Saturday, September 19, 2020

Review: DiMarzio Blaze neck 7

Hello everyone and welcome to this week's article!
Today we're reviewing the neck version of the original 7 strings pickup: the DiMarzio Blaze!
The first mass produced 7 strings guitar was the Ibanez Universe 7, a revolutionary guitar designed by the guitar hero Steve Vai, which adapted to the electric guitar version the extended range possibilities offered from a low B string, already experimented in some avant garde classical guitar.

The first pickup that came with the Universe 7 was the Blaze 7, in bridge and neck position: the bridge version was rumored to be a modified version of the Steve Special, while the neck one is a medium output pickup which sounds like an overwould PAF, with a particular eq.

The eq curve is scooped compared to a PAF pickup, because the neck position of a 7 strings guitar can sound extremely muddy, so the designers removed some mid range leaving a strong low end and a cutting high end for clarity, this makes the pickup a very balanced choice for the neck position, and it's still today loved by Steve Vai, Herman Li of Dragonforce, James McIlroy from Cradle of Filth and it has been used for decades also by Korn.

The pickup has a distinctive tone which is very suited for metal solos, because it cuts through the mix very well and doesn't have that excessive high end typical of solos done with a bridge pickup, and the fact that still today it is considered a well established standard is a proof of its quality.

I suggest everyone to try it, also because compared to other brand pickups the price is not bad, you will not be disappointed.

Thumbs up!  

Specs taken from the website:

- Recommended For: All positions

- Quick Connect: No

- Wiring: 4 Conductor

- Magnet: Ceramic

- Resistance: 15.8 Kohm

- Year of Introduction: 1990

- Output: 280

Saturday, September 12, 2020

Ep, Demo, Single, Lp, Full Lenght, Mixtape, Split, live album, box sets, what are they?

Hello and welcome to this week's article!

Today we are going to see a small glossary with the definitions of words that often are used in the music environment but that not always are so easy to explain: here's your cheatsheet.

Demo: a self produced record to be used for promotional purposes, which can have any number of tracks, but on average it features 2 to 4 songs. 

Single: a recording of 1 to 3 tracks, with a runtime of less than 10 minutes which contains the main song and one or more B sides. It was very popular in the past, as it was used for radio airplay purposes. Today the concept of single is more tied to the release of a video in the streaming platforms.

EP: it means "extended play", and it usually features 4 to 6 tracks, with a runtime of less than 30 minutes.

LP/Full Lenght Album: "long play" is referred usually to vinyl records, "full lenght" is for the other supports, and it features any number of tracks, with a runtime longer than 30 minutes (but there can be also exceptions, for example Slayer's Reign in Blood is an LP that lasts 28 minutes).

Split: it could be, according to the duration, either an EP or a Full Lenght, but it consists into two or more bands putting their songs together in the same record in order to split the production and distribution expenses and to promote themselves to each other's audience. If there is a different artist for each song, that it's a Compilation.

Mixtape: a serie of recordings (it doesn't matter the number of tracks) usually released for free for promotional purposes, which does not necessarily contain only original material but also covers, remixes, B-sides and so on (for some reason mixtapes are used mainly in rap and hip-hop though).

Live album: the recording of a live performance, usually of songs previously released in studio version.

Greatest Hits/Box sets: collections of previously released songs, usually united with some rare or unreleased bonus material, such as medleys, a cappella versions, alternate lyric versions, live, covers, acoustic version, remixes and so on.

I hope this was helpful!

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Saturday, September 5, 2020

Review: Zoom MRT-3 Rhythmtrak

Hello and welcome to this week's article!

Today we're going to review another legacy product, which came out the same time as the Zoom MRS-4 (in 2003 click here to read the review): The Zoom MRT-3 Rhythmtrak.

This is a very portable MIDI drum sampler which can be powered either by batteries or by DC plug, and it offers 7 pressure-sensitive pads, 199 sounds and 396 patterns divided in many genres, from jazz to reggae, from hip hop to heavy metal.

Today all these drum modules/samplers are more or less computer based, meaning that they somehow offer a way to plug them to the pc to use also the computer samples, and they are used more as an interface to play in real time, since it is basically an instrument, but back then, when still not every studio had a PC, these sequencers were very popular because, as the name says, they would let you create loops with the integrated sounds (either by writing them or by playing them real time) and put them in sequence in order to create the song you needed, then they were synced via MIDI to the tempo of the song by connecting them to the audio workstation (or to a digital master clock, which was an external rack unit that would set the tempo for all the other devices connected).

