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During the mixing step, you will balance instruments, place them in the sound space, set their volumes, modify their frequencies, fix or hide the flaws. The purpose is to define each instrument's location, make them separated from one another and create a song that is pleasant to listen to.


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Dynamics is the interval between the lowest and the highest volumes in a sound. OK, fine, but we are doing Rock music, who cares about low volume? We want to kick ass! Drums must beat, guitars must yell, the bass must make the walls shake and the vocals must be loud! Pump up the volume! Wooo... calm down!

If everything is loud, then nothing is

Yes, that's logical. If everything is loud all the time, nothing stands out. But it's the contrasts that bring both subtleties and raging moments to light. How do you expect to surprise your listeners with a sound hurricane if you've already given them thunder and lightning without a break? You have to spare some quiet moments before the storm. Let's compare this with drawing. A black dot on a white background will stand out more easily than the very same black dot on a dark-gray background.
Black dot
Obvious? Yes, but many musicians just care for that "powerful sound" to such an extent that it eventually becomes tiring to listen to.

You can increase the contrasts through the very writing of the song (guitar/vocal bars, followed by a passage with 3 guitars, 1 distorted bass, big drums...), or through interpretation (the musicians play louder, faster...), or through the use of effects (saturation, compression...).

Contrasts don't have to be huge, it all depends on what kind of emotion you wish to convey, on the music genre, on the topic of the song, on what you are looking for...

Nowadays, too many songs just look (and sound) like this:
Over compression
Do your ears a favor: don't do this
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EQ stands for equalization. It's a tool that allows you to set the level of the frequencies that make the sound. Basically, it's like a volume knob, but instead of changing the volume of a whole song, you lower or raise only the frequencies of your choice. Just like when you use a stereo amplifier for instance: you can find "bass", "medium" and "treble" knobs. They are a simple EQ that help you to tune the sound.

Well, you are going to apply this same principle to each of the instruments of your song.

The purpose of equalization is simple: changing the sound, so that each instrument primarily occupies the frequency range that defines it the best, without encroaching on the other instruments.

In other words, lower the level of the frequencies that are not necessary to an instrument without distorting it, and see to it that the various instruments main frequencies do not overlap. Each instrument will thus be alone in its own sound space without encroaching on the others, ensuring all instruments are clearly distinct from the rest.

The hard part is to do this without distorting the sound of the instruments, and also to see to it that instruments which have common frequencies by nature be well-marked.

You will deduce that the more instruments there are, the more complex mixing becomes.
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Music is usually mixed in stereo. Not always, but we have two ears, we use headsets or speakers that usually come in pairs, so it only makes sense to use left and right channels when mixing.

That's what panning is: acting on the width of a sound, from left to right.

On top of the EQing described above, we can also part instruments from one another by placing them in different spaces. If two guitars play simultaneously in your song, you may want to place one on the left and the other one on the right. This will help to distinguish them and it will also balance the sound.
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Compression affects the dynamic range of a sound signal.

It reduces the amplitude modulations of the sound, that is to say the gap between the lowest and the highest levels.
Kjaerhus Audio Classic compressor
I will use the above compressor as an example. All compressors have more or less the same basic functions, so it will be easy to reproduce the manipulations on alternative models.

Let's see how this works:

- Threshold: this defines the minimum level from which the compressor will start affecting the sound. 0 is the highest sound level.
- Ratio: this is the level of compression you are going to apply. A 1:1 ratio has no effect, whereas an infinite ratio will limit the sound to the threshold level. In this case, the compressor acts like a limiter and nothing will go higher than the threshold. The higher the ratio, the more compressed the sound will be, and the volume level will be reduced in proportion, until we use the output button.
- Knee: this knob sets how much the compression will affect the sound right before and after the threshold. The softer it is, the softer the transition will be. A hard knee will result in a more abrupt transition. It is rather difficult to hear the effect of this function.
- Attack: this knob indicates the time it takes for the compression to start acting once the threshold is reached. Set it short (0 ms) to compress the sound as soon as the threshold is reached, and set it longer to avoid compressing the attack of an instrument. For example, with a solo guitar, you may want to preserve the string attack and thus set the attack knob on 10 or 20 ms. It all depends on what you wish to do.
- Release: this knob indicates the time it takes for the compression to stop acting once the volume level goes back lower than the threshold. A long release allows for a less brutal change of gain.
- Output: it can also be called "Gain". This knob sets the output volume level after compression of the sound. It compensates for the loss of volume due to the compression.

