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!

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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.

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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).

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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

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Using a wah in fixed position to boost a solo

 

Hello everyone and welcome to this week's article!

Today we're going to talk about a particular way to make a solo cut through, which is used by many musicians: to emphasize only a certain mid range area which is obtained by using a wah in fixed position.

Let's start by reminding what is a wah (click here for a dedicated article with free vst plugins): it's an envelope filter, which means basically an equalization filter that can be fixed or controllable in real time via a pedal control and that according to its settings will emphasize a certain area of the midrange of the guitar tone adding gain and harmonics.

A wah boosts and cuts a certain area of the midrange of a guitar to make it cut more through the mix, and when it's used with the foot control in a tasteful way it can really make the guitar almost talk, or cry like a baby (the famous Dunlop CryBaby is called like that for this reason).
This mid range boost, which is usually a rather narrow range between the 300 and the 3000hz makes the sound from dark to very bright and harmonic rich, but the most usual application when doing distorted solos is usually when it's kept around the central position, which means a boost around 1000-1500hz, which makes the tone very honky and recognizable.

This nasal tone is the core of the solos of many famous guitarists, which sometimes just set the wah in a certain way without even moving it, just to have that distincitve mid range, boxy boost when doing solos, and the tone obtained can be heard for example in many Queen, In Flames or Blind Guardian songs, bands very different among themselves which make extensive use of the wah both moving the filter in real time or leaving it in a fixed position according to the song, and this use of the wah also just in a fixed position as mid boost has led to the creation of several stompboxes which replicate just that: a wah with a fixed setting (for example the Dunlop CryBaby Q-Zone or the Magnetic Effects Midphoria).

When trying to recreate this type of effect in the studio when we don't have a wah at hand we can start with a not too steep high pass and low pass filter, between 300 and 3000hz, then we can draw a bell quite wide, around 1200hz and add some db in this area according to taste.
Obviously these settings are just a starting point, feel free to play with the filters amplitude and the position of the bell according to the tone you are looking for, and if you feel like you lose a bit too much sparkle try to be less aggressive with the high end roll-off or maybe to add a harmonic exciter.

I hope this was helpful!

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Audio Assault - Bulldog (with video sample)

 

Hello everyone and welcome to this week's article!

Today we are going to review the latest virtual amplifier by Audio Assault: the Bulldog!

The Bulldog is the amp sim of a VHT Pitbull, a 3 channel hi-gain tube head famous for being very tight and aggressive (and also very expensive).

This virtual amp is quite a faithful representation of the original one and offers several solutions to make it a very flexible suite: a tuner, 3 stompboxes (gate, booster and a Tube Screamer emulation), a rack with 3 effects (a graphic eq, a delay and a reverb), a preset manager, the function "my amp" which modifies slightly the tone in a way that is unique for every purchase (like it happens with real amps, which usually never sounds exactly the same among them) and a handful of IRs made just for this amp.
The fact that there are few IRs compared to other Audio Assault amp sims can be due to the fact the one I'm trying it's a beta version, but if it's not, I consider this a good thing anyway, since they clearly took time to pick the IRs that sounds the best with this amp rather than put hundreds of them and forcing the player to navigate for hours between them before finding one that suits the head.

Among the other features it's worth to mention the resizable UI and the double track emulation, which doubles the track you are recording in order to make it sound like there is a second layer, and it can be a good songwriting tool, to put down ideas faster and that already have that "wall of sound" feeling.

How does it sound?

This is clearly one of the best and most usable Audio Assault amps, they have improved constantly through time and in this specific one I must say (also thanks to a good choice of impulses) the low end is very present and in focus (which is something not very common in amp sims, realistic low mids that doesn't sound muddy) and the highs are very clear and defined, making this a very usable amp simulator.

This plugin is good not only for high-gain metal tones (in terms of pure metal sound Sigma is still superior), but also for clean and crunchy ones, which sound very warm and credible, and I can only suggest anyone interested to check it out.

Thumbs up!

Specs taken from the website:

- 3 channels

- 3 stompboxes: gate, booster and Tube Screamer

- Fx rack with eq, delay and reverb

- Tuner

- L, R and Stereo Routing

- 2x, 4x, 8x, 16x Oversampling options

- 2 Graphics engine

- My Amp Feature for an unique tone

- Double track emulation

- Presets system

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Track Grouping Cheatsheet

 

Hello everyone and welcome to this week's article!

Today we're going to check out a visual representation of the track grouping of an average project, I have borrowed the idea from a similar cheatsheet I've found posted from a user (thank you, Toby!) on the URM Academy Facebook page, but I have modified it a bit according to my workflow and naming conventions.
I consider this visual representation an useful tool because it tidies up the concepts we have been already analyzing in detail in our home recording main article, and in our project preparation one.

Let's start by saying that this picture is by no mean a fixed rule, you can modify it according to your workflow, remove and add all the tracks/groups you need etc, but it could be considered as a solid starting point if you're new in mixing a project with with a full rock band and don't know where to start. 

Let's begin from the left, here you have all the individual tracks: almost all of them are routed into subgroups and/or groups, this is made to make you process the tracks in groups, if possible, thus saving time and computer resources (as opposed to processing them individually), and when you have your sounds right and the relative balance within the group, you can literally just move the group faders to balance the main parts of the mix among them (eventually balancing only the drums, bass, guitars, vocals, synths and fx groups, just 6 faders, is much easier and gives us a much better perspective once the ground work is done).

Why the Sub Drops etc... track is alone and goes straight into the Stereo Out? Because we don't want it to be part of the "Fake Master", we don't want all the low end of these tracks to hijack completely the buss comp creating a horrible pump effect. Eventually some pump effect can still appear when this track will reach the limiter, and we will have to be good in finding the right volume for it to arrive to the limiter without creating problems.

