REAMPING: How to reamp a guitar track (part 2/2)

CLICK HERE FOR PART 1/2

Following the instructions of the first part of the article, the setup should be ready: we should have a nice DI track, recorded at the right level, and all the routing in the DAW and in the sound interface software should be correctly set up: our guitar track is ready to be reamped!

4) Now we need to go from the audio interface output we have chosen for the DI track (in our example we have choosen the output n.3) into a REAMP BOX. What is a reamp box? It's a box that does exactly the opposite of a DI box: it takes a balanced signal (the one that comes out from the audio interface) and turns into an unbalanced one, meaning one at the right level to be fed into a guitar amp input (if you want you can also pass through pedals etc. before entering the amp, but remember that the signal comes out from the reamp box usually with more noise compared to if it would come straight from a guitar). Once the reamp box is plugged into the amp, we need to adjust the output level of the box in order to be the as loud as if it would be coming directly from a guitar.

5) Once the cables are plugged, it's time to press play on the DAW and let the song go: if we have done everything correctly, from the DAW monitors (or headphones) we should be able to hear the whole song, and from the amp we should be hearing only the correct guitar track. 
This is the moment in which we find the right tone in the amp, so let's take our time in finding the right gain, eq, effects etc.

6) The last step is obviously the microphonation one: once we have a tone we like, we can start having fun, trying out all the microphones we have until we find the best combination and positioning among them (click here for some ideas), and the only remaining thing to do is to press record and enjoy our reamped track!

I hope this was helpful!

 

CLICK HERE FOR PART 1/2

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

REAMPING: How to reamp a guitar track (part 1/2)

Hello and welcome to this week's article!

Few years ago we have done an article about reamping (click here for the main article): now it's time to see in detail, step by step, how to reamp a guitar track.

The first thing to lear is how to route your channels to make sure that from the output channel of your audio interface only the guitar di signal comes out, while from your headphones or monitors you can keep listening to the whole project.

First off: you need to have in your project a DI guitar track, then, since usually all the tracks go to the stereo out, we need to steer this one away from there and make it go to a separate exit of our DAW and audio interface, an exit in which only our guitar track will be.

Today we're using the Presonus Studio One interface, but it should work more or less like this in every DAW: 

1) Go to song setup -> inputs and outputs and create a new mono output besides the standard stereo one (I've called it reamp, and the M there stands for Mono).

I have chosen "LINE 3", which means that when I assign the DI track to the output named "reamp" it will be listenable only by plugging the headphones to the output n.3 of my audio interface.

2) While all the tracks are assigned by default to the main out, meaning they will all end up into the stereo out buss (in the pic it's called "principale", because my interface is in Italian), I have changed the out for my DI track into "reamp", the new output we have created.

3) now we need to assign this "reamp" out to the physical out n.3 in my audio interface (you can assign it obviously to any out you want, just make sure the DAW and the output you're using in your interface are matching), and in order to do this we need to open the control panel of our audio interface, in my case it's the Saffire Mix Control from Focusrite.

In this case I assign "DAW 3" (which means the output to which we have routed our DI track in the DAW, which as you can see in the first picture is the out called "line 3") and we assign it in the "line output 3" slot (red arrow in the bottom of the pic), and we also assign it to a channel in the virtual mixer (red arrow in the top).

Now if everything went according to the plan, if we plug the headphones to the main headphone out of the audio interface we should be able to hear all the tracks going into the stereo buss EXCEPT the guitar DI one, while if we plug them into the out n.3 we should be able to hear only our DI track.

Once this complex preparation phase is done, it's TIME TO REAMP!

CLICK HERE FOR PART 2/2

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

Review: Harley Benton G112 Vintage

Hello everyone and welcome to this week's article!

