Labels

BASS (39) COMPRESSION (28) DRUMS (36) EFFECTS (41) EQUALIZATION (24) GUITAR (81) HOME RECORDING (58) INTERVIEWS (17) LIVE (9) MASTERING (39) MIDI (15) MIXING (124) REVIEWS (64) SAMPLES (9) SONGWRITING (8) VOCALS (24)

Saturday, December 24, 2011

HOW TO RECORD AND MIX THE ACOUSTIC GUITAR (free Vst Plugins included)


Hello everybody and welcome to this new tutorial! Today we're going to talk about how to record an acoustic guitar, an instrument that needs to be reproduced as natural sounding as possible to be good, and that is the reason why it cannot be digitally recreated via synth in a credible way.
First off let's say that there are two ways to record an acoustic guitar: using a microphone, or (if the guitar has a pickup), using the jack output of the guitar itself.

- Using the Microphone: first off you need a microphone suited to record acoustic instruments, such as the Shure Beta57, but there are many others very good and probably cheaper, for example made by Sennheiser. Condenser microphones may be even better, but they need an acoustically treated room in order to play at their best, so check the environment carefully before choosing.
Now we need to find the "sweet spot" of the guitar where to point the microphone, keeping in mind those rules: if you point the microphone toward the bottom of the guitar or behind it, the sound will result very deep and full of bass frequencies, same is if you point toward the sound hole: you will get a bassy sound that captures the internal resonance of the guitar.
The more you move toward the neck, the more you'll capture the strings sound, so the higher you will go in the neck, the more the sound will be brilliant.
Obviously you'll have to hear and decide for yourself, but I would consider a good spot the area near the 12th fret: this is where the neck joins the body and is usually regarded as the best place to record the acoustic guitar, since it is where we capture a full and even frequency response from the instrument; not too bassy and not too brilliant. If you need some extra string sound, you could also place another mic at the first fret to capture the nuances of the strings.

A typical miking setup for the acoustic guitar is with two Condenser microphones: one pointing the 12th fret, the other one pointing the bottom of the guitar, always on the front side as the other one: the one pointing to the bottom of the guitar will catch some low end that will later be mixed with the sound acquired with the other mike. At these two microphones can be added a third condenser one, set about one or two mt. away from the player in order to catch some of the room Reverb.
Click Here for a dedicated article about how to microphone an Amplifier.

- Using the pickup: Many acoustic guitars nowadays, even the cheaper ones, have built-in a piezo pickup, and a battery-powered preamp, sometimes even with an equalization section too!
To record the sample in the video for this article, I have used an Ibanez V72 Cent, a cheap acoustic guitar with a built-in preamp, a volume, bass and treble control, and a built-in tuner too.
So first off I have found a good compromise in the built-in equalization, slightly cutting the treble and boosting the lows to give it a more "round" tone, then I went with the jack straight to the input of my audio interface.

- Once you have the signal on your D.A.W., is time to process it. You can use any plugin you want, usually any D.A.W. has some processing plugin bundled, but if you need some, I suggest you the Reaplug Suite, which is freeware and it sounds great.
First off I'd suggest to Equalize, but be very gentle: we need to preserve the naturality and the tonal richness of the acoustic instrument, so there's no need to overprocess the signal: we may just use a high pass filter to take away the lows we don't need, such as the frequencies below 50hz (sometimes, if we're on a more dense mix, it's best to take out everything below 100hz, or even 2/300hz, if we want just the guitar strum to pop out of the mix), and gently subtract a little (-3db) in the areas around 100-300 hz and 1-3 khz, if the sound is a little boomy or needs to be more open and transparent. Boosting between 5khz and 10khz will add sparkle, cutting between 1khz and 3khz will reduce harshness.
For the solo parts, we can instead boost some decibel (e.g. 3), to the 5-7khz area, in order to add some presence.

Now it's time for Compression, which can be normal or multiband. For the normal compression, just set a ratio between 4:1 and 12:1, according to the dynamic range of the song (the more volume differencies between quiet and loud picking, the higher the ratio should be), short attack (around 20ms) and medium release times (around 0,5sec), and a threshold set to around -15db.
The Multiband Compression, instead, gives you the opportunity to choose the amount of compression to apply to the single frequency areas, which can be adjusted through the controls.
This may be useful, since it can be used as a halfway between a compressor and an equalizer, and can help to reduce volume and presence of single areas of your signal (for example the lows between 100 and 300hz) and boost others (between 5 and 7 khz), without changing the overall tone.