I hope this was helpful!

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Review: Fender '65 Deluxe Reverb

 

Hello everyone and welcome to this week's article!

Today we're going to review one of the most historical guitar amps of all times: the Fender '65 Deluxe Reverb!

The '65 Deluxe Reverb is one of the most iconic amps ever produced by Fender, and its latest reissue is still on sale today from the producer, hand assembled in the USA with premium components.
This amp, historically, is suited for Country, Blues, Classic Rock, but in general its clean tones have been loved and used by guitarists from all genres.

The amp features 4 inputs: 2 for the normal channel and 2 for the vibrato one (the difference between the 2 inputs is that input 1 is 6db louder, so you should use it fwith lower output guitars, while input 2 should be used with high output ones).
Instead of the usual overdrive channel, channel 2 is a vibrato channel, meaning that has a vibrato effect always on, and there is also (only for this second channel) a reverb, with the knob in the amp acting as dry/wet mix control. 
Finally, to be noted that in the eq section there is no mid control, only bass and treble.

How does it sound? Let's start by saying that like all amps of this type, even if it's a small 22w combo, this amp is LOUD, and the more you crank it towards the breakup, the more the sound becomes harmonically rich (althought a bit noisy) and it can enter also into mild overdrive territories.
The basic clean channel is the definition of Fender clean: squeaky clean, with a lot of attack and top end but at the same time with deep bass, a combination that makes it sound almost like a piano, while the vibrato channel is similar to the clean one but with an adjustable vibrato effect, which can be useful in certain genres, and a very good reverb.
If you need only the reverb you can lower the vibrato settings to the minimum, until it basically disappears, and you will remain with another normal channel to which you can apply the reverb.

Regardless of the genre you play, this is an amp I suggest anyone to try, because it's the history of guitar and because it's featured in countless records, from the '60s up to now, and it lets us peek into a time in which in the amps there were different features, and maybe guitarists also had different needs.

Thumbs up!

Specs:

- 2 Channels

- Power 22 W

- Equipped with: 1x 12" Jensen C12K speaker, 8 Ohm

- Controls for: Volume, treble, bass, reverb, speed, intensity

- 4 Inputs - 2 per channel

- Speaker out

- Spring reverb

- Tube Vibrato

- 4 Preamp tubes: 12AX7

- 2 Preamp tubes: 12AT7

- 2 Power tubes: 6V6

- Dimensions (D x W x H): 24.1 x 62.2 x 44.5 cm

- Weight: 20 kg

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All the types of eq explained

Hello everyone and welcome to this week's article!

Today we're going to deep dive in a subject already touched in our main equalization article, but explaining more the differences between the various types of equalization and which one to use according to the needs.

- Shelving eq and Filter eq: those are the simplest of all, and can be either with an analog interface (with knobs) or with some graphic representation, but they all recreate the fact that you can choose an eq shape (in this case a filter or a shelf), choose the frequency range where to put it and dial in how many dB to add or subtract to that area (if you are filtering, obviously you're bringing to zero db the sound from a certain point up or down). By concept, for shelf we mean that for example from a certain area (for example 1khz) up (or down) we will start adding or removing gain, and this usually happens with a not too steep curve, to not make the change too unnatural.

- Graphic eq: this is a way to intervene in the frequency areas with more precision, and also this type of eq comes from the analog world. Initially the spectrum was split in just few areas, like 4, then they started adding more and more faders to the units up to 30 bands or more. You can intervene with the individual bands (the more the number of faders, the smaller the frequency range for each band), and the models with more bands obviously allow much more surgical correction, to the point that the most precise ones are used mostly for room correction, meaning to clean up specific resonance areas of a sound. When you have plugin recreation of those old units, usually they try also to recreate the way they used to colour the sound, for example adding some harmonic content or some saturation.

- Parametric eq: the way those equalizer works is an evolution of the shelving ones, basically they work the same way but they let you choose among more shapes (for example a bell shape), decide the Q (which is the width of the shape) and add usually many of those shapes, also making them interact-overlap among each other (this is a specific more of the digital ones, like the Fabfilter pro Q). This type of eq allows much more control and by consequence also much more possibility to make mistakes, but it's one of the few that for example allows you to make a wide boost in a frequency area and then, in that same area create several very narrow cuts to tame the resonances.
The most feature-rich digital ones allows also to make corrections only in the part of the sound panned in the middle or only to the one on the sides (MID-SIDE eq) or to affect only one of the 2 sides of the stereo field.

- Dynamic eq: this is much closer to a multiband compressor than to an equalization, meaning that it's dependent on the level of the input. With a dynamic eq you can for example choose an eq bell and decide to reduce the gain on a certain area: the gain reduction in that area will be higher when the gain is higher in that specific area and vice versa. This is used to tame certain frequencies that spike only every now and then like the sibilant consonants in a vocal track.

Note that all these types of eq mentioned are "archetypes", but in the reality of today, especially in the digital domain, you will find equalizers which offers a combination of those functions (for example there can be an eq which can also offer dynamic eq functions, or a graphic eq which offers also a frequency analyzer and mid-side features, and so on).

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Review: Blackstar ID:Core Stereo 100

 

Hello and welcome to this week's article!

Today we're going to review a very interesting digital combo, which is an evolution of the smaller ID Core range (which are up to 40w, while these 2 larger ones are 100 and 150w) and which have as main difference the fact that instead of having small full range speakers, these ones have 2x10 guitar speakers: the Blackstar ID:Core Stereo 100!

