HOW TO RECORD AND MIX DRUM CYMBALS (PART 2/2)

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Once we have seen how to acquire our cymbals sound, we sould now find ourselves with Four mono tracks: two for the Over Heads (Left and Right), one for the Hi Hat and one for the Ride, therefore now it's time to start mixing.

- Overheads: the idea is to Pan those two tracks very wide, often 100% left and 100% right, to give to our Crash Cymbals as much space as possible, then we need to Hi Pass and Low pass them, in order to give to the tracks a proper frequency range that will not make them fight with the others.
First off we should create a Group Channel track where to route the two single tracks, and then we can start off by loading on the Group Channel a Hi-Pass Filter, taking away everything below 100hz to 500hz, until we find the spot where the bulk of the drums disappears (unless we don't want purposely to retain some spill on those tracks too, to beef up the global drum sound), then we can Low-Pass around 10/12khz, to avoid the sound to become too harsh.
Now the only thing left to do is to start manually to pin down the Resonant Frequencies (which are, if we look at the Spectrum of our sound, the frequencies that produces the highest and most annoying peaks), and see if our cymbals are Eq Masking the frequencies of other important tracks (Vocals, for example).
If this happens, it's a good idea to scoop a bit around the 3/4khz area to clear some room for Vocals, and if needed we can also take away something around the 500hz area too, to eliminate a bit of "Room Mud".
The last thing to do, if needed, is to Compress the sound just a little, in order to soften a bit the hardest hits.

- Hi Hat: This cymbal is seen more as a part of the snare sound than an ambient cymbal, and therefore shouldn't be Panned too far from the snare: someone sets it straghit to the centre of the soundstage, someone else pans it slightly to the left (about 12.5 left), as if it's heard from the drummer's perspective.
We can Low-Pass the sound, as for the others cymbals, until most of the snare goes away (300 to 500hz, usually) and then Hi-Pass it at about 10khz, eventually taking out some other frequency if the general sound feels a bit "gongy". To add some brilliance we can boost a little around 6khz.

- Ride Cymbal: this cymbal should be Panned usually somewhere on the right area (like 12,5 right), and its particularity, compared with the Hi Hat, is a usually stronger low end. In order to make the ride sound brighter we can Hi Pass the sound to about 3/500hz, then we can boost a couple of dbs at about 10/12khz to add some air and sparkle. Watch out for resonances on this track too!
Similarly to the Hi Hat, if we feel that there are some frequencies that are fighting with the Vocals ones, we can scoop around 3khz.

We can also route the Hi Hat and the Ride Cymbal on another Group Channel Track to add a little bit of Compression, just to peel off some of the peaks.

Very important: If we're using Compression (and usually it is suggested for hard rock songs, up to the most extreme metal, not for softer genres), we must keep in mind that the settings should be very very low, since it's really easy to create an unnatural effect with cymbals! Same is for Reverb: usually it's not suggested, but in some cases, when the overall drum sound is very dull and lifeless, if we use a Plate Reverb on "homeopathic doses", it can add some body and room to the general sound.

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HOW TO RECORD AND MIX DRUM CYMBALS (PART 1/2)

Pic. 1

Hello and welcome to this week's article! Today we will talk about How to Record Cymbals!

There are many ways to record Cymbals, and the technique has evolved greatly throughout the time: from the early recordings, at the beginning of the 20th century, with nothing but a condenser microphone in the middle of the room to catch the sound produced by the whole band, to the most complex combinations of close miking of any single drum piece with other microphones to take care of the global room sound. 

The technique we're going to show you today it's a compromise between the need to capture the life of the recording room and the flexibility of having two single tracks for Hi Hat and Ride.

This technique can be successfully used with a natural drumset as well as with a triggered one (or a drum with its signal Replaced), although with a triggered drumset, if properly muted, is even better: we will have "clean" cymbal tracks, reducing to the minimum any unwanted mic bleed from other drum parts, and this will help us greatly during the Editing Phase.

Some musician even prefer to record cymbals without using the drumset at all, using just the cymbals and their stands, microphoned: the drummer plays the cymbals as he would do using a drumset, following the song on his headphones. This method may feel a bit unnatural, but it's the "cleanest" way to record the cymbals signal with zero bleed from the drumset :)

Let's start talking about Overhead Cymbals (Crash, China, Splash...): these are the cymbals that are positioned above the drumset, and their sound can be captured with a couple of overhead microphones (Pic. 1), which can be dynamic, but it's very suggested to use Condenser Microphones (e.g. for our recording we have used a pair of Akg C1000), since they can capture many more details. They can be positioned with the X/Y method, as shown in Pic. 1, creating a 90° angle to one another, at around 3 or 4 feet of height (1-1.20 meter) from the drumset. This setup is good to record the whole drumset / cymbals area providing phase coherence (therefore less frequency cancellation). 

There are other methods too, if you want to try, as for example positioning the two microphones at the same height but on the left and right side of the drumset, pointing towards the center of the set, and this method is used to add a sense of width to the recording.

