REVIEW: IBANEZ V72Ce

Hello and welcome to this weeks'article!
Today we're going to review an interesting budget acoustic guitar, produced by Ibanez: the V72CE.
These are the cheapest acoustic guitars in the Ibanez catalog, and are single-cutaway dreadnought guitars.
Dreadnought guitars takes the name from a large english battleship, and this name stands for instruments with a larger body than most of its contemporary guitars, and with a deeper, louder tone.

These guitars feature an interesting in-built preamp and tuner, called AEQ200T. 

The preamplifier has three faders: Bass, Treble and Volume, and obviously an output jack, good to go on a guitar amp or straight to the P.A.

The direct output of this guitar was used to record the sample that can be listened on our article about How to Record and Mix an Acoustic guitar.

Tech Specs (taken from the Ibanez Website):

body shape: Cutaway Dreadnought body
top: Spruce top
back & sides: Mahogany back & sides
neck: Mahogany neck
pickup: Ibanez Under Saddle pickup
preamp: Ibanez AEQ200T preamp w/Onboard tuner
tuning machine: Chrome Die-cast tuners
bridge pins: Ibanez Advantage™ bridge pins
bridge & fretboard: Rosewood bridge and fretboard

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HOW TO RECORD A STRING ENSEMBLE!

Hello and welcome to this week's article! Today we're going to talk about a very particular topic, that is not often encountered in a home recording environment but which deserves an in-depth analysis, nonetheless:

How to record a string ensemble? And more in general, how to record an acoustic ensemble that we're going to insert in our mix, with lead and/or ambience role?

I had the opportunity to record a string sextet (violins and cello) called to beef up a rock song, and its role was to give a realistic feel to what before was a synth strings part, so I had to deal with a complete equalization spectrum that needed to be recorded in as few tracks as possible (to close-mike any single element would have been quite useless).

Surfing the web there are many ways to microphone an acoustic ensemble, especially a string one, but here's my own way, developed by experimenting different mic placements.

I have placed two micro-condenser microphones, of the kind often used to track the overhead cymbals of a drumset, with the X/Y method, in order to avoid phasing issues, and each microphone was pointing to the center of its half of the ensemble, trying to place them as symmetric as possible.

Then I had the players with the cellos to sit as centered as possible, and the violins, which are higher pitch instruments, on the sides (remember, the lower frequencies should be as mono as possible, while the higher ones can be placed wherever we want, as explained in this article).

Finally, I have placed a Neumann large-condenser microphone at the center of the ensemble, at the height of the "F-holes" of the Cellos (no pun intended), which are located much lower in height than the violins, in order to make it catch specifically the sound coming from these instruments. 

The idea, as you may have already guessed, is to create a frequency separation between lows and highs for panning purposes, and to add some reverb and "room" to the sound catched from the panoramic microphones, since the ensemble has been recorded on a "dead" room, so there was almost no natural reverb, which is actually very useful when recording acoustic instruments.

Now we should have 3 tracks: left, right and center, which are ready to be mixed!

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HEADROOM AND GAIN STAGING! a guide for dummies PART 2/2

CLICK HERE TO READ PART 1/2 OF THIS ARTICLE

In the first part we have seen how to optimize the signal-to-noise ratio (click here for an in depth article) while preserving headroom when recording audio.
What do we mean with signal-to-noise ratio? It's the ratio that measures how effectively we have recorded our signal: below a certain level we have nothing but noise, so if we record a sound with a too low input gain it will be recorded very low, and if we will raise the volume fader of our DAW at a level sufficient to make it audible, we will have also the background noise raised significantly.
The ideal therefore, would be to have the sound tracked at an input gain high enough to make it be acquired from the DAW at the optimal distance from the background noise, so that we won't need to raise the track volume excessively, therefore the noise won't be heard.
On the other hand the signal must be far enough from clipping, otherwise we'll have to deal with unwanted distortions, that are even harder to treat than background noise.