Once you are satisfied with your tone, you can add (if needed, especially if you have recorded directly from the jack output) a Harmonic Exciter to add some sparkle to the tone, or a Reverb, like Freeverb, set very low, just to give to your sound that little resonance that it needs to be realistic; we can use for example a Plate reverb to add vitality, with a decay time of between 2 and 3 seconds.

So the Chain is: Guitar->Equalization->Compression->Harmonic Exciter (if needed)->Reverb (If Needed)

Cheers everybody and Merry Christmas!!

Become fan of this blog on Facebook! Share it and contact us to collaborate!!

Saturday, December 17, 2011

THE PERFECT GUITAR EFFECT CHAIN ORDER


Hello!! Today we're going to talk about the guitar effects chain, focusing on the exact order of your stompboxes (or rack effects), from the guitar to the amplifier input.
First off if you see the image, I have divided the signal chain in three areas: the first is the Pre Gain Area, then we have the Gain Related Area, and finally the Post Gain Area.

- The Pre Gain Area:  this is where the signal chain begins, and where we should set the effects that needs to be applied to the signal before the Gain Related Area. 
Here is were we put utility processors like TunerNoise Gate, then we can apply filters like Wah Wah or Envelope Filters, next is the place for Dynamic Controllers, like a Compressor, and the "area" ends with the intelligent processors, such as Harmonizer or Pitch Shifter.

- The Gain Related Area: this area needs to be separated from the others because it changes dramatically the signal, so any effect that expand the sound, like delays, must be put after this area, if we want to avoid to distort the delay repetitions too. 
This is where we put our Overdrive, Distortion or Fuzz.

- The Post Gain Area: this is the last area of our guitar signal chain, and includes time-based modulation effects (chorus, flanger, phaser, tremolo and many others) and pure time-based modulations (such as delay and reverb).

Very Important: If instead of using gain related stompboxes you wish to use directly the overdrive channel of your guitar amp, the preamplifier of your amp itself becomes the Gain Related Area, so you need to move the Post Gain Area to the Fx Loop of your amplifier (via the "Send" output, set all the post gain effects you need, and then go back into the amplifier through the "Return" input). 
And that's it, your perfect guitar effects chain is ready!

So here is our chain by areas: GUITAR -> PRE GAIN AREA -> GAIN RELATED AREA -> POST GAIN AREA -> AMPLIFIER

Here is the effect order: GUITAR -> NOISE GATE -> WAH/FILTER -> COMPRESSOR/LIMITER -> HARMONIZER/PITCH SHIFTER -> OVERDRIVE / DISTORTION / FUZZ -> CHORUS / FLANGER / PHASER / TREMOLO -> REVERB / DELAY -> AMPLIFIER

A small note about EQUALIZATION: its position is variable. Someone likes to add it after the Wah. Someone puts it Before the Amplifier (at the end of the chain), someone puts it in the effects loop. Someone even combines two or more of these positions, for example it's suggestable to cut before the Compressor and boost after, so feel free to experiment!!

Become fan of this blog on Facebook! Share it and contact us to collaborate!!


Sunday, December 11, 2011

HOW TO: THE BASIC MASTERING CHAIN (free Vst Plugins included) PART 1/2




Tutorial version 2.5, august 2016

Hello! Today we're gonna see how to master a song, trying to analize the single steps in order to give to your (already mixed) track the final boost it needs to be loud and sparkling, enough to be compared to the commercial tracks.
Let's start from the assumption that the perfect mix just needs to have its volume raised to 0db on the mastering phase, but more than often other processes are needed in order to achieve a good final result.
Obviously there are many ways to build a mastering chain, as always I'm gonna explain you the basics and suggest some free plugins, then you will adapt these ideas to your project and your plugins.

- First off, load the track containing your mix on a stereo track, on a new project, making sure it isn't too loud (I would recommend to keep the mix it around -10db, to give you enough headroom to work in the mastering phase), in order to avoid clipping; it is also importanto to make sure the track is cleaned of all hiss, pop, crackle and noise before stating to master it, or the problems will get worse.


- The first plugin to add to our buss is an equalizer (for example ReaEq), if you feel there's some frequency to correct; that way you'll be able to shape (very lightly) some general frequencies that doesn't satisfy you completely after the mixing phase, usually eq is used in mastering in order to scoop very lightly the mids (more or less -1db around 300hz), and boost gently lows and highs (+1db around 50hz and 5000hz), but actually the best way to eq during mastering is by using a Mid/Side equalizer, the way described here. After these adjustments, using the same eq plugin, create a high pass filter to remove all of the frequencies under the 30hz, in order to clean the mix of the almost inaudible and useless frequencies, like rumble and breath. 