Pic. 2

Now let's talk about the Hi-Hat (also known as "Charleston"): this cymbal actually consists of two cymbals that are mounted on a stand, one on top of the other, and a pedal which can be used to clash and hold the cymbals together. This cymbal can be close-microphoned, since it's a very important part of a drum sound, complementary to the snare. We need to process it alone, if possible, so the Hi-Hat can be microphoned with a dynamic microphone, such a Shure Sm57 as shown on Pic 2, on the upper side of the Cymbal, at a distance of a couple of inches, at the opposite side of the Snare, in order to catch as less snare sound as possible. 

Pic. 3

The Ride cymbal is another important drum part, with a function similiar and alternative to the Hi-Hat, so it's suggested to record it with another dedicated microphone, which can be a dynamic one as well, but this time the right positioning is Below it, at about 2 inches of distance (Pic. 3), in order  to catch less bleed from the other drum parts.

These four microphones must be connected to four Microphone Preamplifiers, therefore we will need an Audio Interface with at least four Mic Preamps, or an external Mic Preamp / Mixer to be connected to our Audio Interface on a way that lets us stream the sound of each single microphone on a different track.

Ps. Thanks to Francesco Paoli and Cristiano Trionfera from Fleshgod Apocalypse for their help on recording the drums of my band! ;)

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HOW TO CHOOSE THE RIGHT AUDIO INTERFACE (a guide for dummies) PART 3/3

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- Something about Microphone Preamplifiers: As we have already seen, the quality of the microphone preamp components and the quality of the ADC - DAC converters is where most of the price of an audio interface lies. The more an interface is expensive, the better probably those crucial sections are, in terms of construction and electrical components quality.
Which are the characteristics of a Mic Preamp, beside its intrinsic quality? The presence of a 48V Phantom Power switch, that is used to transmit electricity through the cable to a Condenser Microphone, a kind of microphone that features an active electronic circuitry and cannot work without this function; beware not to turn the Phantom Power on when plugging a dynamic microphone, otherwise you may damage it!
Another interesting feature often found on Mic Preamplifiers is the 20db Pad, a function that drops the input signal 20db lower, and this function is useful when plugging to the interface some instrument that has a signal too high, so high that it cannot be compensated just by turning down the input knob.
Beware when recording of avoid clipping at any cost! If the "clip" led (or meter, according to the type of interface) turns red, the input level is too high, this means that some part of the signal gets lost and it's interpreted by the DAW as plain noise. In this case, is very suggested to record again the part at lower settings, because it's practically impossible to restore.

- Something about the Dsp Processors: The most high end interfaces on the market sometimes features an additional processor, created to provide effects (such as Compression and Reverb, for example) without consuming the computer's CPU.
This is particularly useful when recording Vocals, since singers often need some real time processing on their headphones, at the lowest latency rate possible, in order to hear themselves "in the mix" while performing.
We can find a Dsp on the top-range Interfaces of many manufacturers: Motu, M-Audio, Focusrite, among the others.
Another use for Dsp processors is not only to monitor the sound with the effects on, but also to record the sound itself with the effects applied, and under this point of view there's plenty of interfaces, especially for Guitar and Bass, such as the Line 6 Pod (click here for a dedicated article) and many other devices, that will apply to your sound an emulation of Amplifiers, Cabinets, Effects and so on, all without burdening on the computer's resources: let's not forget that some of these effect devices can be used as an Usb Audio Interface too!

- The Right Interface for The Right Job: When choosing an audio interface it's important to have clear in mind the main use we intend to do with it.
Recording: If we need to use our audio interface mainly for recording, we need to choose the one with the right amount of inputs for our instrument, for example if we need to record drums, we will need an interface with many mic preamps (possibly at least eight inputs, for kick, snare, hihat, 3 toms, and 2 overhead). Another interesting choice is a mixer that sends all the channels into separate channels of our DAW.

Then we need a good quality ADC (analog to digital converter) with a high dynamic range, at least 96 dB. A high dynamic range for the DAC (digital to analog converter) isn’t needed, because the audio interface is only used for recording.

Mixing and Mastering: For these tasks we will need instead a good quality DAC (digital to analog converter), at least 110dB of dynamic range: this is necessary since we want the result of our mixing to have the same headroom it had when we recorded it, plus a good mixing quality will help us in the Mastering Phase too.
Another thing we may need for mixing is the correct number of inputs and outputs, as we've already seen in the second part of this tutorial: if we have some hardware processor such as an equalizer or a compressor, we will need at least two inputs and outputs, and maybe some S/pdif connection will help too.
If we use it just for mixing or mastering, our audio interface for mixing won’t need a high dynamic range of ADC (analog to digital converter).

- Happy First Birthday Guitar Nerding Blog!!: With the conclusion of this three weeks long article we're celebrating the First Atoragon's Guitar Nerding Blog's Birthday!! Thank you to all the viewers, keep on supporting us, becoming fan on Facebook and sharing our links, commenting articles and videos and collaborating with your articles! We need your help us to becoming the no.1 resource site on Home Recording Tutorials around!

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Rock On!

Atoragon.

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HOW TO CHOOSE THE RIGHT AUDIO INTERFACE (a guide for dummies) PART 2/3

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Once we have seen the different types of connection of the various audio interfaces on the market, it's time to take a look at the main features that an audio interface should have, starting with the ADC-DAC conversion: what is it? It's the conversion from an analog signal to a digital one (usually in binary system), and the reverse operation.