During the Mixing Phase, as we have already seen, we'll have to deal with the single tracks making sure that everything is euphonic and perfectly audible, and to do so we can process the single tracks, process them in Groups and/or process the whole mix buss as it is already one single track.
The Mix Buss (also called Stereo Buss) is the DAW channel in which all the single tracks pass, and its volume fader affects all tracks at the same time, so we must be very careful when loading plugins on the insert of this track, because its effects will be cascading on every track, especially when talking about compression.
Mix buss compression in facts will affect each track and will sum with the single tracks compression, so we must keep it very low with settings, just to attenuate the occasional peaks, and lower accordingly the settings of the compressors on the single tracks to not drain excessively the life of the sounds: we must not forget in facts that we're going to compress and limit the sound rather heavily in the Mastering phase too.
Everybody has a different vision about Mix Buss compression; in my opinion, I prefer to compress to perfection the single tracks so that, unless I'm not working on a live recorded track with strange dynamic problems, there won't be need to attenuate and normalize peaks on the mix buss, leaving the general sound a bit more dynamic.
The overall sound that passes throug the mix buss should never be higher than -10 dbFS, that way there will be enough headroom to work on the exported tracks in the Mastering phase; if the track volume is already too high, very little mastering improvements can be applied, so we must lower the master volume until our peaks are in that area.

Once we have everything on its place, whether the peaks are compressed on the single tracks and on the mix buss, and the volume is automated (this is another solution to avoid excessive compression that can harm the tone), it's time to Master the song.

In the Mastering Phase we can greatly improve the song or completely ruin it, especially with gain staging matters.
Let's start with the condition that we have perfectly respected all the rules exposed on this article: we have recorded everything at the proper levels (eg. -10dbFS), we have mixed with a transparent compression and exported the tracks at -10dbFS, pristine clean, balanced, with good dynamics (= not already squashed) and without distortions:  the ideal it's to load on the last slot of our DAW a metering tool such as the TT metering tool (which will also give us a judgement about the gain structure of our song, if it's gonna be ear fatiguing or pleasant) and proceed with some final compression.

In this phase we have 2 solutions: to use Multiband compression, which will help us in correcting also some mix problems, such as adding some low end if the general sound it's too weak, or to compress just the mix areas that needs to be processed bypassing the others, OR to use broadband compression (single band compression): this solution it's probably the best one if we're happy with our mix and don't want to change the overall balance (actually, 7 times over 10, to alter the mix in the mastering phase will result in a disaster, if we don't know EXACTLY what we're doing).
To compress in the mastering phase can make us lose some Transient, in this case we can use a Transient Shaper to bring back some of the wave parts we are squashing (but the ideal would be to squash as little as we can, so that we can avoid this remedy), and if we have used mix buss compression but we feel the mastering compressor is harming our sound, my suggestion is to go back in the mixing project and to remove the mix buss compressor: the mastering one will probably sound more transparent and pleasant, and in my opinion it's more important than the mix buss one.
Usually a mix buss compressor in facts is used as an alternative to the mastering compressor we can use to mix hearing an already compressed sound, so that we can compress less the single tracks, and it's rarely a good idea to sum up single track compression, mix buss compression, mastering compression and limiting altogether, as the TT Metering tool will probably tell us.
My suggestion is therefore to choose between Mix Buss compression and mastering compression, not to use both of them, but this is not a general rule: many producers thinks that it's a better idea to use both.

Once we have prepared the mix by lowering the peaks with the mastering compression, it's time for the ultimate gain processor: the Limiter. This is the wall in which all of our tracks will be pushed, and the more we will push them by lowering the threshold and raising the ceiling, the more the volume will raise, the more transient we will lose, and eventually the sound will start distorting.
Since today most of our music will probably be heard on Youtube or on some low quality mp3, it's a good rule to set the ceiling at -1.0db, it will prevent unwanted distortions, and the threshold should be lowered until we see a little of gain attenuation, -3/4db at maximum, because if we push more we will probably waste all the careful work we have examined in these two weeks, to preserve the gain structure of our mix and to result in a dynamic, clear, pleasant song.