If we feel that the final sound is still a little bit thin and dry, and we didn't use a lot of Reverb during the Mixing phase, we can also try adding some Mastering Reverb. It consists into adding a Reverb with very low settings, a regular "Room Size", a low Wet/Dry ratio and a Low Pass and High Pass filter between 100 and around 2000hz. It's very important to not overdo, though, since this can really screw up everything :) The position in the Chain is Typically between the Equalizer and the Compressor, but it can be moved after the Compressor if we feel that the Comp is making the effect too strong.

- Now we must focus on the different areas of the mix, and try to point out if there are certain parts (for example, the drum snare) which are too low, or others (for example, the cymbals) too loud, and try to correct the problem; you can do it with the eq, like on the last point, or use this other method, that is less invasive and "coloring", which is the multiband compression (Click Here for an in-depht article on the topic). 
There are many multiband compressors around, and there are some bundled in almost every DAW on the market, but if you need a freeware one, here is a list where you can choose from. Using a multiband compressor lets you choose graphically which part of the curve (i.e. only the highs on a certain frequency) to compress, leaving the other frequencies of the mix unprocessed, and it's a very useful tool to make aimed corrections.

- Now we can add to our mastering buss a Tape Saturation plugin, that produces a slight Compression and Saturation without squashing too much the overall sound, but this depends mainly by the song and by how much compression you have already applied on the single instruments and on the Mixing Buss. Alternatively we can use an Harmonic Exciter, in order to give some sparkle to the high frequencies, and some thump to the lows.

CLICK HERE TO READ PART 2/2 OF THIS TUTORIAL!!

Become fan of this blog on Facebook! Share it and contact us to collaborate!!

Saturday, December 3, 2011

HOW TO MIX A GOOD ROCK / METAL BASS (free Vst Plugins included)




Hi everyone! Today we're gonna talk about how to get a good rock-metal bass sound, and how to find its place on the mix, which is one of the hardest things to get; often, in facts, bass tends to disappear, covered by the lower frequencies of the guitars, or to be too loud, making the mix muddy. Our aim should be to find a place where to place where to set the bass, avoiding to make it fight with the other instruments, and if we'll succeed, we will obtain a much more glued and punchy mix.

Let's get started!

First off you obviously need a decent bass, and an audio interface of any kind, in order to get the signal to your daw. Don't use "the mic in" of the integrated sound card that comes in bundle with the pc, because the sound will be awful :)
You can help yourself by tweaking with the bass knobs until you find a good starting sound, then you can record your track, making sure that the signal isn't too high nor too low. If it's too high, it will distort/cut frequencies, if it's too low, you will have to raise the volume later, raising the noise too, and obtaining a high-volume weak sound.
We can use a Line recorded bass, a microphoned bass amplifier or a Virtual Bass (Click here for a dedicated article).

Now there are two ways to proceed:

1. Single track: first, to give punch to the sound, use a bass overdrive-preamp simulator, like the TS-B.O.D. which is a great freeware vst that simulates the legendary Sansamp Bass Driver, then you can add a Compressor, with a fast attack (tending to zero), a longer response time (250 ms), a -30db threshold and a ratio that may go from 10:1 to 20:1, to infinite:1, according to how steady is your hand :)
After that, is time for the Equalizer, which may vary strongly from mix to mix.
My general suggeston anyway is to filter everything below 40/50hz with a high pass filter, and above 7000hz (cut even more, if you thing you don't need so many highs) with a low pass, then you can scoop-subtract some more frequencies (but not too much) on the 2/3khz area to leave some more room to vocals, and boost around 80/100hz to give to the track some more low-end.
In order to reduce boxiness we can also cut between 180 and 250hz.
So this bass chain is: OVERDRIVE->COMPRESSION->EQUALIZER

2. Dual track: this method is a little more tricky, but should give you a more extreme distortion, while keeping the same punchy lows, at the same time.
First, take the unprocessed track you've just recorded and duplicate it, then the first copy and compress it like on the first method, then add the equalizer, filter everything below 40/50hz with a high pass filter, and above 7000hz with a low pass one, and cut (9 or 10db) a bit around 500hz, with a small Q (so a wide range of frequencies affected). Then take the second track and add a distortion on the vst chain (Click Here for a dedicated article about FREE VST BASS AMP SIMULATORS), crank it really high, making it sound almost like a guitar (ALMOST, not TOTALLY, we'll need this track only for the grit), and add an equalizer, setting a low pass filter above 4500khz more or less (you choose the curve), and a high pass below 500hz, just to keep the growl. Finally, mix the two sounds, finding the right amount of volume to obtain a good, tight and defined low, end and some crunchy mid-highs.
So the chain with this method is:


T.1 COMPRESSOR->EQUALIZER 
T.2 DISTORTION->EQUALIZER 

Sometimes it's also a good idea to put a SECOND COMPRESSOR (click here for a dedicated article about Serial Compression) or a LIMITER at the end of the chain, not to squeeze the sound (for this task we have already used a Compressor) but just to set a threshold, to make sure the bass will stay on its place and will not consume headroom later, in the Mixing and the Mastering phase.