Once all the sounds are chosen (either pre-made kits or custom ones, to which also the velocity can be adjusted), all the loops are selected and put in sequence, not only this device lets us play the song, but it let us also improvise in real time by playing with the pads during the playback, and our performance can be either recorded or not. 

What to say about this device? I have never seen so much functions packed in such a small device, and back then it was quite a breakthrough (plus the samples were decent sounding and fit for every genre), but today it's quite useless due to the lack of connections with a pc (except the MIDI one), and because with DAWs these tools are used only to play with the samples in real time, all the other functions have become rather obsolete and clunky.

I wouldn't suggest to buy it today unless you're vintage lovers, but nevertheless thumbs up for the technological content back then, it was quite impressive.

Specs taken from the manual:

● 199 16 bit-48khz drum and percussion sounds, 396 preset patterns contain a wide variety of preprogrammed rhythms. 99 additional patterns can be programmed and stored by the user.

● Create a backing sequence (song) with up to 99 patterns. As many as 99 such songs can be stored for immediate use at any time.

● Internally lit pads let you follow the rhythm pattern visually during song playback or when using a pattern. 

● Choose up to 14 sounds from the built-in drum and percussion sources, and then adjust level, tuning, and panning individually to create your very own drum kit. 

● Optional foot switch FS01 allows pattern start/stop control or tempo switching. You can also operate an assigned sound such as bass drum or open/closed hihat.

MIDI IN connector allows use with an external MIDI sequencer or other device. The Multitrak Recording Studio ZOOM MRS-4 is an ideal match, letting you synchronize the audio tracks from the recorder with the rhythm track from the MRT-3. Playing the sounds of the MRT-3 with an external MIDI component is also possible.

Saturday, August 29, 2020

A-side and B-side in albums and singles

Hello and welcome to this week's article!

Today we're going to elaborate more on our songwriting articles specifying something about the perfect composition of a tracklist (click here for the original article), as my friend Zoltan suggested.

A-sides and B-sides are a thing of the past: they come from a time in which the physical support for music came with 2 sides (vinyls and cassettes) because there wasn't enough room to store one whole album all in one side.

Why is this relevant still today? Because, historically, quoting Wikipedia: "The A-side usually featured the recording that the artist, producer, or record company intended to receive the initial promotional effort and radio airplay, to hopefully become a hit record". Basically the producers wanted the radio hosts to listen to the best songs first, and then, only if they wanted, to deepen the knowledge of the artist by flipping the side and listening to the other material.

This rule applied mostly when talking to singles: the artist would relase a single, which is a record with the main song to be promoted for radio and tv airplay, and on the flipside there would be another song or two considered by the label as "less catchy" but usually part of the same recording session.

When talking about full lenght records instead the situation is slightly different: it's true that usually the album presents its best shots in the first side, but the second side, which usually contains the same amount of songs, needs to be perceived with similar dynamics: it wouldn't have sense to put 5 fast songs in the a-side and 5 slow ballads in the b-side, the b-side needs to have some good song too, and a good mix in dynamics to keep the listener engaged until the end.

One final mention goes to the rare "double A side", a single in which both sides are considered to be the song to be promoted (for example in the case of Queen's Bycicle Race/Fat Bottomed Girls single).

Today, in an age in which the concept of "side" doesn't have meaning anymore, we say "B-side" to refer to something that the artist or the label consider secundary, and it's usually some bonus songs like live versions, alternate versions, demo versions and so on, which are used to enrich certain particular editions of a record.

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Saturday, August 22, 2020

Review: Zoom MRS-4

Hello and welcome to this week's article!

Today we're going to review another legacy product I have owned in the past, which has been a part of the early stages of my recording experience: the Zoom MRS-4.

MRS stood for Micro Recording Studio, and it was an extremely portable and lightweight recorder that hit the market in 2003, and that was storing the tracks in wave format into an sd card (unlike previous models which were recording on tape or buring the songs directly into a cd).