So basically, compression takes all the sound energy there is between the threshold and the highest level of the original signal and crushes it in a smaller gap. Then you raise the whole signal with the output button, which eventually renders a denser, richer sound compared to the uncompressed signal. This can be subtle or obvious, depending on your settings and the original sound.
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Reverberation brings a little air to a recording that would otherwise sound hollow or confined. It is not necessary to add much of it, or it could drown the sound and give the impression that the recording took place in a cathedral, unless this is what you are looking to obtain. A lot of reverb can be nice at some specific moments in a song, on one specific instrument, but I doubt anyone would apply this effect at a high level on a whole song.

It is recommended to put some on vocals. This will add depth. The more you add, the more the sound will seem to move away and get drowned in the mix. On the contrary, the less there is, the more vocals will seem to be close to the listener. A nice little touch of reverb makes vocals sound richer, deeper and more beautiful. Just find the right level.
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Delay is a sound effect that allows to... delay an audio signal and have it repeated. It's very much like an echo: shout and you will hear your voice first, then the echo of your voice repeated one or several times.

This effect is somewhat related to the reverb effect, with one major difference: delay repeats at regular intervals without generating a continuous surrounding sound. You can use it instead of the reverb to avoid overcrowding the mix.

As a general rule, set this effect by basing it on the song tempo. Choose whichever interval you wish to hear the echoes. Here is a chart to help you:
Delay by tempo
Click on the chart to download the corresponding Excel file ("Delay by Tempo & Instrument Frequencies.xls" - 64 KB)
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By changing not only the volume, but also the frequencies or some effects like reverb, you can give the impression that a sound is more or less near the listener. This adds depth to a recording.

In everyday life, how do you recognize a distant sound?

Its volume is low. At least relatively. A motorbike engine roaring in the distance will be much less loud than if it were 2 metres away from you. Thus, the first thing to do is to lower the volume of the instrument you want to place in the background.

Distant sounds don't have much bass frequencies. Bass frequencies require a lot of energy to move. Unless you hear a very powerful sound (like an explosion), the bass tone won't get to you. So you may cut low frequencies with a high-pass filter or an EQ.

On the other end of the sound spectrum, high frequencies of a distant sound are also fainted because they are absorbed by obstacles as they're moving. So you also need to lower these high frequencies.

Distant sounds are indistinct, unintelligible. Human voice, for example: if you hear someone speaking in the distance, you will of course recognize that this is a human voice (yes, your brain is that smart), you will probably know whether it belongs to a man or a woman, but you will not understand a word. You are going to have to reproduce this lack of clarity to simulate distance. A little reverb, a little chorus or some phaser touch will do. Give it a try and see what suits your music best.

You can't always tell exactly where the faint sound you're hearing is coming from. So pan the background instruments in the middle position and leave the sides to foreground instruments.

When listening to a music, a listener will mainly focus on changing elements. The use of a repeating, short musical pattern, treated as described above, will be easily forgotten and place itself in the background. The purpose of a background instrument is to add some fullness to the song, without being so interesting that it draws attention. It should be kept simple to have it blend into the song without being noticed too much.
And of course, on the contrary, foreground instruments will be the ones bringing volume, lowest and highest frequencies, melodies, and elements of surprise.
Example of a voice that gradually goes away from the listener
(video: click to play)

Flash video player for the Home Studio tutorial

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When adding effects, check the result by activating and deactivating them as the song (or the instrument) plays. It will help you to see whether an effect is too present, or not enough, whether it is necessary, whether it is too agressive, etc. You may like heavy reverb, but too much of it just drowns it all and makes the song inaudible. Control is essential. Too little is useless, you might as well remove the effect completely, whereas too much will make everything weighty and will surely waste the whole piece.