Become fan of this blog on Facebook! Share it and contact us to collaborate!!

Saturday, November 26, 2011

THE BATTLE OF THE PODS (comparison)




The Pod Hd Patch for this sample can be downloaded freely! Open the video on Youtube, the link is in the description!


Hello! Somebody asked me to write an article to explain more clearly the differencies between the various Line6 products, if you think that there is some mistake please report it and I'll correct it.

POD 1.0: the first, revolutionary amp modeling simulator, created in 1998, is a milestone in the world of digital sound processors, and set a new standard of digital tone quality, changing the perspective of digital guitar tone: not a stompbox that adds colours to the amplifier, but a digital workstation that simulates itself the sound of other amps. The technology comes from the knowledge accumulated through the two amplifiers created in the previous two years: the combos Axsys and Ax2.

POD 2.0: the most famous. Published in 2001, this device has updated the 1.0 version with newer technologies (32 amp models, plus various effects), and it's still considered by many a standard, though talking about accuracy and harmonical richness it's been already surpassed by many competitors. The pod "classic" has received severeal firmware updates through the years, one of which  very substantial in 2006, to add the floorboard compatibility, and still today its technology is used for the Pocket pod, the Spider Serie, the floor pod/floor pod plus serie, the Flextone serie, and the Hd147 serie. The particularity of the spider (and pocket pod) serie is only in the presence, instead of the base sounds, of numerous presets done by simulating the sound of many famous artists.
This unit works in 16bit and 44khz, which may result limited for today standards, when compared with other units.

POD PRO: a Pod 2.0 in rack version, which adds more input and ouput connections, like the XRL output, and features a 24 bit signal routing, instead of 16. It occupies 2 rack units. 

POD XT: it represents a further evolution of the Pod family, featuring a new and updated hardware and software engine. It contains more amp models, an internal 24 bit routing, and a lcd display which graphically shows the amp-effect chain. It also introduced the concept of downoadable content packs with additional amplifier models and effects, sold separately (for example, fx pack, metal pack...), which expands even further the simulations available, adding more contents. The Pod XT technology is also used on the Flextone III amplifiers, and there is a floorboard version too: the Pod Xt Live.

POD XT PRO: it stands for Pod XT as the Pod Pro stood for the Pod 2.0. It has more outputs, which makes it very useful in the recording studio, especially thanks to an out dedicated to the REAMPING. Clearly there are more simulations and functions too, compared to the "bean" version. It occupies 2 rack units.

POD X3: It's an updated version of Pod XT which uses the same models in terms of software, but adds the possibility to pile up the sound of two different amps for the single guitar signal (for example you can obtain a sound with a Marshall amp and a Mesa Boogie amp layered), thanks to a remarkable hardware upgrade. It features also for the first time some preset specific for bass and voice, expanding the versatility of the unit up to trying to make it a "swiss knife" for the studio, as a multi use preamplifier, and Di-box too. The X3 Live is the floorboard version of this unit.

POD X3 PRO: Is the 2 unit rack version of the Pod X3, like for the other pods.

POD HD (bean, rack, floorboard): It brings even further the Pod Family, thanks to a hardware consisting in an upgraded version of the Pod X3's one, and a completely new software dotation. New amp models (for a total of 22, with the 1.3 firmware update), a better interface and display, presets for vocals and bass like for the Pod X3, and like that unit is possible to connect both vocals and guitar (or bass) on the same Pod at the same time, or using two different amps for a single guitar signal. The rack version features a dedicated out for the Line6 Variax guitar, and severeal other ins and outs.

POD HDX: it's the 2013 upgrade of the Pod Hd. It features more routing options (e.g. a Variax input in all models) and a more powerful processor, since the original Pod Hd had some problem when stacking multiple amplifiers on the chain (memory overload).

I hope I've cleared your doubts through the ocean of the Line6 Pod offers: in the end they're all different generations of technology used for all of the Line6 products of that particular moment (the same that happens for example, in the car industry).


Become fan of this blog on Facebook! Share it and contact us to collaborate!!