The premises (and promises) were as usual stellar: to be able to record your own songs directly in the rehearsal room, with 4 independent tracks at the time that could be bounced one to another (a practice that derives from the old tape recorders, in which you would merge two or more tracks into one in an irreversible manner), an array of digital effects and a metronome, all in a super small, battery powered package.

Unlike other more expensive all in one recorders of the time, this one did not come with a drum machine integrated (only a metronome), but you could connect to it another piece of hardware, a separate Zoom drum machine called MRT-3 Rhythmtrak, which would sync to the MRS-4 via MIDI.

How did it sould? Well, this has been a brief phase of my recording life (probably 6 months), and it has been replaced after not too long with a PC, because it had some serious limitation that today would be considered unconceivable, but that already back then were a red flag: the maximum recording quality was 32khz, somewhere between a tape and a cd, and there was no way to connect it to a pc (no USB etc), so the only way to export the files for further editing was via SD card.

Regarding the sheer sound quality, I must admit that it's also my fault, back then I wasn't able to nail a good gain staging, that's why probably in my tracks there was a lot of background noise, but more recently I have heard from someone who have used it with a bit more experience that the preamps and the recording quality were not that bad considered the price and the year it came out.

Together with the separate drum machine, this MRS-4 was the smallest way to record a small demotape or a rehearsal, even if the '90s style interface and the controls were all but intuitive and forced you to spend a lot of time in setting up everything, even if this unit, unlike a tape one, would allow you (with some struggle) to do also some editing to your tracks. 

All in all it has been an interesting phase in my journey in home recording, (actually I was already using a PC and a DAW since the year 2000, but I wanted to try also one of these portable hardware consoles everyone were still talking about back then, before realizing they were not only less flexible but also less intuitive and more time consuming than a DAW), but after few months I've sold it and never looked back.

Thumbs down!

Specs taken from the manual:

- Simultaneous 4-track playback/2-track recording 8 virtual takes per track add up to a total of 32 takes available for recording. 

- Flexible track parameter settings Hi/Lo EQ, effect send, and other parameters can be set individually for each track. 

- Bounce feature supports recording from 4 tracks of simultaneous playback Even when there are no empty tracks, the MRS-4 allows you to bounce existing material onto 2 tracks, while performing simultaneous playback of 4 tracks. 

- Versatile effects The MRS-4 incorporates an insert effect for processing the input signal, a send/return effect for use in a mixer loop, and a mixdown effect for use on the master bus. 

- Other sophisticated features Metronome, MIDI output, AUX input, and long-stroke faders

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Saturday, August 15, 2020

How to remove breath from a vocal track (and what is a Debreath plugin)

Hello and welcome to this week's article!
Today we are talking about a type of plugin which is very useful when mixing vocals (click here for a dedicated article), and which is closely related to a Deesser: the Debreath.

A Debreath plugin is a plugin that, as the name suggests, analyzes a vocal track, identifies the parts in which the singer is breathing in the air, isolates them and eliminates them through a compressor/gate, similarly to what a Deesser does.

This leads us to a background choice: is it really necessary to use it?
It depends on the singer or on the style. Personally, for certain heavy genres I tend not to eliminate too much the air inspiration, because it can give a sense or realism and preparation to a scream, but for less sonically busy genres this can become bothering, especially if the vocal part is the center of the mix and the breath is very loud.

For these specific cases a debreath is very useful, and the one in the pic, the Waves Debreath, is probably the best plugin for the task, since it finds and isolates the breathing parts, and lets you even hear only those parts, to tweak the treshold to perfection.

Even if this plugin is great, though, it's not the only way to eliminate breath, since you can (by putting a little more work into it) use either a gate or a multiband compressor.
The gate is good when the singer is singing quite loud, because the breathing part will be obviously much lower in volume, so by putting a gate exactly to the breathing level will eliminate only that, but if the singer is not singing loud or the breathing parts are as loud as the singing ones this solution won't cut it.

In this case, when in terms of volume the breath is at the same level of the singing, we cannot operate with a gate so we need to be more surgical.
We can move 2 ways:

- By editing the track, literally cutting away all the parts in which the singer is breathing in.

- By using a Multiband Compressor, trying to isolate as much as we can only the narrow frequency area in which the breathing happens, and by applying on it a healthy amount of gain reduction.
In this case we're not aiming to kill the frequencies but just to lower them a little bit, so they are less noticeable.