CLICK HERE TO READ PART 1/2 OF THIS ARTICLE


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HEADROOM AND GAIN STAGING! a guide for dummies PART 1/2

Hello and welcome to this week's article!
Today we're going to talk about Gain Staging!
Gain staging means finding the right level for our tracks when recording, mixing and mastering, in facts we must think of all those different phases as parts of one single workflow, therefore the decisions we make in the beginning will be crucial and increasingly relevant further in completing the job.

Why do we need to watch carefully the gain structure of our mix? It's simple: with any modern digital audio interface (even the cheapest) we have plenty of headroom, which means that we can record at not too high levels, in order to preserve all the dynamics and the transient of our instruments; then, in the mixing phase, we must decide how much to compress the single instruments and wheter or not to use a Mix Buss Compressor.
In the Mastering phase we must decide wheter to use a single band or multiband compressor (or more than one), and finally a Limiter.

All those choices will impact on the final sound of our mix: it can sound very natural with a lot of headroom (with "headroom" we refer to the free space among the highest peak of a track and the ceiling of the full scale of decibels before clipping) or on the opposite side extremely compressed, on the verge of clipping, and sometimes it's also hard to go back finding what we have done wrong to obtain this unpleasant result.
My suggestion is to use these tips from the early stages of our production up to the latest steps of mastering, in order to have clear in mind what could go wrong and avoid it.

First off, the Recording Phase. When recording, we must make sure (using the Gain knob of our interface) that there is enough incoming signal, which means that for example the gain knob of the preamp of our audio interface is set properly: when recording a source, the ideal peak would be -20 to 10 dbFS (decibel full scale, the level before the maximum limit), as you can see in the orange circle on the picture above.
A lower level could translate in loss of data, since, some sound could be so low to fall below the noise level.
The peak of dbFS is the one in the orange circle, and we should adjust the gain knob in the audio interface to make sure that the peaks of our instruments stays on that range, without touching the virtual fader of the DAW. We will move it later on, in the mixing phase.
Plus, everything we put between the source (instrument or microphone) and the preamp will increase the noise, so it's better to place every other processor after it.
Before completing this phase it can be good to take an additional step: Trimming.

In the Mixing Phase, once that we have all instruments recorded at the right level it's time to make everything sound properly, and as we have already seen there are many tools of the trade, but the two most relevant ones about gain staging are Compressor and track volume fader.
The idea would be to use a compressor on the tracks with the highest volume excursion in order to lower the peaks (for example if we have a song with a vocal track in which the singer alternates quiet singing to loud screams): a properly configurated compressor would lower the higher peaks, leaving some headroom to raise the volume fader, in order to make the quieter parts to be heard better.
If we wouldn't compress, by raising the volume fader we would make inevitably the screams to distort.
I like to mix my projects at -12db to -10db (someone mixes at -6db, but sometimes this leaves not enough headroom for a powerful mastering)
An important rule when mixing is that if we have more than one plugin on our track and the final gain is too high, we should go back to the first processor (for example a compressor) and lower it, it's not a good idea to adjust the final gain with the gain knob of the final plugin, it will damage the gain structure of our sound.


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ROCK/METAL DRUMS MIC PLACEMENT!! PART 2/2

Pic. 1

CLICK HERE TO READ PART 1/2 OF THIS TUTORIAL!

Hello everybody! Once we have set up properly the microphones for Kick and Snare drum, it's time to microphone the Toms.
As you can see on pic. 1, I have drawn a red circle on the "blind spot" of the microphone to explain their positioning: since those are all dynamic, directional cardioid microphones, they take only the sound coming from a certain direction, as explained on pic 2:

Pic. 2

"Cardioid" means "heart-shaped", because this kind of microphone takes only the sound source that stands in front of them, and their sensitivity is depicted with a shape similar to a heart. 