I was looking for a small 1x12 cabinet for my travel setup (which consists in a Boss Katana 100 head), and my idea was to look for one with a Celestion Vintage 30 speaker, which is a standard for rock and metal.
The V30 speaker is particularly good for hard music because it can handle high wattages (60w nominal power rating) and because it provides a strong low-mid thump and mid range, which is particularly suited for lower tunings and palm muting.

While browsing around for cabinets, I've stumbled upon the Harley Benton G112 Vintage, a cabinet made in China and imported from the German Shop/Distributor Thomann, and given the surprisingly low price (lower than the price of the speaker itself if bought separately!) I have decided to give it a try.

The cabinet is black, of the size of the average 1x12 combo amp, not particularly light nor heavy, and with a half-open back to retain a bit more low end, which is useful since it's quite small.
On the back there is only one input, 8 ohm mono, which makes it not very flexible but it's the most common choice and it's supported by basically every amp, and in general this cabinet has a pleasant look and feels solid in build quality. 

How does it sound?
It sounds quite well! The speaker has the unmistakable V30 tone, and the Katana head roars through it with no problems, plus this setup makes also a very good companion for home recording, because the Katana head can be used also at lower wattages, and the cabinet lends itself nicely for any type of microphoning, also at low volumes.

Do I suggest it? Hell yeah! At this price it has basically no competitors, and I don't see any reason not to buy one.

Thumbs up! 

Specs taken from the website:

- Equipment: 1x12" Celestion Vintage 30 speaker

- Power rating: 60 W

- Impedance: 8 Ohm

- 18 mm Poplar plywood housing

- Half-open rear wall

- Rearloaded

- Trim strip

- Carrying handle

- Dimensions (W x H x D): 460 x 470 x 299 mm

- Weight: Approximately 14 kg

- Cover: Black Tolex

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

Different types of microphones for guitar and common combinations among them

Hello everyone and welcome to this week's article!

This time we're going to check out the various types of microphones we can use to mic a guitar amp, and this article can be considered as a supplement to the basic one "how to mic a guitar amp".

Assuming that you have read our basic article and you are familiar with how the horizontal distance from the dustcap of the speaker can make the tone brighter or darker, here are 3 common mic combinations that you can try, it doesn't matter the exact microphone model you have (for example whether the condenser one is small or large diaphragm: the sound will be different, but the basic concept stays the same). 

First off: why to combine two microphones? Because every type of microphone has a different eq curve, curve that changes also according to the position from the speaker, so it happens often that one single microphone is not capable of capturing a tone that is full and has for example a detailed high end and a full low-mid area: most of the times trying to make everything with one mike leads to a compromise that can be good, but that rarely can be perfect in every aspect.
Using two microphones therefore allows us to use one for the low-mid area and one for the high end, and we can also use the faders in the DAW to choose the right balance.

Second note: every microphone type has different requirements, for example a ribbon microphone is fragile if left in front of high sound pressure, the ribbon can bend or break due to the air movement, so you need to use a volume that is not too high if you are using it for close miking, or a condenser microphone needs phantom power, but you need to make sure that the phantom is deactivated in the channel of the ribbon one, otherwise the ribbon microphone will break.
It's a good idea in the studio, when using condenser and/or ribbon microphones not to crank the amp volume too much, it's sufficient to arrive to see a little bit of movement in the speaker.

Dynamic + Condenser = this is a popular choice both in modern music and in the '70s one: the dynamic microphone should be placed straight or angled, mid way between the dustcap and the edge of the speaker and its role is to pick up the mids and low-end: the more it's pointed towards the edge of the speaker, the darker it will get. The distance should be a couple of centimeters from the grill cloth. 
The Condenser microphone instead will take high end, so it should be pointing towards the center of the dustcap, and its distance should be regulated according to the mic sensitivity: if it's very sensitive it's better to keep it 20-30cm from the speaker, maybe even 50, while if it sounds too thin or you hear that there is too much room in the track (and if the amp volume is not too high), it can be put as close as 5-10cm. If you feel like the tone capture by the condenser mic is clipping, lower the gain in the audio interface and/or back it off a few cm. 

