Before anything else, we need to prepare the recording session. We are not going to look for a melody, nor improvise or rehearse... we are going to properly record a song that we have already written. To avoid doing the same things over again each time I wish to record a song, I prepared a blank template which contains all the tracks and buses I need. I may naturally add or delete some elements if the template is not appropriate for the current project.
What does my template look like?
- Rhythm guitar tracks (from 2 to 8 depending on the project)
- Solo guitar tracks (usually 2 tracks to make the sound thicker)
- Two bass tracks (one with the direct raw sound, and one with an amp simulation)
- Lead vocal tracks (usually one or two tracks, depends if I record it twice or not)
- Background vocals tracks (if the project demands it)
- Keyboard tracks (same thing, the number of virtual instruments will depend on the project, can be none, can be 5 or 6...)
Then we have drum tracks. There is one track per drum element. They are automatically created when I insert my virtual drum plugin:
- Kick drum
- Snare drum
- Low tom
- Medium tom
- High tom
- Crash cymbal
- Ride cymbal
- Splash cymbal
- Overhead microphone
- Room Ambiance microphone
- Piezzo microphone
- One MIDI track on which the drum score will be placed.
Bus-wise, I have one group for the guitars, one for the bass, one for the vocals, one for the drums, each of them is redirected towards the Master bus, which goes out throuh my studio monitors. It looks like this:
Recording an acoustic real drum kit is far from being easy, even for professional sound engineers. It a time-consuming process, it's frequent to spend several hours placing the microphones around the drum elements before you can actually record. But we are in a home studio, and we will have to deal with a software drum kit, based on midi files...
First of all, why start with the drums? The answer is simple, we will use drums as a metronome. The drums sound will guide us and help us follow the tempo. This will allow for an even recording and the song will not speed up or slow down unintentionally. Of course, variations can be interesting and bring some life to an otherwise mechanical tempo, but let's consider that a studio session seeks recording perfection, even though it's only a home studio.
I usually have no idea what my final drum track will sound like. Actually, I only adjust it when the rest of the song is finished. But I still need its metronome function to record all other instruments. Thus, I create a drum track which repeats itself over and over again, and I try to have this loop match what I am about to play (no punk rhythm to record a ballad). For instance, I'll use one of these patterns:
Let's not forget we are recording in a home studio, in an appartment and it is simply impossible to play with a good old 100-watt tube amplifier, without the neigbors calling the police. So we are going to have to record the guitars and the bass directly through the audio interface. No real amplifier, no microphones involved. The latter solution would be preferable, but on of the benefits of direct recording, coupled with amplifier simulators, is that you can always edit the sound later, without having to re-record. Just change the settings and you're done.
So now... bass or guitars first? There isn't one clear answer. Bass and drums are the foundation, the rhythm base of a song and everything else should rely on them. But other factors could also be taken into consideration: for instance, the person recording may be more comfortable with a guitar than a bass, and will rather play guitar first. Or maybe the song has a very important bass riff that compels you to record it first. In any case, you are the one who can decide. If you are uncertain, then the drum / bass duo is a safe bet. If this is in place, then the rest can easily be added.
Guitar or bass, the recording process will be the same. Plug your guitar into the pre-amp, the pre-amp is connected to the audio interface (or plug your guitar directly into the audio interface if using the interface's pre-amp), and set the recording level. This is very important! Before recording anything, check that you are not going to go beyond the maximum level (0 dB, zero decibel). In a home studio, you won't have a sound engineer besides you to make adjustments on the fly, while you are playing. You are the one to take precautions. How can you do that? Simple: try and adjust, it doesn't take long and will prevent you from making a perfect take, then realize the levels were too low or too high, forcing you to do it all over again.
Have a try: for a rhythm guitar for example, play the loudest parts and set the preamp and audio interface volume levels in such way that when you play the loudest, the recording level doesn't go beyond -6 dB. The absolute maximum that you should not reach or go beyond is zero dB. If you play in your try the same way you play during the actual recording, then you can be certain the recording level will be correct. If your average level is between -9 db and -6 dB, then your level is sufficient and you have a margin of error before clipping.
Clipping is the term used to indicate that you reach or go beyond 0 dB. Clipping is your enemy :-)
I prefer to record them last but there are no rules. If you prefer to record them first, then do so.
To record vocals, make sure the place is quiet, shut the door, tell the people who live with you to be quiet, and do not record while your neighbor is drilling holes through his kitchen walls! Also, turn off your monitors and use a headset instead to avoid recording the playback with your microphone.
Condenser or dynamic microphones?
Dynamic microphones are solid, they don' need a power source, they can take heavy acoustic pressure (like a kick drum or a saxophone) and they are not too expensive. They are also less sensitive to surrounding noises than condenser microphones. The cons are they lack clarity in the high range, which renders takes less clear and defined than with condenser microphones. They can be used with Jack or XLR plugs.
Condenser microphones are much more responsive and accurate. Their high sensitivity is double-edged, because they will capture any noise when recording. The fans of your PC are noisy? Chances are this noise will be recorded. Sound comes out of your headset? It will be recorded by your condenser microphone. Children are loudly playing outside? You might get that too. However, some condenser microphones are called "cardioid", or "hyper cardioid", and they only record what comes from a specific direction, ignoring (more or less) other sound sources from other directions. On the contrary, omnidirectional microphones record what comes from anywhere. Not ideal for a home studio. Condenser microphones are also more fragile (don't knock them) and must be powered through a "phantom power", whose standard is 48 volts. This kind of power is either present on your audio interface and can be turned on and off with a button, or it will require the use of an external phantom power source that you will then connect to your audio interface. You have to use 3-pin XLR plugs that carry the phantom power current. Finally, condenser microphones are usually rather expensive, some of them cost several thousand euros (or dollars, or pounds), but only professional studios or rich amateurs can afford those. On the plus side, the sound you get with a condenser microphone will have the best quality.
Be cautious though, a good dynamic microphone is worth better than a bad condenser microphone. No big secret here, for microphones like for anything else, very low prices are rarely synonymous with good quality.
A few known and renowned microphone brands: AKG, Milab, Neumann, Rode, Sennheiser, Shure...
Jack plug (left) and XLR (right)
No need to go on and on forever, recording is rather easy. As long as you pay attention to your recording levels and take care over your takes, you should get a satisfying result, good enough to finalize the song
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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.
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?
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.
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.
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.
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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.
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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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.