Saturday, November 19, 2011

HOW TO MIX VOCALS (free Vst Plugins inside)



In this tutorial I'm going to explain my way to mix vocals.
I'm not saying it's the best method, it's just the one I have chosen after severeal experiments, but I'd LOVE to hear comments, feedback and opinions from you!
First off we need to import the recorded tracks on our DAW, or to record them (CLICK HERE FOR A TUTORIAL ON HOW TO RECORD VOCALS)

Once you have the vocal tracks recorded at the right level (the signal must not be too low nor too loud to minimize data loss), and the proper Editing and Autotuning (if needed) is done, the first vst to put on the insert chain (or bus, according to the number of vocal tracks you have and how fast is your computer's cpu) is a DE ESSER, a plugin to remove the "hiss" frequencies from your vocal track (Click Here for an article about Deesser). This plugin is needed if there is a sibilance problem, that cannot be solved by  changing mic or moving the singer sliglhy back from the mic.
After you've found the right frequencies to remove, it's time to insert a COMPRESSOR, and there are many free HERE, so just try some and choose, the functionality are basically the same for all of them.
If you want to use the KJAERHUS compressor we've already seen, for example, you could set a 5:1 ratio (adjustable from 4:1 to 8:1), a fast attack (often as fast as possible), a medium release (around 0,5 seconds) and then adjust the treshold in order to activate it at the right time, like in this picture:


After the compression you'll need an EQUALIZER (for example the NYQUISTEQ), and luckily most of DAWS have one, so you can retouch some frequencies, but this is really variable according to the kind of vocals, so it's hard to suggest the right frequencies to modify.
First off remember that subtractive eq is always better than additive eq, anyway you might want to filter off some of the useless frequencies below 80/100hz, reduce a few db (like -2,5) around 200hz, and boost a little (+2,5db) around 2500hz, which is the main frequency area the human ear captures, and it helps to put the vocals even more at the centre of attention.
If you feel that the signal is too weak (because of the poor microphone, or of the lacking of a decent preamplifier), this is the place where to add a TAPE SATURATOR, like the JSMAGNETO, to thicken the sound a little bit before passing to the effect phase.
Talking about effects, someone uses delay and reverb, some just one of them, personally I prefer to use just the DELAY (click here for a dedicated article with free plugins).
The ideal would be to create an FX BUSS where to put the delay and then to send it to the various vocal tracks, so you can adjust the right dry/wet signal ratio for every track, but this time we're just going to add it to the vocal insert chain (click here to see a dedicated article on how to use an FX Buss), and set it with a short delay (100ms) and a short feedback, in order to make it sound more like a reverb, but without that "ambient" feel.

In order to give the vocals a bit more "room", to make them sit better in the mix, we can also add a short REVERB, to give them a less "in your face" position, possibly on a FX channel track to make it less "Cpu Hungry", but this choice is optional. The reverb recommended setting is with a decay time of around 3 seconds and a Pre Delay of 50ms.

The last thing to do is to CLEAN UP your sound of breath and other various noises recorded before the beginning and after the end of each take, and you can just cut them away, or make those takes to fade in and fade out.

So, basically my chain is: DE ESSER->COMPRESSOR->EQ/FILTER->TAPE SATURATOR (if needed)->DELAY->REVERB (if needed). and then I clean up the tracks.

Sometimes it's also a good idea to put a LIMITER at the end of the chain, not to squeeze the sound (for this task we have already used a Compressor) but just to set a threshold, to make sure the vocals will stay on their place and will not consume headroom later, on the Mixing and the Mastering phase.

Additional awesomeness:

If you want to add an interesting effect that thickens up your vocals and gives them a cool "chorus" effect without using an actual chorus, just copy the vocal take into a new track, apply the same effect chain and then add a pitch shifter at the end of the second track. Put "semitones" to zero and change the variable "cents" to -20, to create a track just slighly different, and mix between the two tracks to give your vocals a cool effect that works really well with clean singing.


Become fan of this blog on Facebook! Share it and contact us to collaborate!!

Saturday, November 12, 2011

SOMETHING ABOUT REAMPING


Today we're gonna talk about reamping. What is it? It's an intelligent technique, in order to add infinite possibilities in terms of guitar tone shaping.
Once you have recorded a guitar, usually you can add processors like equalizers, compressors, delay, reverb, and then alter the base sound in the mixing phase, but you won't be able to obtain a sound COMPLETELY DIFFERENT from the one you've recorded. Reamping talks exactly about this: did you record with a Mesa Boogie Dual Rectifier instead of a Peavey 5150 and you've painfully regret that choice? No problem, we've got the solution :)

There are two types of reamping: the "standard one", and the "plug-ins one".

The standard one is the simpliest solution: while you record, you need to split the signal from your guitar through a DI-BOX (there are some made specifically for reamping, like the Radial ones), in order to have a clean, balanced track straight to your audio interface or mixer, while the other guitar track goes to the amplifier (or other hardware signal processor, as a digital amp simulator) and it's recorded as usual (with a microphone, or from the line out if there is a speaker simulator output). At the end of the recording, so, you're gonna have two recorded tracks for each part you've played: one that comes from the amplifier, and another one, exactly identical, but clean. Just the bare guitar sound straight to the DAW (digital audio workstation).