Dynamic + Ribbon = this tone was used a lot in the '80s and produces a warm tone with a nasal mid-range (for example imagine a Guns'n Roses type of mid-range), the most classic technique is to put the ribbon microphone 2 to 10cm from the grill cloth pointing towards the dust cap, and the dynamic one right on the side, so that it points towards the edge of the dustcap, at around 2 cm from the grill cloth. This will create 2 complementary tones, with the ribbon microphone that is more dark and nasal to provide the body (but also part of the high end) and the dynamic one to bring more detail in the high end. 

Condenser + Ribbon = this is a less used technique but it's pretty interesting: the ribbon microphone placed as in the Dynamic + Ribbon technique, but pointing a bit more towards the edge of the dustcap (so the sound is even meatier), and the condenser one placed like described in the Dynamic + Condenser technique, to take all the detail in the top end. This technique is a bit more complicated but if handled well it can create very good results.

Finally, it's important when doing mic placement to check the phase coherence in order to avoid cancellations! Click here for a dedicated article.

And you? Do you know other good microphoning techniques? Let us know in the comments!

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

Ribbon microphones: what are they and how they work

Hello everyone and welcome to this week's article!

Today were going to talk about ribbon microphones.
Ribbon microphones are a a particular type of microphone that use a thin ribbon of a conductive material (for example aluminum) between the poles of a magnet to record the sound, and this particular construction allows it to sound very pleasant, round and sensitive to certain frequencies, but at the same time makes it quite fragile, therefore it needs to be handled with more attention than a regular dynamic mike.

Ribbon microphones were first introduced in the early '20s of the past century, and they were praised to be the type of microphone more suited to reproduce the whole spectrum of human hearing (20hz to 20khz), and with the time, as per the other types of microphones, technology evolved and today we can count on microphones that are less fragile than before (although it's still importato to observe caution when handling them), made with more solid and durable materials, with various types of sensitivity pattern (cardioid, hypercardioid, variable, uni-directional, bi-directional and so on), active and passive.

The active ones work more like condenser microphones, meaning that they need an energy source in order to work (never use the phantom power on a passive ribbon microphone or some internal component will break!!), while the passive ones work like dynamic mikes, and are used often to microphone brass instruments and guitar amplifiers.

Speaking of guitar amplifiers, there are 2 main techniques which are very popular in recording studios, and that involve both a ribbon microphone (such as the famous Royer R-121) and the omnipresent Shure SM57: the most famous is the one in which the ribbon microphone is in vertical in front of the center of the dustcap, next to the speaker grill (to take the high end of the tone, since it doesn't sound too brittle or harsh), and with the SM57 pointing towards the edge of the dustcap, so it's slightly off the center and this way it captures a bit more the body of the tone.

The second technique is the other way around: the SM57 points towards the center of the speaker, essentially capturing the high end of the amp, and the ribbon mike is slightly away from the speaker grill (we're talking around 15cm) and off-center, so it will take more the low-mids part of the guitar tone.

Either way, it's essential to check out the position between the two microphones to minimize phase cancellation, so if you notice there are a lot of phase issues try moving one of the 2 microphones in order to align the phase, plus if the guitar amp volume is extremely high, it could be also a good idea to tilt slightly the ribbon mike off axis in order to prevent the sound pressure to hit the ribbon too frontally.

Obviously these are just a couple of the thousands of possible mic placements, but they are a good starting point when trying out a ribbon microphone.

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

How to randomize the velocity of a drum sampler to make it more realistic

Hello and welcome to this week's article!

Today we're talking a bit more in depth about a topic we have already mentioned in our article about MIDI dynamics: the fastest way to make more realistic a MIDI drum track, by randomizing the velocity.

One of the many parameters you can assign to a MIDI note is the Velocity: it's a value from 0 to 127 (like the other MIDI parameters) and it is meant to mimic the natural strength variations in our playing; 
basically it represents how soft or hard we hit a note, that can be a keyboard or any other instrument.