The choice of full range speakers or 10 inches speakers may set somebody off, since the standard for guitar is a 12 inches one, but this is a compromise solution to fit 2 speakers in a smaller, lightweight package (which is almost as small as the size of a regular 1x12 combo) and allowing the player to take full advantage of the "Super Wide Stereo" technology, which lets us playback an mp3 (via Aux) and practice along with it listening in stereo, while most of the practice combo amps, having just 1 speaker, lets you playback only in mono.
This speaker choice targets the amp towards practicing musicians which play more at home along with mp3s than for live musicians, since live there is usually more need of a 12 inch speaker to have all the low end thump, which is less strong in a 10 inch one, but at the same time with 100w there is enough power to stay on top of a live drumkit, also for rehearsals.

Besides the speaker, the amp offers 6 voicing controls, which range from clean to the most extreme hi-gain, a looper up to 30 seconds, 12 effects, a software suite for in-depth editing, usb recording, speaker emulated out and so on, making the ID:Core a Swiss army knife for the practicing musician, and the 100 and 150w versions have enough power to play live with no problems, all in a convenient 12.5kg package.

How does it sound? It sounds quite well! Blackstar Amps, especially for high gain, are some of the best sounding practicing amps, with musical mids and the proprietary ISF, Infinite shape feature, which is a knob that lets you pass from a US type of gain to a British one and everything in between.
The cleans are good and the distortions sounds almost "produced"; in this the amp the tone in general is less raw and "in your face" than a Boss Katana, but it sounds brighter (maybe also because of the speaker choice) and very pleasant, it just has a different grain and you'll have to compare them to find which one is more suited for you.

Thumbs up!

Specs taken from the website:

- 100W (SUPER WIDE STEREO 2x50W)

- 6 Blackstar voices

- Looper with 30 seconds record time and infinite overdubs

- Patented ISF control

- 12 vintage style effects including Octaver

- Store up to 36 sounds

- USB Audio

- FREE INSIDER software

- Stereo MP3/Line In

- Speaker Emulated Line Out

- Stereo Effects Loop

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Pre-count, Pre-roll and auto punch! What are they and why you should use them.

 

Hello everyone and welcome to this week's article!

Today we're going to talk about three functions, present in basically any DAW, which are fundamental when recording any instrument, and that sometimes gets overlooked: pre-count, pre-roll and the auto punch.

Pre-roll is something that comes from the analog world, it means that the track would start playing, making you hear what you have already recorded, and then the engineer would, by pressing a button, start recording you on the fly.
Today we can automate this function, and order to enable it in Studio One you need to press the O hotkey, or click on the pre-roll icon (the first arrow in the pic).
Then, by clicking on the wench icon (the second arrow in the pic), which brings us to the metronome setup menu, we can simply tick on or off the pre-roll and/enter the number of bars to play before starting recording. 
This way, once you will hit record, first it will make you listen to the previous 2 bars, so you can start already playing along with them, and then it will start recording automatically at the right point.

From that same menu you can activate also the Pre-count, which means that before playing or recording it will first play a set amount of metronome bars, to be ready with the right tempo before starting recording.
This function is used if you're starting your track from zero, so you don't have any part to play as pre-roll.

How does the computer know exactly, especially if it starts from 2 bars before, where to start recording and where to stop?
You need to select the area with the mouse, the recording will start and stop exactly within the selected part of the track.

Auto punch instead is another function that puts you in and out of record, as the track is playing back, meaning that you need to have the record button assigned to a hotkey (in Studio One is by default the NumPad * button), and as you play a track you can press the * key in real time to start recording, and press it again to stop recording, all without stopping the track from playback, which can be useful especially when tracking vocals, if you want to re-do just a word here and there without interrupting too much the flow.

I hope this was helpful!

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Review: Rivera Clubster 25 112

 

Hello everyone and welcome to this week's article!
Today we're going to talk about a legendary combo produced by Rivera amps in 2009, of which the latest version is still today on sale: the Rivera Clubster 25 112!

Rivera Amplifiers comes from the mind of Paul Rivera, a technician which worked also for Fender before opening his own brand, and today Rivera amps offers a full lineup of tube amplifiers which are known for their uncompromised sound and build quality.

The Clubster serie has several versions, 25w, 40w, with a 10 inches speaker and a 12 one, and the one I'm reviewing today is the 25w, 12" one.
The Rivera Clubster is a 2 channels tube amp, with several push-pull knobs in the eq to shape it furthermore and a spring reverb, but the most surprising thing is the volume and headroom: thanks to its oversized transformer, even the 25w version is more than capable of staying well above a rock drumkit, both in rehearsals room and live, all in a compact combo format.
The amp features 2 preamp tubes and 2 6V6 power amp ones, and immediately after switching it on you can feel the tube tone, which is loud, mid rangey, in your face and full of harmonics. 
The clean channel is really clean and chimey, and it has quite a lot of headroom considering it's 25w (obviously if you need even more headroom you must switch to the 40w version), and the high gain one has a very credible tone: balanced, pleasant and that it sounds almost "mix ready".
The overdrive channel can easily deliver heavier tones for genres such as hard rock without the need of a booster; a booster is necessary if you need the type of tightness used in death or thrash metal, though.
This said, I have seen a band using it also to play Dream Theater style (without a booster), with a 7 strings, and the tone was very similar to the Mesa one used by Petrucci: mid rangey, creamy and with a very loud power-amp roar.

The version currently on sale is the Clubster Royale Recording, which features also a loadbox integrated and speaker emulated output, so that you can record also bypassing the speaker, among the other new features, but the original Clubster soul is still there.

Still today a great, uncompromise sounding amp! Thumbs up!