This is to let the microphone to have purposely a "blind spot" behind them, so that the sound source placed right at the opposite side of the capsule is almost completely ignored. 

We need to use this property with strategy when miking drums, since the sound sources are many and located everywhere on the set.

The mic positioning on pic. 1 shows that I have chosen the closest and noisiest Cymbal, and placed the tom microphone with its "blind spot" (or we could call it more appropriately "deaf spot"), shown with the red circle, pointing to the closest Crash Cymbal, which is the thing that it's most likely to bleed into it. 

As for the Snare top, the right positioning of all tom microphones it's the one with less bleed, as close as possible to the skin, pointing exactly where the stick will hit. 

Pic. 2

Pic. 3

Moving to the last essential (in my opinion) microphone of our drumset (except for the Cymbal microphones, that we have already explained HERE), we need to talk about the Room Microphone. 

The Room microphone, shown at the centre of Pic. 2 and alone on Pic. 3 it's a Condenser mic of the kind used also for Vocals, which should give us a sound as "global" as possible: we need to find a placement that if we listen to its track alone will give us a complete, balanced reproduction of the whole drumset, so if for example the cymbals are covering everything on this track, we should lower it, but if we're hearing only the kick, maybe it's too low :)

We should just move the microphone and hear what it's capturing, until we're hearing everything decently.

We will use this microphone to add realism, fatness and natural reverb to the drum track, since we have close-miked every single part, and the final sound may result a bit dry and sterile. 

Once we have all drums, room and cymbal microphones set up properly and we like the result (rememeber, the original sound is 60% of the work, we can make it better in the mix, but if the original sound doesn't sound right, neither the final mix will), it's time to take a single, clean hit of every drum part: first off hit the snare, then when the tail is completely over, hit the kick, and so on, so that we have a single hit ready, because in the Editing Phase it can be useful to have a clean sound available to replace some drummer error, as a weak hit.

Then, once everything is recorded properly we can move into the Editing Phase, if needed, and then we're ready to mix!

Summing it up, the list for a good drum microphoning is:

2 dynamic mics for the snare (a good choice is a Shure Sm57, or a Beta 57)

1 dynamic mic for each tom (so usually 3, the ideal would be a Sennheiser Md421)

1 dynamic mic for the kick drum (for example a Sennheiser E602)

2 Condenser mics for the overhead 

1 Condenser mic for the room

1 Dynamic or Micro Condenser mic for the Hi Hat

1 Dynamic mic for the Ride Cymbal (if used often, otherwise this mic is unnecessary, the overheads are enough to catch it).

We also need an Audio Interface with at least 10 Mic Preamplifiers. Often the Usb audio interfaces offer 8 mic preamps and other unbalanced inputs, so we must take another "2 channel mic preamp" and send it to 2 of the unbalacend ins of the Interface in order to have a total of 10 mic pres, or we can use a mixer such the Alesis Multimix usb 2.0 or the Phonic Helix, that sends all the separate tracks to the DAW.

Hope this was helpful!

CLICK HERE TO READ PART 1/2 OF THIS TUTORIAL!

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ROCK/METAL DRUMS MIC PLACEMENT!! PART 1/2

Pic.1

Hello and welcome to this week's article! 

Finally here we are, on the toughest of all recording topics: how to mic a drumset!

Let's start by saying that there isn't just one good way to do it, there are many ways that can be discovered on the various studios, and the philosophy varies according to the music genre, the room treatment, the number and the brand of microphones used and so on.

In the beginning the first recording were made with just one Condenser Microphone in the middle of the rooms, and all instruments were tracked with the whole band playing, then with the perfectioning of the recording techniques, audio engineers felt the need to increase more and more the separation among the single instruments, in order to be able to mix them individually and obtain different tones without modifying the other tracks.

The most common microphoning technique today consists into tracking every drum piece with a microphone or two, and the cymbals divided in groups (Click Here for a dedicated article about How to record and Mix Cymbals), and before getting started we really recommend you to do some acoustic treatment to your room, for example placing some absorption panel around the set to avoid unwanted resonances, and to TUNE your drumset in a coherent way.