This process is very important if we want to make music, since it's going to impact on the whole production, therefore a good conversion leads to a good digital representation of the wave, and the better the electronic components of our device are, the better the conversion will be.

Working in the digital domain, we need to have a final wave at a resolution of 16 bit and 44'100hz (the double of the highest frequency audible by a human ear), in order to be burned on an audio cd, but if we want to produce music that sounds really well on this device we need to operate at higher resolutions, and then Dither the signal down to that amount.

- RESOLUTION AND BIT DEPTH: This higher resolution basically consists into operating at a bit depth of 24bit (which is a fidelty to the original wave twice as high as the 16 bit one), and at a sample rate higher than 44'100hz (to have even more headroom available, thus less distortion) in order to have the freedom to create a wave that sounds better then it could be reproduced by a regular cd player, so that when it comes the time to dither it down to 16bit 44'100hz it will sound better than if it was recorded directly within those limits.

So when choosing an audio interface keep in mind that it should let you work at a bit depth of 24 bit at a sample rate of at least 48khz (some devices arrives even at 192khz, but I personally don't think that much is necessary: many plugins don't work properly at that rate, plus the wave files becomes bigger and bigger on our hard drive).

- INS AND OUTS: another important feature that we must analyze when choosing an audio interface is the amount of Ins and Outs. What do we need to do with this interface? Do we need to record our whole band or just guitars and vocals in our bedroom? We need to have a clear idea of how many inputs we need.

The Input - Output jacks are Mic inputs (balanced), Line Inputs (unbalanced), and Instrument Inputs: the Xrl jacks, or microphone jacks, are balanced jacks (signal level is higher, because it passes through a mic preamplifier), while the 1/4 jacks (also known as TRS) may be balanced or unbalanced (the unbalanced ones have a significantly lower signal level and are called Line Inputs).
Line Inputs: These TRS jack labeled "Line Input" are the ones optimized to receive sound from an external preamp, or an external mixing board, since this input bypasses the preamp section of the interface.
"Line Outputs" are instead used to send the signal to external outboards, like for example hardware effects processors, or other amplifiers, if we want to do a Reamping.
Instrument Inputs instead have a higher impedance (therefore a lower signal) than a balanced input, but the signal is higher than a line input; ideally these inputs stands halfway between a balanced jack and an unbalanced one in terms of signal, and are suited specifically for guitars, basses, and other instruments.
Most of modern audio interfaces anyway features one or more combo jacks balanced-unbalanced that automatically switch between the two formats recognizing the type of instrument, plus they often offer some other unbalanced input.

If we need to record a whole band there are interfaces with several Xrl or Combo balanced inputs, or even some mixer that can directly connect with the pc via Usb or Firewire, just make sure that if the mixer connects with the pc, you can track independently all the single channels on your Daw (an option possible, for example, with the Phonic Helix serie), because most of them will just sends all of them to a stereo output, making further mixing impossible!

Sony / Philips digital interface (S/PDIF), this input-output consists in two plugs: In (usually red) and out (usually white). This connection has many uses: to connect an external mixer to the interface, connect the interface to another interfaces or to studio monitors, to connect reproduction devices such as an audio cd player to the interface without significant signal degradation.
Midi input - Output: this is a kind of connection not featured in every audio interface, and it lets your keyboard, or hardware drum sampler, or any other midi-driven device to be connected with the computer and to record midi tracks;
lately this connection on the instruments is often replaced directly with an usb connection, so it is possible to plug the keyboard directly to the pc without the need of a Midi port.
ADAT Optical Input - Output: it's a pretty uncommon connection, featured usually on the upper-range interfaces, capable of carrying large amount of data without any additional digital to analog conversion. It may carry the data stream of up to eight digital audio channels simultaneously, so if we connect it to a device with more inputs, we can use this port in our interface to expand the number of channels streamed to our Daw.
Headphone Output: a headphone output or more is very important, and it features an internal headphone amplifier with volume control to make it loud enough even for a singer, so that he can hear the base while he's singing. If more outputs or more volume are needed, it is possible to add to those outputs additional headphone amplifiers.

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HOW TO CHOOSE THE RIGHT AUDIO INTERFACE (a guide for dummies) PART 1/3

Hello and welcome to this week's article! today we're going to talk about audio inerfaces, and how to choose the one that suits your needs!

Back in the day, like 20 years ago, professional audio hardware was very expensive, at the point that it was impossible for an amateur to buy a decent equipment in order to do a good home recording, but lately, more or less in the last five/ten years, the manufacturers have started producing an increasignly wide range of interfaces of good quality, at a reasonable price, that allows anyone to try to record their own music at home, and this is the reason why the internet is full of "bedroom producers", me included :)

Why do we need an audio interface? Because we need to send somehow the signal to the Digital Audio Workstation, and the internal mic preamp of our motherboard it's just not accurate enough to reproduce it with fidelity.