Now, when talking about virtual drumkits, nowadays there are on the market very extensive drum libraries, that arrive to dozens of Gb in size, and the larger is the library the more samples there are, also for a single drum part: a single snare, for example, can have 15, or even 25 velocity layers, that represent various intensities a drummer can hit it.

The more the velocity layers the more the sampler will sound realistic when playing it live, but even if we are writing down the drum MIDI parts note by note with the mouse, modifying the velocity it's quite important to make the drums sound less robotic, to the point that some drum sampler have also a "humanizer" function inside that randomizes a bit the dynamic variations in strenght of the hits.

Some DAW have directly the humanize function, for example in Studio One you just highlight the MIDI part you want to make more realistic, right click and you can choose "Humanize" or "Humanize Less" to reduce the effect, but the humanize function will also slightly move the MIDI notes in timing, so if you want only to randomize the velocity you need, while the same notes are still selected, to pull down the Action menu again and select Restore Timing.

In other DAW, where no humanize function is available instead, you can usually choose the MIDI part and under the Velocity section there should be a randomize velocity function that lets you dial in a minimum and maximum value, and the notes will be with a randomized velocity within that range.
What range to choose? It depends on 2 things: the genre (if the dynamic excursion is huge, like there are press roll parts, it's better to randomize them separately or write them note by note) and the number of velocity layers in the sampler. If you have some pre-made MIDI groove for that specific sampler you can load some and take note of the minimum and maximum velocity of each drum part, so you can be sure that the range you enter will have samples available, otherwise you can be conservative and choose quite a narrow range, in the area that sounds better to your ears (for example 70 to 90, or 80 to 100).

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

Review: Audio Assault Blacksun (with video sample)

Hello and welcome to this week's article!
Today we're going to review an amp sim from Audio Assault that is currently free if you are member of the AA mailing list: the BlackSun!

The BlackSun is an emulation of a Blackstar head, probably an HT Club 50 (even if it's not explicitly stated), which is a 2 channel tube head with EL34 tubes.
Among the other features, this plugin has a preset manager with several presets ranging from clean to hi gain, a "double track sim" which emulates recording 2 layers of the same riff, the "my amp" function which generates some slight hardware variation unique for every registered plugin, and it has a knob that is typical for Blackstar amps, (in the original amps is called ISF, in this plugin it's called Mode), which lets you choose between a "Marshall" and a "Mesa Boogie" voicing and all the gradients in between;
a "Marshall" voicing will mean a tone with more pronounced upper mid range, slightly more nasal and twangy, while a "Mesa Boogie" one will be a with a stronger low-end and more scooped mids, that lets out more high end than upper mids.

As per the other recent Audio Assault amp sims, also this one features a very nice, resizable GUI, 3 stompboxes (Gate, Booster and Tube Screamer), an FX Rack in the loop with Graphic EQ, Delay, Reverb and Chorus, a Dual Cab Loader which lets you also move around the microphones with any ir (thanks to a system of eq and envelope filters I imagine), and many other features.

How does it sound? It sounds very well, in my opinion, especially with the "mode" knob in the "Marshall" side, because a problem that I have found in some Audio Assault amp sim is the fact that they are all very "American sounding", for modern metal, while this one can replicate very well a British tone that is also hi-gain, tight and defined, and that it has that nice clear upper midrange.

This makes Blacksun one of the most versatile Audio Assault amp simulators, because the original head itself is extremely versatile, it has a lot of gain and tonal possibilities and I think anyone should definitely try it, also because they just released a pack with '80s style presets which adds a lot of fun to this amp.

Thumbs up!

Specs:

- 2 channels

- 3 stompboxes (gate, boost and screamer)

- Dual Cab loader with IRs created by Seacow cabs

- FX rack with eq, delay, reverb and chorus

- preset manager

- mode knob to change the flavor of the amp

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