Specs taken from the website:

- Channel 1 controls: Volume, bass (pull switch for boost), middle, treble, master

- Channel 2 controls: Volume (pull switch for channel selection), treble (pull switch for bright), bass

- Common front-panel inputs and controls: 1/4″ High Gain input, 1/4″ footswitch input, reverb control, presence control, power on/off switch

- Rear-panel inputs and outputs: 1/4″ effects loop send, 1/4″ effects loop return, 1/4″ line output, 1/4″ 

- external speaker output

- Preamp tubes: Two 12AX7A

- Output tubes: Two 6V6GT

- Output power: 25 watts RMS into 8 ohms

- Speaker: 12” Celestion 70/80

- Footswitch functions: Channel switching, reverb on/off (footswitch is included)

- Reverb: Accutronics® 3-spring reverb

- Height: 19.25″ (w/feet)

- Width: 18.25″

- Depth: 12″

- Weight: 38 lbs

- Cabinet material: 5/8″-thick plywood

- Construction: Dadoed joints

- Covering: Black Tolex

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Pre fader listen vs post fader listen

 

Hello and welcome to this week's article!

Today we're going to check out an important concept which is part of a proper gain staging setup, a simple check that will save us from having unwanted distortion into our tracks, the pre fader listen!.

By default, all the DAWS are set on post fader metering, meaning that if we are moving the volume fader in one channel, the meter of that channel (and the stereo buss) will rise or reduce accordingly.

The problem is that if the volume fader has as consequence the rise of the volume until a track clips, we don't know whether that channel, that recorded track with all the plugins processing it, is really already in distortion (or if it's too low in volume) BEFORE moving the volume fader (in that case we will have to check out whether there is some plugin that is clipping the signal or if the actual track is clipping).

To solve this problem we can choose in the DAW options (basically in every DAW) pre fader listen (something that is present also in many physical mixers), which swaps the position of the volume fader and the meter in single tracks or in the whole mixer.

What does this mean practically? It means that the volume fader still does its job, raising and lowering the volume, but the meter will keep showing us, regardless of the volume we decided, how hot is the audio track before entering in the volume fader, and whether the signal is already in clip or not (sometimes for example it's already in clip because the track is passing through some plugin, like some Compressor which maybe has a little too much make up gain).

By verifying the input and output level of all tracks (and therefore of all plugins on each track) we can make sure no track is clipping, and this is essential to perform a proper gain staging, thus having a perfeclty clear song.

One final note: press only one PFL button at a time, otherwise the meter will show the combination of all the active channels and plugins (in the stereo buss) and you won't know which one is too high or too low.

I hope this was helpful!

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Review: Fender Mustang GTX50

 

Hello and welcome to this week's article!

Today we're going to review a combo amp which came out in 2020, but that actually is (along with the LT50) the latest iteration of Fender's take on digital combos, with presets, amp simulators, effects etc all in the box, all at an affordable price: the Mustang GTX50.

The Mustang serie is the low price digital serie of amps produced by Fender and aimed to the beginners, and even if I don't remember the year the first amp of this serie was produced, I think it was 2010 max, I know several updated versions have been put in the market, and this GTX keeps on building on top of the original one.

In 2020 Fender proposed 2 versions of the Mustang: the LT ones, (25 and 50w), which cost around 100$ less than the GTX and offer a stripped down software with less editing possibilities and it's aimed to the beginner guitarists and the GTX one, which offers a full range of in-depth parameter editing, and many more presets and tools to carve your sound.

I had the chance of trying the GTX (unfortunately without using the app Fender TONE, which is new for this version and allows the player to edit the presets and download new ones from the cloud), and I must say I have been impressed by the amount of bells and whistles this small amp features: bluetooth, fx loop, lcd color screen, 200 presets, usb to turn the amp into an audio interface, aux, headphone out, wi-fi, Celestion speaker and so on, plus the design is quite cute.

How does it sound? Well, it depends on the genre you need: if you like the typical Fender cleans, which are ringing, chimey, with emphasis on the high frequencies, this amp is quite good in recreating them. Moving towards more crunch territories though, the amp starts showing its flaws, which makes it (and its previous iterations) more similar to the Line6 Spider: the distortions sound quite digital, meaning compressed, scooped, scratchy, and the higher the gain, the more the problem is noticeable.

The conclusion is that this is a good bang for the buck if you like clean tones or slightly overdriven, but if you're looking for high gain simulations, there are much better candidates around, for example the Boss Katana or the Blackstar ones, unless you are in for a long and deep tone editing session.

Thumbs down!

Specs taken from the website:

- New models include: Fender classics like the Blues Jr and Vibro King, as well as other amplifers including JC Clean and Silver Jubilee

- New effects include: Models of classic Overdrive, Fuzz, Delay and Pitch Shift effects

- Newly-designed stage-ready cabinet and cosmetics

- 12-inch Celestion® guitar speaker

- 200 onboard user presets that can be modified for any style of music

- All-new Fender TONE 3.0 for iOS and Android for deep editing, preset browsing from the Fender Tone community, preset back-up and restore, and more

- Upgraded seven-button footswitch (optional) features individual bank up and down functions and easy effects selection, as well as a tap tempo and 60-second looper

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Sidechain EQ: what it is and when to use it

 

Hello everyone and welcome to this week's article!

This time we're going to use sidechain in a different way than the most common one, the sidechain compression that we have already covered in this article: we're going to see what happens when applying the sidechain to an equalization.

With the term "sidechain" we define an interdependence between a trigger event (for example a kick hit) and an effect (usually a compressor that lowers the volume of another track for few instants), for instance in a song in which there is a synth pad, every time a kick hits, a sidechain compressor could lower the pad volume creating an artistic "vacuum" effect on the synth, like in the min. 1.43 of the song "Follow me" by Muse.