Now let's analyze the main characteristics of an audio interface, in order to choose properly, starting by the kind of connection with the pc:

- PCI: nowadays only few interfaces are Pci (the internals interfaces), and they are used mainly for their AD-DA conversion, usually associated with a mixer or an external preamp. Those are often the cheapest solution and usually are also the best in terms of latency. The amount of data transmitted is very large too, the only downside is that once you have installed it on a pc, with screws and everything, it stays there, so it's unconfortable to carry it around and install it on other computers. Plus it's not compatible with laptops, and the sound quality depends on the quality of the external preamplifier you will use. (example of Pci soundcard: the M-Audio ones) 

- USB: it's the most common connection and it is already featured on every computer, so nothing else is needed, just make sure that your audio interface is at least USB 2.0 (3.0 is even better, although currently I am not aware of interfaces ready to support it), otherwise it can't transport enough data to handle a medium-sized project, and for larger projects even the 2.0 version may not be enough (it carries 400mbit/s, but it's slower and less stable than a Firewire port), so it may give some latency problem. 

For most of projects, though, usb 2.0 it's ok, plus Usb interfaces are fairly cheap and produced by a large amount of manufacturers, so the price spetrum it's very wide too, ranging from the cheapest ones (like Behringer), to the most high-end and expensive ones, like the E-Mu or the RME. 

Beware of the sample rate, though, since the higher it is, the less tracks the interface can handle, since their dimension increases greatly at higher rates.

The price difference is because of the specs (that we will analyze in this same article) and the components quality, especially the the converters and the Mic Preamplifier quality. When the Preamplifier section is good enough, there is no need for an external preamp, therefore we can save some money.

Another important thing to keep in mind is that Usb devices are driven by the Cpu. That means that they will keep busy some cpu just to run, and this might be a problem if we have a large project and try to make some resource economy. 

- FIREWIRE: this is a connection that is alternative to the Usb one, and it's not featured on every motherboard (altough a firewire Pci card can be bought for around 10 bucks), it does not support hot plugging (you must connect the device while the computer is still powered off, otherwise it may burn your motherboard), and also the average firewire interface costs a little more than an Usb one, so why does many people (me included) prefers those to the Usb ones?

Simple: a Firewire port carries more data than an Usb one (around 400 and 800 mbit/s, there are 2 different Firewire ports), it's more stable and reliable, offers a lower latency even handling big projects, and most importantly it doesn't make your Cpu work just to be active, therefore you have more resources to use on your project. There are many firewire interfaces, some of which are considered a standard among the semi-professional scene because of their price-to preamp quality ratio, such as the Focusrite and the Presonus ones.

- THUNDERBOLT: it's a new kind of port introduced by Apple, it can carry 10 to 20 gb/s, and today there are just a few (and expensive) audio interfaces that can use it, such as Apogee and Universal Audio, but probably (along with the Usb 3.0 standard) this is going to become the standard that over time will replace the Firewire connection. It carries a huge amount of data, it's reliable and today it's featured on the latest generation of Mac computers.

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HOW TO USE GUITAR OVERDRIVE (with Free Vst effects inside)

Hello and welcome to this week's article!
Today we're going to talk about Guitar Overdrive Stompboxes!
Overdrive is the sligh distortion effect generated, at the beginning unintentionally, from the tube amplifiers used around the fifties/sixties on a live environment: in order to produce a higher volume, they were driven at maximum, and the musicians started to notice that the sound was changing: it wasn't anymore clean and full of headroom like it used to sound at low volume, it was angrier, with more harmonic overtones, more compressed and the hot rodded tubes were producing a roar-like effect on the lower frequencies.
This effect became more and more sought after through the years, so the producers started manufactoring tools to add Saturation to the amplifiers without driving their volume to the maximum, therefore preserving the life of the Tubes;
these tools consisted basically into adding an Overdrive Channel to the amplifiers, with a higher input level and a sequence of Gain Stages in order to make te signal more and more distorted, and also in creating Overdrive Stompboxes of various kind, in order to push the level before even entering in the amplifier input.

Today Guitar Overdrive stompboxes are used basically for two reasons:

1 - To add sparkle and a little grit to a clean channel, especially by blues, jazz players and anyone who is looking for a vintage sound.

2 - To boost an already overdriven sound, in order to make a guitar solo cut more through the mix, or to play with the interaction between the overdrive and the distorted channel of the amplifier. This last method is mainly used by hard rock and heavy metal players: to get in the amplifier input with an already high level sound, but with no distortion, adds to the distorted channel of the amp a pleasant boost effect on the mid-hight frequencies, helping to achieve the "chugging" sound requested by the most modern and extreme metal genres.

Focusing on a Rock-Metal sound, the boost effect is achieved setting the overdrive with the level at maximum, the gain at zero, or anyway very low (this control is often used when trying to drive a clean channel, not when boosting an already distorted channel), and with the tone control around 12 o'clock. You can anyway cut or boost the tone setting in order to bring out more or less the mid frequencies, especially when using a strongly mid-oriented overdrive like the most famous of all stomp boxes: the Ibanez Tube Screamer.