An effect often turns out to be... effective if you miss it when it's gone, even though you were not certain to hear it when it was on.
Recording - Previous | Next - Mixing in practice
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(Leave a message)

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03/03/2018, 14h10


Je viens de prendre connaissance de ton site, c'est pas mal du tout, je voulais me mettre aux IR depuis un bout de temps, et là j'aurais quelques questions.

J'ai énormément de matériel de musique et j'aimerais tout centraliser dans un Helix.
Pour ça il faudrait que je puisse séparer chaque élément de mon rack et de mes pédales de manière indépendante, est-ce possible?
J'imagine que oui, il suffit simplement d'avoir une différence entre l'entrée et la sortie des appareils, comme les tutos que tu proposes.

Par contre j'aimerais pouvoir faire une IR de mon ampli de puissance et là je bloque un peu.
En effet, si je veux sortir le son de mon ampli de puissance il faut obligatoirement que je passe par un baffle et là ça va colorer mon son. Existe-t-il un moyen de récupérer juste un fichier IR de mon ampli de puissance un Mesa Boogie fifty-fifty.
Et pouvoir aussi récupérer un IR de mon baffle seul.

En fait ce que vous je voudrais faire dans le Helix c'est pouvoir charger plusieurs IR de chaque élément de mon rack, un pour le preamp, un pour l'ampli de puissance, un pour le baffle...

Si tu as des idées pour faire quelque chose, je serais ravi d'avoir tes conseils, merci d'avance

Mouss Kerr
01/31/2018, 19h08

Félicitations pour ton site. J'essaie de télécharger les simulateurs d'amplis gratuits sans succès. Je m'y prends certainement mal car je débute. Mon matériel : PC I5 64 bits. DAW Reason 9.5. J'ai créé un dossier VST ajouté dans préférences ... Avancé. J'y ai transféré le fichier DLL mais impossible de l'ouvrir sous Reason. Merci par avance

01/13/2018, 19h29

Solid work across the board. Found myself here because of your amp heads and then never left!

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Glad you find the place nice and cosy. You're welcome here!

11/11/2017, 00h35

Merci pour ce tuto !
Le seul bémol que j'apporterais, c'est la dynamique finale : ne pas céder au chantage mais rester à -14 / -12dB RMS si on exporte sur CD. En revanche, -12 / -11 dB pour un mp3 192kbps paraît acceptable.

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Oui, dans la mesure du possible, je confirme que c'est bien de rester à des niveaux inférieurs à -12dB. Il faut tendre vers ça le plus possible. Mais en arrivant à -10dB dans mon exemple, je fais malgré tout déjà mieux que la plupart des albums actuels.
Quant à l'export CD, il est quand même de plus en plus rare. La plupart des gens écoutent désormais leur musique depuis leur téléphone portable, et bien souvent en mp3. Et honnêtement, la différence entre un fichier wav et un fichier mp3 à 320 kbps est indiscernable pour 99 % des gens. Même à 192 kbps, rares sont ceux qui sont capables d'entendre la différence, surtout quand c'est écouté au travers d'écouteurs ou d'enceintes bas de gamme ou moyenne gamme.

10/20/2017, 09h00

Bonjour et merci pour votre site sur la MAO que je ne connaissais pas il y a encore quelques mois.
Je suis sur PC Windows 10 système 64 bits, séquenceur Reaper. J'aurais voulu savoir comment récupérer les réglages des presets (.fxp). J'ai téléchargé le preset exemple pour les amplis LePou, je l'ai enregistré dans mon répertoire VST, et là je ne sais pas comment récupérer le réglage. Autre petite question, dans le preset, y a-t'il la définition et le réglage des impulsions ? Merci d'avance et encore merci pour toute l'aide que fournit votre site.

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