Saturday, November 26, 2011


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).

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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.

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Saturday, November 12, 2011


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 ;)

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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!



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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!



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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.



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Wednesday, November 9, 2011


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:
Stay tuned for more info and useful links and secrets!

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