The "ducking" effect (that's how it's called), it's a creative choice that's functional to the arrangement of the song, but there are other instances in which a mix engineer would simply need to carve a little bit of room and avoid frequency masking in a very dense mix (for example one that is very fast or with many layers) without producing an effect that would take the attention of the listener away from the arrangement.

In this case, more than a compressor that would affect the whole tone, it's better to use an equalizer or a multiband compressor, because in this case one could clear a little bit of space just in the exact area in which we want for example our kick drum to cut the mix more clearly, without touching anything else.

How do we do it? In the Studio One interface (but surely it's very similar also in all the other DAWs) we need to load the eq or the multiband compressor in the insert of the track we want to affect (usually synths or bass guitar, but it could be really anything) and click on the sidechain button on top (as in the picture in this article), then in the track that should trigger the effect (for example the kick track) we click on the "+" button next to "sends" and there we'll see a menu with all the effects with the sidechain function active.
From there is sufficient to choose the eq or multiband compressor we've loaded on the other track and every time there will be a sound in this "trigger track", the eq or multiband compressor will activate simultaneously in the other, and in this case it will lower the eq in the other track, in the area of our choice, which should be the same that we want to emphasize in our "trigger" track;
for example if the kick track we are using is covered by the bass in the low end, for example in the 100hz area, we can sidechain an eq that every time the kick hits, it will lower 2 or 3 db exactly in that area of the bass track.

This way the kick will be more prominent, but without us boosting it and without ruining the balance of the song.

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The bass dual track mixing technique

 

Hello everyone and welcome to this week's article!

Today we're going to check our more in depth one of the techniques already explained in our more general "how to mix a good rock/metal bass" article, the most popular one: the dual track mixing technique.

This technique consists basically in having 2 tracks of the same bass take, they can be either the recording of a bass d.i. duplicated in 2 independent tracks, or 2 microphoned tracks, pointing to different parts of the cabinet (one needs to capture the high end and one the low end of the bass tone).

On the individual tracks, narrow down the eq of the low-end track by using a lo-pass filter from 500hz down (you can use also a hi-pass from 40hz up if you feel like you need to clean also some low end rumble), and on the high-end track use a low pass filter from 7khz down and a high pass from 500hz up.

Now that we have our 2 tracks nice and separated in terms of eq we can use any type of distortion we want in the high end track, for example an amp simulator (some also like to use very nasty metal amp sims for guitar to add grit to this track), and in the low-end track we should add a nice broadband compression to start making the low end stable, by reducing 2/3db of gain.

Once the 2 tracks are properly processed it's time to balance them in volume (it's likely that we will have to put the high end track quite lower than the low end one, because of its capacity of be more ear-piercing.

Now that the 2 tracks are well blended together is time to route them into a group track and load in this track an eq to sculpt the sound, if we need it (for example some like to cut a bit around the 300hz area to remove a bit of low-mids mud and to boost a bit around 920hz to add some nasal tone), then we can load a multiband compressor, with 2 bands that should be matching the frequency areas of the 2 tracks, so that we can shave off some other unwanted fluctuation in dynamics without changing the general tone, because if we would use the same compression settings for both tracks there's a chance that what sound good for one of the 2 tracks would have unwanted results for the other.
Bear in mind that the low end track already has a bit of compression, so we should keep it in mind when applying this second compression to the low end track (so let's be less aggressive with the gain reduction than we would if there was just one compressor), and that in the distorted track the gain as well acts a bit as compressor.

Finally we need to make the bass tone rock solid, stable on its railway in which it needs to stay in terms of volume and dynamics, and in order to do so we nee to load a limiter (or a maximizer) and push it until we reach 3db of gain reduction (click here for an article on how to keep the Bass stable with a Limiter).
Once you find the right setting to keep the bass track tamed and stable all the time, you can basically use the output ceiling of your limiter as volume knob, and in order to dial the right level it's important to connect it with the kick drum, as explained in this article.

I hope this was helpful!

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Using a VU meter for better low end balance

 

Hello and welcome to this week's article!

Today we are going to see a trick used to have a visual reference to improve how the low end in our mix will translate from our DAW to any other source.

Let's start by saying that we're going to need a VU meter, a metering plugin that recreates the one of an analog mixer (or a real analog one), and by specifying that 0db in the VU meter equals to -18db in the fader of our DAW.

Now that we have clear this concept, let's get to business: 
The idea is to get right the balance between the kick and the bass, and then to balance all the rest of the mix around these 2 elements.

Let's start by loading the VU meter in the Stereo Buss and by setting at zero the kick and bass faders. The first thing to do is to play the kick track (or group of tracks) and raise the gain until it peaks at -3db on the VU meter.
Now it's time to bring in the Bass: let's raise the gain until the peak of Bass and Kick combined reaches 0db in the VU meter (when they play together).

Why 3db of difference? Because if you would duplicate the kick track and play the 2 kick tracks together there would be a 3db in volume increase. By making sure the bass adds 3db to the total, means that the bass is equal in volume to the kick, balanced, and that if we mix the rest of the instruments around this equilibrium there's a good chance that our mix will translate better in the real world.

Once the 2 tracks are connected, you can raise and lower them in volume together to fit them in the mix but try to keep the same proportion, so that the equilibrium remains stable.

I hope this was helpful!

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Low latency monitoring (in Studio One)

 

Hello and welcome to this week's article!

This time we're going to talk about a theme that is often overlooked, but that is of paramount importance: the low latency monitoring, and in ordering to do it, we're going to use the interface of Studio One, but the same rules applies to every DAW.

There are 2 types of low latency monitoring: hardware and software.

Hardware low latency monitoring is the ASIO protocol (also known as Hardware Direct Monitoring), a protocol created by Steinberg, which basically lets the drivers of the audio interface to dialogue with the DAW and work together to allow the lowest latency possible for the configuration.