There are many Vst simulations of guitar overdrive stomp boxes, reproducing virtually any overdrive pedal on the market, included the legendary Tube Screamer, and among those, we suggest you to try these free ones, which are impressingly good, especially when used for boosting a virtual amp:

Ignite Ts-999, based on a modded version of Ts-9

TSE 808, very realistic, now version 2.0

TS Secret, another good Tube Screamer emulator

Fretted Synth Drive Stomp, a guitar overdrive stompbox emulation


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HOW TO DO AUDIO EDITING (a guide for dummies) PART 2/2

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- Parts that needs to be removed: At the opposite side of filling the gaps, there are times when to cut away the silent parts is suggested: especially for drums, is suggested to cut some silent parts, for example on toms, which are actually played just for few seconds every few bars, in order to eliminate the unwanted "Bleeds", which are the sounds of the other drum pieces, taken from the microphones (for example the hi-hat microphone will surely have a large amount of snare bleed). The hi-hat, snare and kick bleeds are best not to be removed, but the microphone bleed of drum parts that are not played continuously, such as the crash or toms can surely be cut/muted when the part is not playing, in order to reduce the bleed effect.
Another thing that is often cut out is the noise of electric guitars: often, especially with distorted and microphoned electric guitars, you can hear a strong background noise (or "hum"), which, if summed in many tracks, can become unbearable. This problem can be solved by using a Noise Gate, but this option often takes away from the guitar sound some good harmonic too, so many producers tends to solve this problem just by cutting away the silence parts, where only the hum is perceived.

- "Combing": This is an editing technique which is used used for Vocals and Guitar Solos, but it can be used on any other audio recording:
Combing consists into combining two or more takes in order to create the "perfect take".
There are two methods for combing:

The "Full Take Method": Some producer prefers to have the singer to perform numerous "from top to tail" recordings, taking note while the singer is performing of the parts of the recording to be kept, and what needs to be discarded, in order to take the best part of each version, in order to create the best take possible. This method is more fatiguing for the singer, but sometimes lets the music flow more naturally and gives the producer some interesting interpretation nuance that with the other method would be lost.

The "Patchwork" method: this one is less time consuming, and consists into singing the single part (for example a verse) again and again until it's satisfying, and then pass to the next one; this method lets the performer to focus on the single small part, and usually works better on amateur recordings.

There is also another use for Combing: if you feel that, when recordings are finished and can not be re-done, there are parts repeated more than once in the song (for example, a chorus), and some of these repetitions are not satisfying, we might as well take the best version and just copy-paste it in the place of the other repetitions too, and as long as there are not particlularly distinctive elements, the listener won't notice that, for example, the guitar under the second chorus is exactly the same take copied from the first one.

There are many Audio Editing Softwares around, some of them even Free and Vst Compatible. 
Among these we recommend: Wavepad, Wavosaur, Ocenaudio. Give'em a try!

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HOW TO DO AUDIO EDITING (a guide for dummies) PART 1/2

Hello and welcome to this week's article! Today we're going to talk about audio editing!
Audio Editing is the single most important and time-consuming task to accomplish before starting mixing, in the Project Preparation Phase, and it consists in the correction of all the little mistakes that needs to be fixed on the audio tracks, in order to sound properly and not to draw attenction.

Let's start by saying that not every small error needs to be fixed: sometimes some slight acceleration or deceleration made by the drummer may improve the groove, sometimes producers even speed up the metronome of a couple bpm on purpose on choruses, in order to make them even more energetic, on a subliminal level.
Also the interpretation nuances of Vocals or guitar solos should be left alone, what really needs to be edited are the macroscopic errors done especially on the rhythmic side, manily by Drums, but also by Bass, rhythm Guitars or some attacks of Vocals and the other audio recorded instruments.

Today there are different solutions in order to help us on our editing tasks, especially for drums, Pro Tools has a feature called "Elastic Audio" that helps editing by marking the audio peaks and quantizing or moving them automatically, also there are other plugins called Beat Detection Tools, which are capable of finding the bpm and help you modifying the track, but we will today focus on the "Slip Editing" which consists into slicing and dicing the audio tracks and moving them, and is the only Free option among these.

- Techniques:  Slip Editing consists into cutting the tracks at the "Zero Point", which is the last silence (0db) part of an audio track right before the start of a sound, and moving the part forward or backward of a few milliseconds, in order to set it on perfect time. To do this many DAWs offers the "Snap to Grid" option, which helps us finding the right place, where the bar starts, if we set the right metronome and quantization for our project: this is very important, since many details of our mix are driven by metronome and quantization, for example the Delay Repetitions, so set this carefully. The metronome and the quantization of our project are MIDI driven, so the "Snap to Grid" option will snap the audio parts on the MIDI grid, which can be divided in 1/8, 1/16, 1/32 and so on, in order to be more precise for the positioning. 

