are a million (well, not a million, but many) different dB scales, depending if you measure the pressure of the sound or if you are into digital or analog recording, or looking at the actual electrical signal that the soundwaves generate. There are a lot of material to read about decibel, and you can start by reading the wiki article if you are interested to know more about it. It may look like greek, but it is the reason why I do not go in depth with it.

The lower the energy the sound, the lower dB value it will get (no matter what scale you use) and the more silent we experience the sound (as do the microphone).

The more energy in the sound, the higher dB value it will get (no matter what scale you use) and the louder we experience the sound (as do the microphone).

There are two aspects of amplitude that you need to be aware about. The first located in the silent part of the scale and the other in the loud part of the scale.

I start with the loud part of the scale.

Should the sound contain very much energy it is very loud, and may cause damage in our ears. Too much exposure may cause permanent damage to the ears.

As for microphones, even if the membrane not very often break due to too much energy in the sound, it still causes unwanted distortions in the signal, since the membrane can not move past it’s maximum range, even if there is more energy that keeps pushing on it.

This results in a clipped signal which you can see (if you use some kind of audio recording software) as a straight line on the soundwave.

When the energy is too much for the membrane to handle is very individual. You’ll need to test your microphone and find the boundaries it has.

General thumb rule is mic close to sound source gives a greater risk of unwanted distortion, too far from source gives too much of reflecting sound in the room, rather than the sound source itself (which may not always be unwanted).

On the other end of the scale there is when the sound is too silent. Every electronic (analog or digital) component has it’s own noise. The lower the better. And of course the lower the components self noise, the more expensive it gets (in general). If you want to read more about this, look for signal/noise ratio. The basic principle is that you want the sound from your sound source to be louder than the noise that the components emit. Much louder. A low quote between the signal and the noise means that the unwanted noise from the equipment is a big part of your wanted sound (the noise pollutes your signal).

Frequency. This is the last aspect of what sound is (that I will go into here).

The scale you measure frequency is Hertz (Hz) and what it measures is the distance between two cycles. Should you compare it to waves on the ocean, the frequency would be the distance from the top of one wave to the top of the next wave. The longer distance between the tops, the lower frequency. The shorter distance between the waves, the higher frequency. (Same example with amplitude would measure the height of the waves, the higher a wave, the more energy, giving a high dB, the lower the wave, less energy, giving a low dB.)

A low frequency gives a low (dark) sound and a high frequency gives a high (light) sound. If you sit in front of a piano, you have darker (lower) tones to your left. They have lower frequency. And to your right you have lighter (higher) tones. They have higher frequency.

The human ear can catch frequencies from 20 Hz (the lowest) and 20 000 Hz (often written as 20 kHz as in 20 kilo (1000) Herz). As we age, most of us lose the highest frequencies.

Different sound sources emit noise in different spectra of the frequency range. I.e. the human voice has a span of 200 Hz to 3000 Hz (3kHz) – roughly, a violin that has about the same range as here a cello goes from 65 Hz to 1050 Hz, and a piccolo flute around 550 Hz to 4000 Hz (4 kHz).

A strange thing about most sound sources is that you have the pitch of the sound (the precise vibration it causes) but it is not only that frequency of that tone you hear, the sound is built up by overtones and subharmonics in different combinations, which adds up to the whole character of the sound.