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To obtain various sound effects, edit, control or polish your sound, you will need to use plugins. There are several plugin file formats, the most common on PC being the VST format, invented and standardized by Steinberg (the publisher of Cubase). There is a multitude of effects, from non specialized to highly specialized ones. Here's an overlook at some of them.

Compressor and Limiter
Reverb and Delay
Chorus, Flanger, Phaser
Other plugins

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Compression allows you to boost the sound. Its purpose is to reduce or make denser the dynamics of a sound. Dynamics is the difference between the lowest and the highest level of a sound.

For instance, when you watch a film on television, whether the actor is murmuring or talking normally, you will hear more or less the same volume: the sound was compressed in order to avoid big differences between the weakest and the loudest sounds.

Compression should be used with some moderation in order to avoid unpleasant side effects: a sound that is too compressed will seem to be "pumping", because the compressor will be permanently compensating the differences in volume of the signal. Some instruments are easier to compress thant other and compression will be used differently, depending on the expected result or the musical genre.

Compression will be used much more in Dance music that in Classical music. In the latter, it is common to hear very soft and quiet passages, abruptly alternating with violent passages, which creates striking contrasts. In modern music, compressed sounds (when done right) sound more flattering to the ear. The purpose of compression is also to make your song sound as loud as other songs, which leads to various degrees af abuse, harmful for the music... and your ears.

Free compressor to download

Classic compressor (, 357 KB) by Kjaerhus Audio: a simple compressor that has everything a compressor must have. A threshold button, which defines the volume from which the compressor starts compressing, a ratio button going from 1:1 (no compression) to 'infinite' (at this level, the compressor becomes a limiter), a knee button to set whether the compression will start as soon as the threshold is reached or if it will act smoother, an attack button to define how fast the compression acts once the threshold is reached, a release button to define how long it takes for the compression to stop once the sounds goes down below the threshold level, and a level button to define how much gain you want to compensate for the loss of volume of the compression. The more you compress, the more the volume will be reduced, and you make up for it bu using the level button, and the final sound will seem louder than the sound without compression, even though the volume is the same.

Kjaerhus Audio Classic compressor
A limiter is another sort of compressor, with a slightly different purpose. It is still about boosting the sound signal, only this time, you are to limit the volume at a specified level. You can decide that an instrument will never exceed a given volume, whatever happens. This can also be applied to an entire song: you decide that the final sound level will not exceed a specified level.

To do that, you have to define a threshold (minimum volume level) from which the sound will be boosted, and a maximum volume level. Like for compression, the lower the threshold, the more compressed the sound will be, and if the setting is wrong, the sound will become unpleasant to the ear. You then have to look for the settings that do the job without spoiling the sound.

Free limiters to download

Classic master limiter (, 357 KB) by Kjaerhus Audio: A simple threshold button that defines the level from which the compression will start. As there is no ceiling button, you define the ceiling volume by setting the volume of the track where the limiter is placed. If you want a ceiling control, try the W1 plugin below.

Kjaerhus Audio Classic master limiter

W1 Limiter (, 162 Ko) by George Yohng, without a graphic interface: it's a free clone of the retail L1 Ultramaximizer limiter from Waves. Very simple to use, you define a threshold, a ceiling and a release time.

The zip file includes both 32 and 64-bit versions. There is a version with a graphic interface (shown below and available on this page), but it makes my DAW (Cakewalk Sonar) and my video editing software (Sony Vegas Pro) crash. So I uninstalled it and kept the very stable version without a graphic interface.

Official website of the developer of this limiter: here.

W1 limiter     W1 limiter     W1 limiter
With a graphic interface, then without a graphic interface as it appears in Sony Vegas Pro 12, and in Sonar Producer.

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An equalizer (EQ), is a tool that is used for increasing or decreasing the sound level of specific wavebands.

Basically, it is like a volume knob, but instead of altering the volume of a whole song, you may choose to only increase or decrease the volume of the frequencies you want. Take a hifi amplifier for instance: you have "bass", "medium" and "treble" knobs. That's a simple EQ, that allows you to adjust these parameters to your taste.

Graphic EQs let you adjust fixed wavebands, while parametric EQs let you edit the frequencies of your choice, as well as their width. Parametric equalizers are therefore more acurate but also more difficult to use.
Cockos ReaEQ
(, 771 KB - This is an install file which contains several plugins)

Multiband parametric equalizer. You can add as many bands as you wish. I tried to find the limit, but stopped clicking on the "add" button when I reached 275 bands... I got bored of it! Needless to say that there is no use in having that many bands, but for those who find it convenient to have 10, 12 or even 20 bands, it is possible.

Cockos ReaEQ 0.75
Kjaerhus Audio Classic EQ
(, 387 KB)

This is a 7-band graphic EQ from Kjaerhus Audio. You can independently set the left and right channels, or link them.
The frequency bands are 20 Hz, 63 Hz, 200 Hz, 630 Hz, 2 KHz, 6.3 KHz and 20 KHz. This is a very effective EQ, with clearly audible changes to the sound. You can set the frequency bands by steps of 1 dB, from -10 dB to +10 dB.
Although it's rather old (it was made in 2005), it remains an excellent choice among free EQs, and can either be used for individual tracks or for the entire mix on a master bus.