When editing, thus applying a change in the continuity of an audio track, there is the problem of how to fill the gaps at the beginning or the end of the track that we have moved: usually if our track slips below another track that starts at the right time there is no problem, since the new sound will take the place of the old one at the right moment, but if we create a gap of silence, this is going to be noticed, so we will need to solve this problem with Time Stretching:
Time Stretching tools (like the Cubase bundled "Audio Warp") are plugins that helps us in widening or shortening the lenght of a sound while mantaining the pitch unaltered; his can be a solution for filling the small silence parts that are generated when cutting an audio part.
Another tool we have in order to "blend in" two parts that were originally distant, in order to make them less feel like one is replacing another, is Cross Fading: this technique consists into creating a quick fade out on the first track and a fade in on the new one, in order to give the impression that there's perfect continuity between the two sounds.
Luckily all the new DAWs and Audio editing Softwares features a Fade In - Fade Out tools.

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HOW TO DO PANNING - STEREO DISPOSITION OF THE MIX (a guide for dummies)

Hello and welcome to this week's article! Today we're going to talk about Stereo Disposition on a mix.
The moment when we usually decide the position of the single instruments in our mix's spectrum usually happens right after the Project Preparation phase, at the beginning of mixing, in the "Balancing Phase", the phase where, without using any plugin, we just try to move the volume faders up and down in order to find a good balance between the tracks, and take note of the tracks that feels "unstable": those tracks that we cannot just let sit in the mix on a single fader position, and that will require further processing in order to find their place.
It's a good idea to listen the mix in mono before starting panning (and check again at the end of the mixing phase too) in order to see if there is any Phase Cancellation, that may go unnoticed once the project is panned, but that can create problems (Click Here to read an article about Phase Analizers).

"To pan" an instrument means to set it on a position in the mix that can range anywhere from 100% left, to 100% right: if we put a track on a position that is (for example) 75% left, means that 75% of the sound will come out from the left speaker, and 25% will come from the right one, thus this sound will result closer to the listener than a 100% left sound.
We can play with the disposition in order to help every instrument to sit better in the mix, and give to the listener the feeling to be "surrounded" by the atmosphere of the song, or make him feel like he's in the middle of the rehearsal room, or even to have something that is moving from one side to another of the "soundstage". The creative uses of stereo dispositions are countless, and some of the first pioneers of stereoscopy have been Jimi Hendrix and Pink Floyd, while the first Beatles albums were originally recorded in mono.

Let's start by saying, as already stated on our Stereo Expander article (Click Here to check it out!) that Low frequencies are non-directional, while highs are very directional. As a result, pan low frequency sounds (kick drum, Bass) toward the center of a mix, and higher frequency instruments (shaker, tambourine, cymbals) further out to the left and right.

- Drumset: there are Two "schools": the "Performer Perspective" and the "Concert Perspective". I personally prefer the first one, as the second one is generally used just in certain live recordings: to set the "Performer Perspective", pan Snare and Kick to the centre, the Hi-Hat slightly to the left, the Ride to the right, and set the toms at the disposition you would find while playing the drumset. Finally, set the OverHead Microphones should be set pretty wide, often one full left and one full right. To give the "Concert Perspective", instead, just flip the left and right sides, as if someone else is playing and you're in front of the set, facing it :)

- Vocals: It's a good idea to set the lead vocals on the centre of the soundstage, and so the backing vocals, but if there is more than one backing voice you can set them one sligthly left (like 27%) and one slightly right; just remember that is a good idea to keep them balanced in the mix, unless you are not aiming to some particular effect. You can also try to pan two copies of the lead voice slightly left and right and "slip" one of them forward in the timeline of a few milliseconds: you will obtain a particular "chorus-like" effect that is sometimes heard on some commercial song.

- Guitars: the most common technique is to pan a rhythm guitar to the left (usually 60% to 90%) and one to the right (someone even pan them 100% per side), and if you feel that there's still a void around the centre, you can record a third guitar track and leave it to the centre, or copy the two guitar tracks and paste them to the centre of the spectrum (or a bit off centre, e.g. you can put the left guitar copy at 25% right, and the right guitar copy at 25% left) lowering them of 5-6 db, just to "beef up" the mix. The lead guitars are usually panned to the centre, as long as there is no voice on that part of the song, otherwise we can set them slightly aside, in order to avoid them to interfere with the vocals.

- Keyboards / Orchestrations: It's common use to pan the background pads or violins to the extremes, while treating the leads (whether they are orchestral leads/hits or synth leads) like solo instruments, so put them in the centre, or just slightly off-centre if the middle of the soundstage is too crowded; just a few points (like 5%) can make an instrument to interfere with the others.

- Addictional Awesomeness: there are some effects that lets you play with stereophony, such as the Ping Pong delay (a delay whose repetitions bounces from left to right, and that has to be loaded on a stereo track), but if you want a track, a virtual instrument or just an effect to MOVE inside of a song, you can Automate it (Click Here for an article about Automations!).


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PROJECT PREPARATION (what to do before starting mixing)

Hello and welcome to this week's article!! Today we're going to talk about how to prepare your project before starting mixing, and how to organize your layout in order to not go completely mad with the Daw interface, making the workflow easier, faster and more CPU friendly. Assuming that you have imported (or Recorded) on you project all the tracks you need, let's get rollin':

1) Step One - Track Naming and Disposition: The best way to start is to name every track with a name that makes it easy to find even in the most clogged workflow, for example "Lead Vocals", "Rhythm Guitar Left", "Rhythm Guitar Right", and so on, then organize their disposition in groups of tracks by dragging them higher or lower in the layout, for example many studios starts with the Drum Tracks, and Colour them with the same Colour, then the Bass Tracks, and choose a different colour, then the Guitar Tracks, with another colour, then the ‘extra’ instruments (keyboards, strings…) and finally the vocal tracks. Click here for a dedicated tutorial on how to Create a Template for your DAW.