In order to change the sound of this clean track (and this is the reamping), you take back the clean track from the output of your audio interface (lowering the volume in order to match the right level to be sent to the input of another amplifier), and go back to the input of the second amplifier, in order to play the same track you've recorded before on this other amp (or other hardware processor, like POD), and capture again (with the microphone or from the line out) the new sound.
You can do this all the times you want, until you're gonna have a certain amount of choices, played flawlessly and reducing to the minimum the amount of time wasted, instead of playing the track again for every amp you want to use.
This method will allow you also to try different combinations of sounds and layering, to enrich your tone (for example, for the Evanescence's first album's guitar tone, they say they've first played the guitar tracks on a Marshall head, and then they've reamped the same tracks on a Mesa Boogie Dual Rectifier, and finally they've layered the two sounds).

About the "Plug-in reamping", (plugins like the Virtual Guitar Amplifiers seen on This Article) instead, keep in mind how these plug ins work: you activate it on a track of your DAW (digital audio workstation), then you record the part clean, and the effect (e.g. distortion) is applied on it in real time, giving you the freedom, once finished recording, to keep on changing it, adjusting the amplifier and the effects, or even taking away everything, leaving you back the clean track.
In order to play properly, you're gonna need a good low-latency audio interface, otherwise you're going to hear the sound to come out with some delay, and will be almost impossible to do a good recording. A latency of 10ms or less is tolerable, but if it's much higher, is better at least to download the ASIO 4 ALL drivers and try to set them to reduce the latency.  

To reamp a guitar (both with real amps or plugins), has some limitations. There is no way to create a feedback loop from a clean, direct-recorded guitar, and this is one of the reasons to use a real amp, at least as a monitor, because it can create some feedback that makes the strings to vibrate, and this effect is recorded on the clean track. Not only: other limitations may be because using digital simulators (both hardware or software) would not appeal the purist of the real tone (guitar->jack->amp->microphone), so keep that in mind, when working with other people.
Beside these limitations, in the recording situations where you have to keep in consideration the need of radical sound changes on the mixing phase, you will find in reamping a very useful and time-saving tool, and today as we've seen, there are more ways than ever to use it.
For example I always suggest to everyone, when recording, to split the sound and record a clean track anyways, it's just a matter of adding a jack and a d.i. box to the chain, and it may always turn out to be a blessing, later on.
In case the extra track turns out later to be unnecessary... A click it's all it takes to get rid of it ;)


Become fan of this blog on Facebook! Share it and contact us to collaborate!!

HOME RECORDING at ZERO COST (...well, almost). PART 3/3




Once you've acquired on your project all the tracks you need, it's time to Prepare the Project for Mixing, a process that can be divided in four steps (Click Here for a dedicate article about Project Preparation): track disposition, group channel tracks, editing and autotuning, and then pass to the "Balancing Phase", which is the phase where you must try, just by moving the volume faders, to reach a good balance between the tracks, taking note of the faders that feels "unstable": those those are the ones in which you can't find a stable position throughout the whole song. After the Balancing Phase is time to Pan the tracks (Click Here for a dedicated article), in order to create a stereophonic Soundstage.

Moving onto the Mixing Phase, a suggestion is to use as less effects as possible before entering in the DAW, and to use the real time VST effects (unless you have a very expensive external piece of hardware that you need to use for some particular purpose), in order to be able to change "on the fly" the settings we need to fix, without having to record the track again.
This topic brings us to the "effects chapter".

Vst is the standard used for the plugins on most of digital multitrack softwares (other standards are Direct x, Audio Unit, Rtas), and they can usually work real time, which means that you can insert-modify-delete them while you're listen to the track, hearing the changes live.
There are many freeware vst suites of good quality, if you don't want to use too expensive software (although WAVES does incredibly good and professional plugins): my suggestion goes to the complete KJAERHUS CLASSIC SUITE, which comprehends Chorus, Delay, Compressor, Limiter, Reverb, and everything else you might need, on an interface that resembles the classic rack devices, and, specifically for the guitar, Here you can find a collection of the best Amp Simulators, both free and commercial.
About the effects, this blog offers a dedicated article for every type of effect, also all the modulation ones, with a selection of free Vst for each type, you can find them HERE.
These plugins can be used also entering with the guitar directly into the audio interface or mixer, the effect is applied real time, or on the playback (according to if the software supports real time effects and the computer is fast enough).
Once you have balanced volumes and panning (the spatial disposition we've seen earlier), you'll find yourself with a (hopefully) decent sound, but since the sound of all instruments will be pretty "natural", there will be some frequency that will "fight" between the instruments, and the strongest sound will cover the weaker ones; this is not just a matter of volume.
You will have to work with the equalization and compression, which are the two main sound sculpting tools, click on the two links to learn more.
Once you think that every sound is intellegible (volume levels, equalization, compression) and you've applied all of the plugins that you wish, you can polish even further the sound using a noisegate on the tracks that need it, to get rid of the noise, the hum, and crackle, and once done, we're heading toward the conclusion of the project.
You can pass to the MASTERING phase, or just set the "beginning" and "end" markers on the project, and export, in Wave, or Mp3.