A particular type of low latency monitoring is called Blue Z, or "low latency monitoring for instruments", which is a bit more advanced than the classic ASIO one and it's a function supported only from certain audio interfaces (such as Presonus, RME etc), which allows you to monitor with almost zero latency without stressing too much the CPU, but will not let you hear the sound processed by regular plugins, just for some which are made for that (for example in Studio One you can add a delay or a reverb to the real time signal, those plugins are called DSP plugins).

The "software" low latency mode instead (which still anyway involves hardware as well) is the one called Green Z, or "Native Low Latency Monitoring". This mode is more advanced, and it's supported only by few, more recent interfaces and computers, and allows the signal to pass through the whole chain of effects and come out as fast as with the regular hardware low latency monitoring, but it requires also a comparatively faster computer.
Studio One, unlike other DAW, allows you to choose different settings for the buffer size, which is the protection from jitters when doing playback, and the so called "device block size", which is the buffer size for recording: the lower it is, the faster.
This way you can set "dropout protection" medium or high, to make sure there are no jitters, and then just set the device block size as low as possible to minimize the latency, and finally enable the "green z" in the mixer on the tracks you are recording in real time to use this function that minimizes the latency and also lets you hear all the plugins in real time.

The more the cpu is struggling by making many tracks passing through real time processing, the more we will need to balance, for example by increasing the buffer size (and therefore the latency), or the dropout protection (to avoid hearing weird noises during the playback, which is anyway a function tied to the buffer size), so anyway monitoring is a balance game: we need to be aware of how much we need low latency (if we are recording we do, if we're mixing we don't) and change the settings according to the phase we're in.

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Difference between old school and modern guitar distortion

 

Hello and welcome to this week's article!

Today we're going to clarify a bit what are the characteristics of the old school guitar distortion and the modern day hi-gain tone.

Let's start from the base: a distortion is an effect achieved by taking a signal and boosting it on purpose in order to make it become degraded, but in a controlled, euphonic way, with the aim of making it sound more aggressive and compressed, and this is exactly how it went initially; guitarists were plugging their guitars into tube amps and cranking them to the maximum (also because at the early stages there wasn't even what we might call today a p.a.), and the amps just couldn't handle it, so distortion was an unexpected (but usually welcomed) side effect.

Then Hendrix and rock n'roll in general exploded, guitarists were experimenting with their tone like crazy and pushing the boundaries with every record, and the amp producers simply adapted to the needs, creating amps (and pedals) capable of distorting on purpose without the need of putting everyting on 10.

Back in the day of the first hi-gain amps, distortion was permeating the whole frequency range, because it was generated both by the gain stages of the preamplifier and from the power amp, stressing its power tubes, and a good example of the early amps which featured a high-gain tone were the Marshall and the Orange amps of the '70s among the others.
An Orange amp is a good example of old school distortion: it's prominent in the low-mids, which makes it a bit dark sounding, and it's very distorted in that same area, making the tone not very defined and not easy to control, but very punchy for example with groovy single string riffs, like many Led Zeppelin ones, in which every note hits you like a truck.

Modern guitar distortion instead is all about control. This is obtained from amps which through the years had more and more gain stages and which relied less and less on the power amp to distort, until we arrived to amps with large tube power amps (e.g. 120w), with 6L6 power tubes which have a lot of headroom, and all this is to make the power amp sound clean and the amp to rely mostly on the preamplifier for its gain.
This different setup has 2 objectives: 1) to stress less the power amp, thus prolonging the life of the tubes 2) to have a cleaner, tighter low end.

To make the distorted sound even tighter, guitarists like to take a hi-gain head (for example a Mesa Boogie or a Peavey 5150), use the distorted channel and boost it with an overdrive, because it makes the input signal even hotter and the result is a tone with a fast attack, reduced low-mids and a tighter low-end (meaning that when playing a fast, palm muted riff the recoil of the lows is faster and less prominent, so you can hear the riff more clearly).

This solution is particularly good with low tunings (to be honest today it's very hard to even find a band that plays contemporary hard rock-heavy metal in E), because it adds clarity, so in conclusion the modern distortion is about a tight, controlled low-end (which needs to sound clean, because if it's too distorted the riff will sound like a mess, to the point that some metal amp has a separate gain knob for the lows and one for the highs) and a screaming, very defined upper mid range, with a clear attack and good string separation, if you want to play some articulate djenty riff like those of Periphery.

Clearly I have provided the example of 2 very different type of tones and there are surely a million tones in between the old school and the modern one, but I felt like explaining this difference, because it can help defining more the type of sound we have in mind.

Which one among the 2 do you prefer?

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Review: Audio Assault Klank (with video sample)

 

Hello and welcome to this week's article!

Today we're reviewing a new bass amp simulator from Audio Assault: Klank!

Klunk is the simulation of a preamp pedal that looks very simlar to some of the Darkglass ones: it features 3 levels of eq (one with knobs and two graphic ones, one of which in the rack after the preamp), 2 types of distortion, 2 shaping flavors (vintage and modern), compressor, noise gate, tuner and a dual IR loader with many bass IRs by Seacow Cabs.
One thing to notice about the ir loader is also the fact that you can move freely the virtual microphone on the speaker not only with the bundled irs but with any ir! 

This simulator is the third one for bass, after Bassgrinder and Duality Bass Studio, and it's the first one to emulate a pedal preamp, and it's the most advanced of the 3, it has also a very nice resizable UI.

How does it sound? It's sounds quite good, as you can hear from the sample, but it sounds also very versatile, thanks to all the IRs and the eq capabilities, which makes this plugin very versatile in terms of tone shaping.