2) Step Two - Group Channel Tracks: now that we have all of the single tracks in order, we will find that we have, for example, four guitar tracks that shares the same settings, the same eq and volume, and the same effect chain, four vocal tracks, ten drum tracks and two bass tracks. Instead of effecting and applying the same settings on any single track, multiplying the Cpu load and the the waste of time, we can create Group Channel Tracks in order to apply the same settings to all the similiar tracks that we have, for example we can create a Vocal Group, a Drum Group and so on, and we can even create Sub Groups that goes into other Groups. If you Mute or Solo a Group Channel, all the tracks routed on it will be affected. If you need, you can also Link Channels, so that if you move the Volume Fader, Solo or Mute one channel, it will affect all the linked ones. Click here for a dedicated tutorial on how to use Group Channel Tracks and How to Link Channels. 

3) Step Three - Editing: Now it's time to cut every background noise in between of the played parts in the audio tracks of our project, since noise it's useless and harmful, especially in projects with many tracks: the noise present on every track will sum up, to the point that it will be really tedious. Once we have got rid of the noise, it's time to do some Time Editing: we can improve the timing of anything, from drums, to vocals, to the single guitar strums, by cutting the track at the Zero Point, which is the 0db area before the start of a sound, and moving the area forward or backward in order to match the other instruments (usually the drums). If you select the "Snap to Grid" button, the cursor will snap the selected sound to the MIDI grid, which can be divided in 1/8, 1/16, 1/32 and so on, in order to be more precise for the positioning. If you feel that for example the vocals on the first chorus are way better than the ones of the second chorus, you can copy and paste the first chorus vocals on the second too, in order to improve what cannot be recorded again. For more info, check out the dedicated article about Editing!

4) Step four - Pitch Shifting / Autotuning: Once everything is perfectly organized and routed, and all the audio tracks are finely edited and in timed one another, it's time to check out the tuning, especially of the Vocals, but also for example for Bass or Guitars (although tuning software works well almost exclusively with Clean or Acoustic Guitars). For more info, Click Here to read the dedicated Article about Autotuning!

Once you have found the layout best fitting your needs, you can also Save it as a Template, so that your next projects will be even faster and smoother!

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HOW TO MIC A GUITAR AMP (a guide for dummies)

Hello and welcome to this week's article! Today we're going to talk about how to get a good tone miking a guitar amplifier.
First off let's say that miking a guitar amp is not particularly difficult, but it should be done on an acoustically treated room, otherwise, if you want to microphone an amplifier, especially a tube amp which needs to be pushed at a high volume in order to achieve that "roaring" sound, your room will be crowded with reverbs, and the final result will be a "live on a cathedral" type of sound.

Let's now talk about microphones: you can use dynamic microphones or condenser ones, which are more sensitive, delicate and faithful in terms of higher and lower frequencies, more suggested for acoustic guitars; my suggestion is if you want to use just one microphone, to use a dynamic one, which is able to handle higher pressure levels, so it's better suited for rock and metal guitars, but if you want you can also use both dynamic and condenser in order to achieve a more harmonically rich tone (of course anyway you could experiment by using as many microphones as you want: rumor knows that Metallica used 9 or 10 mics to get the "...And Justice for All" guitar tone, and if you find that you don't need one, you can just delete it later on your Daw). Two dynamic microphones which are the industry standard for both live and studio sound are the Shure SM57, which features a very strong eq curve made to cut throuhg every mix, and the Sennheiser E906, which under some point of view is even better and more faithful in terms of reproduction of the incoming sound. Those two microphones represent the two standard ways of microphoning: the Shure SM57 and similiar microphones are directional ones, which means that they take the sound they're pointing at, while the Sennheiser E906 is of the "Lollipop" kind, which means they're flat and can be hung on the cab directly by the cable, so they eliminate the need for a mic stand.

How to get the right tone? The idea is to use your ear instead of the microphone, getting close with your ear to the speaker (use earplugs if needed) and moving your head as if you would place a microphone, while you, or someone else, play the guitar: that is the sound that is going to be grabbed by the microphone, so move the equalization and gain control of the amp accordingly, keeping in mind that on record usually is needed LESS gain than you think to obtain even the heaviest tones, preserving attack and clarity.