If you have only 2 microphones and you need to mike a whole drumkit, use this technique.

Let's start with the Snare Drum: we need to microphone this one with a microphone that can capture its snap and brightness, such a Shure Sm57 (or Beta 57), and we should place it in the centre of the drumset (in Pic. 1 the snare is the one in the lower right, because the drummer was a leftie), as close as possible to the skin (but avoiding it to be touched even when the skin gets hit hard), pointing to the exact point where the drummer will hit. 

Why are we placing it on the centre of the drumset? To put it as far as possible from the Hi-Hat (aka Charleston), that usually bleeds heavily in the snare microphone, and we need to avoid this as much as we can.  

Now we will need a second Shure Sm57 to catch the sound of the Snare Wires on the bottom of the Snare: we'll just place it below the snare pointing up, and this will provide us some extra "splash" that if we'll mix carefully with the Snare Top sound will give us a very full and powerful result.

Pic. 2

For the Kick Drum instead we will need a dedicated microphone that is particularly suited for tracking bass frequencies, and every brand has some model made for this task, for example the Sennheiser e602, but first it's a good rule to set a pillow or a blanket inside the Bass drum to reduce the resonance (we don't really need Mondo Bass). To microphone this drum part it's important to find the sweet spot that captures as many frequencies as possible: if we set it too "outside" the hole, it will sound excessively "bassy" and without attack, if we place it in too deep it will only take the "click" of the kick drum beater, so we'll need to find a halfway positioning that will reproduce everything as close as possible to the final sound that we have in mind (in our case, the right position was obviously the one shown in Pic. 2).

Sometimes may also be useful to place a Sub Kick near the hole, in order to capture some extra boom, or another microphone pointing to the kick beater to capture some extra attack, but in this case we were satisfied with the sound we obtained, with just one mike, so there was no need to use another preamp channel.

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

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PHASE COHERENCE IN MULTIPLE MIC PLACEMENT!

Hello and welcome to this week's article!
Today we're going to talk about Phase Coherence, a problem that anyone who has recorderd an acoustic drumkit or a guitar cab with more than one microphone, has already had to deal with.
When miking a snare or a speaker with just one microphone, we don't have to worry about phase, because it's just listening to the source by itself, but when we use two or more microphones on the same source, it's important that the sound hits the capsule at the same time, because if it doesn't, the frequencies taken by the two microphones will cancel (partially or completely) each other out.
Phase can therefore be described as the time difference between two microphones in hearing the same waveform.

Let' make an example: let's record with two microphones a snare drum or a guitar cabinet: we'll end up with 2 mono waveforms.
Now we must zoom into the waveforms very close to see if the two waves are "in phase" one with the other.
To check if they're in phase, they must look as the image below:

If they're in phase, the two waves will help each other in reproducing the sound as precisely as possible, if they're partially or completely out of phase instead, they will cancel some frequency or even cancel completely each other (this is almost impossible, though).

The solutions are to move one of the microphones slighly closer or farther to the source in order to align the phase better, if we have just a sligh, partial frequency cancellation, or to flip the phase with our DAW, if the tracks are completely out of phase.
Usually we need to flip the phase when we record a source with two microphones on the opposite sides, especially when miking a snare drum with a microphone at the top and another one at the bottom of it: in this case it's very likely that we need to flip the phase of one of our two tracks.
To flip the phase, usually in almost every DAW is present a control, with this symbol:

By activating it, the phase of the track will invert, and the waves will sound much fuller, without any unwanted cancelling.
It's important to check the phase coherence of the whole drumset when miking an acoustic drumkit, by trying to flip each track, one at the time, and see if we notice an increase or decrease in tone, to get the best reproduction achievable.

This article must not be confuse with phase issues that may emerge in the mastering phase, and which are relative to the single, final stereo track. Click here for the dedicated article.


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