What would I improve in the future? Here's my suggestions to Audio Assault: I would ship the plugin (not only this but all of them, since it's a problem that I'm encountering often) with a leaflet in pdf that explains exactly what each knob and switch do, because without it we might be lacking or misunderstanding some function; I would also add some graphic representation of the compression (be it a vu meter with the needle, or with the '80s leds, or in any other form), because it would help when doing some fine tuning and would spare us the need of an external compressor.

All in all, when looking for a very quick metal tone I still prefer Bassgrinder, for its capability of separating the distorted part and the clean part and because it features a limiter, but if you're looking for versatility and tone shaping capabilities you can't go wrong with Klank!

Thumbs up!

 

Specs taken from the website:

- 2 distortion modes

- 2 shaping flavors

- Graphic EQ

- Double IR loader with custom bass amp IRs from Seacow Cabs

- Rack with 9 bands graphic equalizer and compressor

- tuner, noise gate

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How to keep the bass stable with a limiter

 

Hello everyone and welcome to this week's article!

Today we are going a little more in depth in the topic of mixing with a limiter, and this article should be taken as an expansion of our "Should I put a limiter on each single track when I mix?" discussion.

My conclusion today of this whole discussion is that depends on the genre. 
There are genres in which the instruments with the biggest dynamic excursion (which are usually bass, vocals, maybe cymbals, clean or acoustic guitars and acoustic instruments in general), those in which the performer can really choose to play some part extremely quiet and extremely hard can use some track limiting, while in other genres like extreme metal, in which there is already a lot of distortion (which acts as natural compressor), drum samples etc, this is less useful and you can stick also just with the compression.

Let me elaborate more: 

If you have a song in which in the first half the singer sings with just a whisper and in the second half sings one octave higher, with very loud peaks, the first thing to do is to do some clip gain, then when the volumes are more or less consistent you can put some compression to make the track more coherent, and finally, since maybe in the highest and loudest parts there might remain some loud peak, instead of manually lowering every peak you can put a limiter with a threshold set only to check those, and bring them down.
In this case the limiter acts as final wall for the peaks that are so loud that still can't be properly tamed by a compressor, because if you would set the compressor hard enough to stop them, it would harm the rest of the vocal track, and the same concept applies to any other instrument.

The trick therefore, if you still hear some part in some track that is popping out too much, is first to try to see if you can solve in some other way, for example sometimes in palm mutings the low-end recoil can be solved simply lowering the bass on the amp or moving the microphone a couple of cm back from the speaker, but if you can't solve, the limiter with a high threshold, set just to stop those occasional peaks, can be a solution, just don't overdo it, like leaving it with a threshold that keeps it active all the time, or you will damage the sound of your instrument.

I hope this was helpful! 

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Review: Nembrini Audio Crunck v2 (free Vst plugin with video sample)

 

Hello everyone and welcome to this week's article!

Today we're going to talk about another interesting plugin, this time free, the almighty Nembrini Audio Crunck v2! 

Crunck v2 is the second version of Crunck, the first plugin designed by Igor Nembrini in the beginning of his career in DSP development more than ten years ago.
It's a virtual amplifier with a very clean and simple layout, which features all the classic controls of a real amp (gain, master, presence and eq section), and which has a bypassable cabinet section with a V30 speaker impulse response.

This virtual amplifier, which is one of the few ones available also for Iphone and Ipad through the App Store (free as well), is quite versatile and provides a very warm tone, with a gain knob that allows to dial a huge amount of gain, and lets you achieve quite a wide range of tones.

How does it sound? I'd say it sounds quite well, but it's more geared towards the rock and hard rock tones than the metal ones, since it has a lot of energy in the low mids that is not easy to get rid of (in the sample I have used an overdrive in front), so if you are looking for a creamy bluesy tone, or a fusion lead tone, this it's probably the plugin for you.

Anyway since it's free I suggest everyone to go check it out in the Nembrini website, a website that offers also other free plugins, along with paid ones.

Thumbs up!

Specs:

- 3 bands EQ

- bypassable cabinet simulator with built-in IR

- Gain, Master and presence knob

- available in all plugin formats and as Apple store app

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How to do drum replacement/layering in Studio One without 3rd party plugins

 Hello everyone and welcome to this week's article!

Today we're going to see how to make drum replacement in Studio One without using 3rd party plugins!

Drum replacement has always been a very sensitive topic among musicians: some love it, many hates it, but the truth that every mix engineer knows is that they are very important, almost essential to any modern rock, pop or metal production.
In order to do it there are many ways, for example by using drum replacement plugins like Slate Trigger, Aptrigga, Addictive Trigger and so on, but today I would like to focus on a technique that is available (as far as I know) only in Studio One and that makes drum replacement (or anyway turning any rhytmical source into a MIDI) very fast and without using 3rd party plugins, just follow these 5 steps:

1) Choose the track you want to replace (or layer) with samples, for example an acoustic snare or kick track, and from the top toolbar click on the Q icon, this will open a sub-menu in which you need to click on "groove" to open a field that will analyze your track.

2) Drag and drop the desired track in the groove analysis field, it will detect all the peaks in the track and mark them.

3) click on the "Audio Bend" icon on top and it will show you more tools to adjust, from there go on the "threshold" one and adjust it until only (or mostly) the kick hits (if we're replacing a kick track) are detected (because usually with the basic setting it's so sensible that it will mark also for example the snare bleed in the kick microphone).

4) When you are satisfied with the peak detection create a new Instrument Track in Studio One, and load there for example your sampler or virtual drumset, and from the groove analyzer field you can drag and drop directly a MIDI of the detected hits into your new instrument track.