How to place the microphones? For a "Lollipop" style microphone, just hang it from the top of the cab, towards the center of the speaker. the more you move it off the center, the less treble it will catch, and the more the sound will be oriented towards the mid-lows. Find the "sweet spot" and you're done :)

For Directional microphones, instead, things gets a little bit trickier, since you can also decide the distance from the speaker, and wether to keep them on axis on the speaker or to put them off-axis, and of which amount. Here is some example of what you may obtain:

- On axis, 2,5 to 5 cm of distance, center of the dustcap: maximum presence, attack and high frequencies.
- On axis, 2,5 to 5 cm of distance, at the edge from the dustcap and the cone: a bit less attack and more bass.
- On axis, 2,5 to 5 cm of distance, halfway between the center and the edge of the speaker: a more balanced sound, compromise between attack and low-end.
- Off axis, 2,5 to 5 cm of distance, 22° or even 45° towards the dustcap: a mix between the sound you would get going on axis towards a point between the center and the border of the speaker, and the one you would get pointing at the dustcap.
- On axis, 15 to 30cm from the speaker, center of the speaker: Medium Attack, fuller sound.
- On axis, 60 to 90cm from the speaker, center of the speaker: Softer Attack, reduced low frequencies.

Move the microphone closer or farther the dustcap until you find the sweet spot that allows you to get a tone that holds all the tonal characteristics that you need!

Additional Awesomeness: you can also use a combination of two dynamic microphones: one pointing towards the speaker using one of the techniques listed above, the other pointing THE BACK of the cabinet. So now you'll have two tracks of the same sound, and if you listen them in mono (or check them with a SPECTRUM ANALIZER) you'll surely notice some phase problem: some frequency of one track will cancel some other frequency of the other. Try flipping the phase of the track recorded to the back of the cab, and choose the one that cancels less frequencies, this technique, although may need some practice to be mastered, may help you thickening your tone!

A very common microphonig technique with two dynamic microphones consists into using a Shure Sm57 to catch the low mids from the cone, and a Sennheiser MD421 for the higher frequencies, this combination creates a very complete and usable spectrum.

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HOW TO USE IMPULSES (free VST plugins included)

Hello and welcome to another post of this blog! Today we're going to talk about Convolution Impulses. 

Convolution impulses are digital simulations of the reverberation of a physical or virtual space. 

Why do we need them? Because today those are the best replacement for guitar cabinet simulation. In facts, you may have noticed that the main issue of Guitar Amp Simulators (both hardware, like Pods, or software, like Amplitube) is the cabinet simulation and microphone response: they sound harsh, fizzy and plastic, because although they are impulse-based, their response is not accurate enough.

The web nowadays is full of impulses free to download and to experiment, and many of them are really accurate and realistic, that is the reason why many studios use them instead of microphoning a real cabinet. They are great for Reamping too, and can be used with any kind of preamp (a preamp vst, a digital multieffect or a real tube head, through its preamp out), taking the place of power amp, cabinet and microphone.

Just connect the preamp/pedal output into your soundcard, load one of the programs made to play them (for example on this guide we're gonna take a look to VOXENGO BOOGEX), load an impulse and play: if you hear a normal tone, with no harshness/fizzyness, you're done.

WARNING: Do not connect the power amp output of your head to your audio interface! Use the preamp output or the Fx-Send. If you have a tube power amp, you have to connect it to the cabinet anyway in order to prevent tube damages! 

Here's how to load an Impulse on Voxengo Boogex: 

First off turn off the cabinet simulation of your software or hardware (for example, with Amplitube 2: go into the cab section and activate the BYPASS switch you see in the upper right corner, or with the Pod Hd just take the cab icon away from the chain of your preset, setting it to No Cab), if you have bypassed the cab correctly, when you play a distorted preset you'll hear an awful sound, fizzy as hell, no bass, harsh... You're hearing how an amp sounds like without a cab. 

Now load Voxengo Boogex (I chose to explain this among the others because it's free, not too heavy for the ram and with zero latency, but there are many others around for free, as KeFir),
Boogex is an amp-sim, if you want to use it ONLY for cab simulation, you have to set all the parameters FLAT, or it will change your eq/sound.
So, open it and set, from the left: Lows Mids and Highs to 0db, Tone to 0%, Drive to 0db, Dynamics to 0%, Phase to 0%, The "Out" control it's the volume control, Dry Pre Cab and Dry Post Cab to "-inf" (knob all the way down).
The "Speaker Cabinet Impulse Response" button MUST be ON, obviously, then click on "file" and select your impulse (a .wav file).
Another interesting thing about Boogex: this software features a built in Low Pass Filter and High Pass Filter. Since they can't be totally bypassed, the best thing to do is to open them to the max, by clicking on the green point in the right area of the graphic section of the interface and move it all the way to the right, then click on the white point in the left area and move it all the way to the left. This will deactivate the filters.

Here are some great free impulses to load:

Andersson - Periphery IR

The Metallica Golden Ir Pack

The Guitarhack Impulses

If you have or know some other interesting impulse to suggest to our readers feel free to tell us!

Additional Awesomeness: some speaker simulator plugin (such as Poulin's LeCab2) gives you the opportunity to load more than one impulse and blend them together, as if the speaker is microphoned with two or more mics. This opens up a very interesting world of variables, if carefully managed, but can also harm your final result, so use it carefully :)

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HOW TO RECORD 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. 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 V70 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, 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!!

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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.

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 (for example, the good RONALD PASSION BASS PREAMP, or Ignite Amps - SHB-1, a Tube Bass amp suited for Rock and Metal, or AcmeBarGig - BIG, another interesting Bass Amp Sim), 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 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, on the Mixing and the Mastering phase.

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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 ;)


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