I'd say we have briefly touched all of the arguments, in a very superficial way, and surely I've left out something, or used methods that not everyone will agree, but the idea behind this guide was to create a small tutorial for beginners, with the tips and suggestions the experience has taught me, with the lowest budget possible, and under this point of view I'd say I've succeeded.
Let me know what you think about it, and keep in mind that experience is the best teacher: try, experiment and believe your ears!

Have fun!


CLICK HERE FOR THE PART 1/3 OF THIS TUTORIAL

CLICK HERE FOR THE PART 2/3 OF THIS TUTORIAL

Become fan of this blog on Facebook! Share it and contact us to collaborate!!

Friday, November 11, 2011

HOME RECORDING at ZERO COST (...well, almost). PART 2/3


First off I will obviously leave you the freedom to organize your base as you wish, choosing the midi instruments (like drums) and creating the structure of the song.
If you're interested in recording just the guitar, on the web there's plenty of free bass, keyboard and drum loops, which can be sequenced with softwares like the free ACID XPRESS, among the others.

If you want to record you own voice, or bass, or other instrument, instead, you can just plug into the audio interface or mixer (if you record vocals is recommended to use the XRL input, click here for a dedicated article about Recording Vocals), same if you want to microphone your amp, either case make sure the input level is high enough, (you can check it into the input section of your DAW), but not so high that it's cut from the program, otherwise it will distort (Click Here for an in-depth article about Gain Staging).
We're going to add the effects later.
The Bass should instead be recorded straight with the jack into the interface, trying to get a good starting sound using the tone pots and the right pickup configuration (click here for a dedicated article about guitar and bass pickups).
Finally, in order to record the keyboard, if we don't want to use them as a midi controller but we just want to record the audio from the line output, we need to keep in mind that, in order to get a stereo sound, we need to use both of its line out (left and right), on two separated tracks to be able to play adequately with the stereophony; in this case too, 99% of the sound shaping should be done directly from the keyboard.


After these few first basic recording suggestions, let's focus on the guitar, which is our main topic :)
We take for granted that you've already installed the audio interface drivers, the DAW, and you have already done a base, with loops, or synths, or recorded instruments. Import the base into a new multitrack project and when everything is set, there are basically two ways to record your guitar on the DAW:

- Directly on your audio interface or mixer, dry (and then we'll add plugins) or with some hardware amp simulator.


- Microphoning the amp.


If you choose to go directly into your interface or mixer, it is probably because you can't make too much noise at your place, so this method allows you to get the sound you need without bothering who's around you, working with the headphones. Nowadays the market is full of good hardware guitar processors that simulates the response of real amps, cabinets, stompboxes and everything else, and the price range in pretty wide on this field too, which goes from the cheaper models (ZOOM), to the mid-price zone (LINE6 and DIGITECH), all the way to the highest-end (FRACTAL AUDIO, KEMPER).
These devices allows you to get a good guitar sound and go straight on your DAW without miking a cabinet, but if you (rightfully) want to hear the full roar of your amplifier (better if tube-driven) and your pedalboard that costed you a fortune, you will surely want to microphone your cabinet. There's plenty of microphones out there, and some are made specifically for the guitar, like the SHURE SM57, or some SENNHEISER, among the others.
There are many ways to microphone a cabinet: close miking, or a bit more far to catch some of the room reverb, straight to the cone or inclined, pointing the center of the cone or some point between the center and the border.
Here Is An In-Depth guide about How to Microphone your Guitar Amp.
As I already said before, whether you chose to record directly into the interface, or to microphone the amp, make sure to not cross the 0db, in order to leave the sound wave on its full integrity.

Important: keep in mind that the tracking phase, which is phase in which the sound is acquired into the project, it's the 60% of the final result! The single tracks can later be modified, obviously, but the original timbre of the sound will remain the same, thus I suggest you to keep recording until you reach the best starting sound and performance performance you can, don't settle with a mediocre result!