5) The final step is to assign the right sound to the MIDI and clean up the track, eliminating the remaining unwated hits, adjusting the velocity and if necessary by moving the whole MIDI track few milliseconds back, if you hear some slight delay.

Voilà! 

You will have a nice MIDI replica of your acoustic track, that you can use to layer new sounds on top of it or to replace it altogether.

I hope this was helpful!

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Review: Audio Assault Hellbeast v2 (with video sample)

 

Hello and welcome to this week's article!

Today we're going to review the version 2.0 of another Audio Assault virtual amplifier: the Hellbeast v2!

The Hellbeast is probably the heaviest and most extreme virtual amp ever made by Audio Assault, and it's modeled on the Randall Satan amp, an amp now discontinued (but that due to some licensing change has later been produced by Fortin under the name Fortin Natas). 

This amp has the particularity of having a master gain knob and 2 other gain knobs which lets you dial separately the gain in the lows and the highs, which is a great thing if you want a tight low end, because you can keep the gain lower in the lows (so the lows are clean and fast) and raise the gain more in the high end to make the sound more aggressive and modern.

On the other hand, if you prefer a more vintage type of distortion you can raise the gain in the lows, and you will have a tone more similar to an Orange amp, which will be good for stoner, doom, or '70s hard rock.

As in all the v2 amps from the producer, also Hellbeast has all the extra features that were lacking in the first version: a completely redesigned, resizable UI, 3 channels, stompbox, rack and cab sections completely redesigned with all new IRs made by Seacow Cabs, a preset manager and (exclusive to this one) a double tracking simulator, which basically takes your playing in real time and make it sound like it's 2 layers of guitars, which is good to create some wall of sound especially during songwriting, if you don't want to spend time double-tracking everything.

All in all this is yet another good simulator that got further improved, I can't say it's my favourite from Audio Assault, it's a bit too extreme for my taste, but it actually gives you some extra flexibility compared to the others, allows you to obtain tones to a level of fine-tuning that many other sims cannot produce, and the new features are absolutely worth the upgrade.

Thumbs up!

Specs:



- 3 channels

- 3 stompboxes: Gate, Boost and Drive

- Rack Fx: 9 band equalizer, Delay, Reverb

- Dual cab IR loader with dozens of Seacow Cabs IRs

- Preset manager

- Double tracking simulator

- Resizable user interface

 

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How to create manually the glitch effect on guitars, vocals etc...

 

Hello and welcome to this week's article!

Today we're going to talk about a mixing topic which can be applied to any type of wave source and that, if used with parsimony, can become a powerful arrangement tool to make the track more interesting.
The "glitch" effect simulates a digital artifact of any kind (usualy the fast repetition in loop of a small soundbyte), and it's used widely in rock, pop, metal, hip hop and electronic music (for example it can be heard in the song Sex, Death and Money by Alice Cooper) as icing on the top of a cake which must already be good, to make it even better.

In order to make a glitch effect of repetition, which is the most common, you need to start with quantization, so take a small slice of the guitar track (or both if we're applying it to the rhythm ones), set the quantization to 32nds and just repeat the slice for i.e. a couple of measures (16 times, for example) and push play.

The sound will be the typical one of the reading error of a cd player, and this is just the basic version of this effect.
Now we can get creative: try to quantize slices twice as long to 16th of note (instead of 32nd) to see if they sound better, or create a rhytmical pattern, for example a 3 slice repetitions, 1 empty slice, other 3 repetitions and so on, and in this the limit is really just the imagination.

So far we have mentioned only taking one slice and repeating it, but if we have for example a riff of 4 chords, we can also simply REMOVE one or more slices, rhytmically, from the track, to create a "stutter effect", or use repetitively the same single slice per chord, or we can even change the quantization from one measure to another and move from 16ths of note to 32nds, to add even more movement.

Finally, it's time to talk about automations: with these, you can take your glitched section and automate on it one or more effects in order to give it even more character.
You can apply on it obviously any kind of effect, but our suggestion is to try one of these 2 (or both):

- a pitch shift, or tempo shift effect, that makes the pitch go down fast during for example the final part of the glitch.

- a bit crusher that goes from 0 to 100 during the glitch, to make it sound like it's deteriorating during the repetitions.

Let us know if you use other cool glitch effects!

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Review: Audio Assault Rvxx v2 (with video sample)

Hello everyone and welcome to this week's article!

Today we're going to review the version 2.0 of an amp sim we have already reviewed (here) but that after the latest update it has been changed quite substantially: the Audio Assault RVXX!

The Rvxx is the Audio Assault take on the Revv G20, the 20w amp by Revv, a small but very angry tube head, and this new version has received the same treatment received by Sigma 2.0: they have added 3 stompboxes, 3 rack effects, a dual cab ir loader with many good impulses by Seacow Cabs and a preset manager with several presets.

On top of that, the interface has been completely redesigned and now it is fully resizable, making it very versatile.

How does it sound? Better than the original, because the stompboxes now are more, so you have more versatility, and the cab section has definitely seen an improvement.
Compared to the Sigma 2.0, you can still tell that this unit has less watt, because it's a bit more upper-mid rangey and with less low end, but with the right impulse to counterbalance it, this characteristics are tamed, and you can enjoy the strong presence and bite without that thin and boxy vibe that sometimes comes with low wattage amps.

All in all this is a very usable amp simulator, oriented for metal, with a very good quality to price ratio, and I really suggest everyone to try it, it could make a good addition to your collection.

Thumbs up!

Specs:

- 2 channels

- 3 stompboxes: Gate, Boost and Drive

- Rack Fx: 9 band equalizer, Delay, Reverb

- Dual cab IR loader with dozens of Seacow Cabs IRs

- Preset manager

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