Kjaerhus Audio Classic EQ

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A reverberation, or reverb, is created when a sound is produced in an enclosed space causing a large number of echoes to build up and then slowly decay as the sound is absorbed by the walls and air. It is possible to reproduce spaces of various sizes which of course will sound differently. Reverberation in a cathedral does not sound similar to that obtained in a bathroom.

A delay is similar to an echo. The sound will repeat at regular intervals and eventually fade out. With a delay, you can set the number of repetitions, the interval between two repetitions and the duration of the fading out. In general, you tie the repetitions to the tempo, on beat or offbeat.
You can use the delay instead of a reverberation when you want to enrich the sound without filling up the mix, especially if the mix is already full. Roughly, the reverb will produce a continuous sound, whereas the delay will repeat at regular intervals and turn out to be less invasive.
Classic Reverb by Kjaerhus Audio: this little plugin may not be the best reverb in the world, but it still gives nice results. (, 367 KB)

Kjaerhus Audio Classic reverb

Classic Delay by Kjaerhus Audio (, 390 KB)

Kjaerhus Audio Classic delay

Sir Audio Tools Impulse LoaderSIR Convolution 1 (, 353 KB)

This excellent free reverb makes use of convolution. What is convolution? In the field of audio signal processing, convolution will simulate the reverberation of any actual or imaginary venue. It is based on the mathematical convolution operation and uses sound samples called impulses.

You can use pre-recorded impulses (see below), or create your own.

In SIR 1, you load the impulse of your choice and apply it to the track you wish in the DAW (guitar, bass, vocals, drums, piano...) to simulate your instrument playing in the chosen location. The reverb will be more or less convincing depending on the quality of the impulse, but it can give excellent results.

You may of course use any sound file as an impulse, which can give surprising results. Experiment. Caution: do not use audio files that are too big, because the longer the file, the more computation will be complex and the processor load might be increasing dramatically.

Sir Audio Tools Sir Convolution 1

Impulses to download

More than 160 impulses from various types of real or virtual venues, to be used in an impulse loader such as SIR:
Impulses for reverb (, 36.9 MB)

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A chorus effect occurs when individual sounds with roughly the same timbre and nearly (but never exactly) the same pitch converge and are perceived as one. It creates a thicker and fuller sound.

Chorus Kjaerhus Audio Classic Chorus (, 364 KB)

Kjaerhus Audio Classic chorus

Flanging occurs when two identical signals are mixed together, but with one signal time-delayed by a small and gradually changing amount, usually smaller than 20 milliseconds. This produces a swept comb filter effect: peaks and notches are produced in the resultant frequency spectrum, related to each other in a linear harmonic series. Varying the time delay causes these to sweep up and down the frequency spectrum.

Flanger Kjaerhus Audio Classic Flanger (, 362 KB)

Kjaerhus Audio Classic flanger

A phaser is an audio signal processing technique used to filter a signal by creating a series of peaks and troughs in the frequency spectrum. The position of the peaks and troughs is typically modulated so that they vary over time, creating a sweeping effect. For this purpose, phasers usually include a low frequency oscillator.

Kjaerhus Audio Classic Phaser (, 369 KB)

Kjaerhus Audio Classic phaser

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Here are some VSTs that produce unusual effects, and also tools that may be convenient for you, such as a guitar tuner or a spectrum analyser.
GTune: A chromatic tuner for guitar and bass. Free and convenient (, 197 KB)

TT Dynamic Range Meter: A tool that measures the dynamic of a track or a full song (, 157 KB)

TT Dynamic Range Meter
SPAN by Voxengo: A good, free spectrum analyser, to download directly from the developer's website.

Voxengo Span
Classic Auto-Filter de Kjaerhus Audio: Modeled after an analog filter to create some unusual effects (, 381 KB)

Kjaerhus Audio Classic auto-filter

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(Leave a message)

Message page # 1 2 3 4 5 6 7 8 9 10 11 12 13 14 15 16 17 18 19 20 21 22 23 24 25 26 27 28 29 30 31

01/18/2020, 18h08

Whatever happened to the Cranked AC plugin? I've been looking all over for it but can't find it anywhere.

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Sorry, I never heard of this plugin. If it's an old plugin, chances are it's also a 32-bit plugin, which may not work properly on modern systems, but anyway I couldn't find it either.

01/12/2020, 23h06

So the chain goes:

DAW > Audio Interface Out > Amp > Speaker > Mic > DAW

This is correct based on my understanding from what I've read, and the few videos I've watch on creating IRs. My question, then, is when I plug into the Amp I've seen people say plug your Interface out into the FX return, but you say the guitar cable jack. What is the purpose in doing one or the other?

Side questions:

What channel should my amp be on? I'm assuming the clean channel.

What should my Amp settings be (EQ, Gain, Channel Volume, Presence, Master Volume)? I can't find a clear answer anywhere.