CLICK HERE FOR THE PART 1/3 OF THIS TUTORIAL


CLICK HERE FOR THE PART 3/3 OF THIS TUTORIAL


Become fan of this blog on Facebook! Share it and contact us to collaborate!!

Thursday, November 10, 2011

HOME RECORDING at ZERO COST (...well, almost). PART 1/3





Hi everyone! 
This is a small guide to set up a home recording studio, to record your own demos and samples. 
I'll try to focus on the cheapest gear and software required in order to achieve good results, to suggest the most effective techniques and to point out and solve the most common mistakes that can be committed. 
This article has to be intended as a TREE article from which start, and then you can go reading the single dedicated articles of the things you need to learn more.
The most important thing: once learned these few first rules, it will all be a matter of experience, so if the first tries doesn't satisfy you, just write down what to adjust, and fix the problems on the next recording!  


Here's what you need to start a small home recording studio:


1) A PC, or a MAC. Just keep in mind that the Cpu and the Ram are equally important, but while the Cpu takes care especially of the real time Processing on tracks, the Ram is crucial when using virtual instruments (like drum samplers), so think about what you need, when it comes to choose the right hardware configuration.


2) Software: in order to do multitrack recording and mixing, you're gonna need a DAW (digital audio workstation). There are many out there, some free (like AUDACITY, PRESONUS STUDIO ONE FREE and KRISTAL AUDIO ENGINE), some cheap but very effective ones (like REAPER, MAGIX or CAKEWALK), and some more expensive, but very stable, compatible and reliable ones (like CUBASE and PRO TOOLS). The most important thing when choosing a DAW is to make sure that it supports VST plugins.


3) Instruments: guitar/bass/keyboard/microphone... everything you want to record.


4) An Audio Interface (Click Here to check out an in-depth article on which audio interface to choose), to connect the computer with the instrument, or with the microphone. There's a plenty of audio interfaces, usb, firewire and thunderbolt, in a wide price range, and they're very important for two reasons: to reduce the latency to the minimum (latency is the delay between the input and the output signals), and to provide a decent preamplification to the signal (no, the integrated headphones preamp is not decent). 


5) Headphones or reference monitors (Click here for a dedicated article!): you will need a decent quality output device too, like headphones (look at the frequency range, the wider, the better), or reference monitors (active speakers made to work in the studio): there are many producers, from the cheapest (BEHRINGER, very unsuggested), to a mid-price standard (M-AUDIO, for example, but there are many more), to the pro choices, which are obviously more expensive (AKG, YAMAHA among the others).
The idea is to have a device that gives us a realistic representation of what we're working on: what we need is a pair of monitors good enough that if we mix a song with them and the song sounds good, the same good balance translates for example on a portable mp3 player or on a car audio system, and unfortunately this realism is achievable only with higher end devices, so our suggestion is to look for the reviews online and check out the best realism-to-price ratio available.


6) At least one good Microphone: we will need a microphone to record Vocals or other acoustic instruments. In the early blues and jazz records a single microphone, carefully placed and treated, was enough to record a whole band; today we obviously prefer to mic every single instrument in order to have more flexibility when mixing.


7) Virtual Instruments: since not everyone has access to a recording room where to track drums or other acoustic instruments, we will need some virtual (or even hardware if we prefer the old school approach) sampler to create the midi tracks with the instruments we cannot play (or record), for instance drums and synths, like orchestrations. A very good drum sampler, with freeware license is MyDrumset, TchackEdrumMdrummer SmallDRUMCORE FREE, or GTG Drum Sampler.


CLICK HERE FOR THE PART 2/3 OF THIS TUTORIAL


CLICK HERE FOR THE PART 3/3 OF THIS TUTORIAL


Become fan of this blog on Facebook! Share it and contact us to collaborate!!

Wednesday, November 9, 2011

BASIC CHAIN: GUITAR AMP SIMULATORS


Hello everyone and welcome to my blog! On this song I used for the guitars three free vst: Poulin Lepou Le456, which  is an awesome preamp simulator based on the engl powerball, Voxengo Boogex, as a power amp/speaker simulator, and the overdrive simulator TS SECRET, which emulates the response of an Ibanez Ts9 overdrive stompbox.
On the boogex I used as a speaker impulse one of the Guitarhack Set, and the vst chain on the guitar was:
NOISE SUPPRESSOR -> OVERDRIVE -> AMP SIMULATOR -> SPEAKER SIMULATOR -> COMPRESSOR.
Stay tuned for more info and useful links and secrets!

Become fan of this blog on Facebook! Share it and contact us to collaborate!!

LinkWithin

Related Posts Plugin for WordPress, Blogger...