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About plugging into the FX return or the guitar jack, I don’t know. Actually, the amps I’ve used myself to make IRs don’t have any FX return, so I didn’t have a choice and had to plug into the guitar jack. I guess there’s no harm trying both (not at the same time!) and comparing if you have that possibility. Chances are there’s not much of a difference, but again, I may be wrong as I have not tried this myself.
About the choice of a channel, and the settings: the channel doesn’t actually matter. You’re not capturing the amp sound, but the speaker sound.
From what I’ve experienced, the EQ and Presence should be neutral, the gain/saturation should not be engaged (or set to a level where no distorsion can be heard). As for the volume, set it to a level that’s high enough for your microphone to be able to pick up a good signal (no need to record higher than -6 dB, by the way, give your signal a bit of headroom).
But you should also be careful not to set it too loud to protect your own ears. It doesn’t need to be pushed too high. I think a level high enough to cover your own conversational voice should be enough. I tried various volume levels, and it did not affect the results notably. I did not get better results with very high levels than with normal, humanely bearable levels. Don’t set it too low, though, because it’s better if your speaker does move some air.

Experiment, try different amp settings and see whether that changes the results.


10/20/2019, 17h06

Hey, I downloaded the plug-in and extracted it. Then put it in the plugin folder but it is not working. C:|Program Files|Common Files|Avid|Audio|Plug-Ins. Would this be the right steps? Please let me know thanks!

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As you explained it to me by e-mail, you were using Pro Tools First, which doesn't support third party plugins. The solution is then to either upgrade to a paid version of Pro Tools, or use another free DAW, such as Cakewalk by Bandlab (Windows only), or use Reaper, which is not free, but can be used freely without constraints. These DAWs do support third party plugins.


08/26/2019, 11h06

Tout d'abord bravo pour ce site.
Je suis débutant et rencontre quelques soucis.
J'ai un PC Windows 10 (64 bits, 8 Go de RAM) avec carte son intégrée en 5.1, driver realteck, et quand je lance un programme de simu type Amplitube 4, il y a un son horrible qui sort, est-ce normal ? Y a-t-il un moyen d'y remédier ?
J'ai essayé également avec Bandlab comme séquenceur mais je ne sais pas comment intégrer le cab et le simulateur.
Merci d'avance

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Bonjour Dam40,
Le son horrible qui sort avec un logiciel de simulation n’est pas « normal », mais c’est peut-être dû au fait que vous utilisez la carte son intégrée de votre ordinateur. Ce type de carte n’est pas du tout adapté pour enregistrer et mixer de la musique.
Pour enregistrer de la guitare par exemple, il faut passer par la prise Jack de la guitare et les cartes son intégrées ne possèdent pas ce type de fiche. D’autre part, les drivers des cartes intégrées ne possèdent pas non plus l’impédance électrique compatible pour avoir un niveau de son correct en provenance de l’instrument, et d’autre part, même quand ça marche, elles induisent une latence, c’est-à-dire un délai entre le moment où l’on joue sur la guitare et le moment où le son est entendu sur l’ordinateur.

Pour remédier à ce problème, il faut acquérir une interface audio, un type de carte audio qui se présente sous la forme d’un boîtier externe connecté à l’ordinateur par la prise USB (le plus souvent, même s’il existe d’autres types de connexions). Ces interfaces sont fournies avec un driver spécifique qui permet de gérer le son grâce au protocole ASIO. Ce protocole est standard et permet d’obtenir de faibles latences pour pouvoir jouer de la guitare et entendre le son, avec ou sans effets, sans délai gênant.


08/16/2019, 04h18

Bonjour !

J'ai testé la quasi-totalité des simulateurs présents ici pour une raison : impossible d'ouvrir un fichier DLL !
Mon PC me demande d'associer l'ouverture des DLL à un logiciel mais je n'ai rien de spécial qui va avec...

J'ai eu ce souci, j'avais formaté mon PC vu que je ne l'avais pas fait depuis des années (1,65 To de données à re-télécharger)
Et là encore le même souci, je teste donc sur 6 PC différents et tous ont ce souci... Je suppose donc qu'il faut un logiciel spécial mais rien n'est mentionné, tu pourrais m'aider ? Merci d'avance !

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Bonjour Blastrax,
Tous les simulateurs d’ampli gratuits sous forme de fichiers DLL sont des « plugins », et non pas des logiciels autonomes.
Je l’explique ici.

Ces fichiers de plugins ne s’installent pas, il faut simplement les recopier dans un répertoire du disque dur. À noter aussi que les simulateurs d’ampli gratuits ne simulent que la tête d’un ampli. Pour avoir également une simulation du haut-parleur, un autre plugin qu’on appelle « chargeur d’impulsions », dans laquelle on charge des « réponses impulsionnelles », ou IR (impulse responses, en anglais). Les IR sont des petits fichiers audio qui reproduisent le son d’un vrai haut-parleur. On peut trouver des IR reproduisant le son des amplis Fender, Vox, Marshall, Orange, Mesa Boogie, etc. Il en existe des gratuites et des payantes.


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