And you? Do you remove breathing from your vocal mixes?

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Saturday, August 8, 2020

Review: Seymour Duncan Distortion (with video sample)

Hello and welcome to this week's article!
Today we're going to review one of the most iconic passive humbucking pickups (click here for a dedicated article) from Seymour Duncan: the SH-6 Distortion.

The Duncan Distortion (that you can hear on a Stratocaster in the song linked above) is a High Output humbucker with a large Ceramic magnet, which grants the guitar a high gain tone but with tight and controlled low end (something harder to achieve with an Alnico one, in which the low frequencies are usually less focused with high gain).

This pickup is used/has been used by several famous guitarists, such as Max Cavalera from Sepultura and Soulfly, Wayne Static from Static X, Karl Sanders from Nile, Ola Englund, Phil X from Bon Jovi, Adam Jones from Tool, many other bands as Papa Roach, Limp Bizkit, Dokken and countless other musicians, and it's considered from many a standard for rock and metal, especially for a certain type of '90s distortion.

The pickup is used mostly in the bridge position, but someone uses it also in the neck position to give clarity to the solos, and there is also some guitar that comes straight from the factory with a dual Distortion setup.

My take on this pickup is this: the pickup sounds quite bright, it doesn't have a huge amount of low end and it's quite high-mid focused, which is good, but it's one of those types of pickups that needs to be balanced with a dark sounding guitar, like a Les Paul or some other with a thick mahogany body, because on a light guitar (such as a Stratocaster or a Randy Rhoads) the highs can become "ice picky", meaning that the attack of the distorted tone when using a palm muting sounds very prominent, like an ice-pick.
To solve this is necessary to intervene with the eq, introducing some of low pass filter or using the treble control, but in general it's a positive thing to have a fast, chugging attack, because on the pickups that doesn't have enough of it, it's impossible to introduce it later with the eq.

The conclusion is that this pickup is quite situational: if you're looking for a nice (slightly scooped) mid range, a good (but not super high) output, a raspy attack and a controlled low end to play hard rock, grunge, punk and a lot of '90s metal, this pickup is for you.
If you are instead looking for something beefier, more modern and with more body (or more output), there are also many other pickups to try, even remaining in the Seymour Duncan lineup.

Thumbs 45° up!

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Saturday, August 1, 2020

What specs to check out before buying an audio interface


Hello and welcome to this week's article!

This article is to be considered as an expansion of our article about audio interfaces (click here to check it out), and this time we're going to check out a bit more in depth the tech specs.

As per every other tech product, also audio interfaces are placed in the market in various price ranges, which can go from 60 bucks to well over one thousand (or more), and yet they all promise the same thing: to record an input signal, convert it from analog to digital and send it to your computer, usually letting you also plug an output device such as monitors or headphones.

Why the price difference then?
Because of two things: the specs and the support.
Let's start from the support: when choosing an audio interface support gets often overlooked, but it actually is even more important than the specs themselves.
You should thing about keeping the interface for several years, and if the support is not good, if there are bugs they will not get solved (and often in the first release of the drivers there is something to be fixed), or if you change your os (for example to a newer version of windows) and the support is not good, it's very likey that they will not relese the drivers for the new version, making the interface unusable (and this happens more often than you think with cheaper brands, which is infuriating).

Moving to the specs, here are some to check out when choosing an interface (obviously there are many more than these, but these are the most important ones):

Sample rate: this number tells you how often the device checks out the audio signal and records its amplitude, so it's basically the speed at which the analog sound is rendered into a digital signal.
The most common sample rates are 44.1 khz, 48 khz, 96 khz, 192 khz.
In case of 96khz for example, the interface "photographs" the incoming sound 96'000 times per second. 
These sample rates also tell us the maximum frequency the interface can record, which is the half of the sample rate number (for example in the case of 44.1 khz is 22.05 khz, which is still more than the usual human hearing, which ranges from 20 hz to 20 khz).
When recording music, it's suggestable to have an interface capable of recording at least at 48 khz.

Frequency response: this shows you the sensitivity of the device, and tells you the range on which it operates, for example 20 hz to 20 khz. This number is also modified from other factors, for example microphones, which can create a funnel effect.

Bit depth: it tells you how many bits are used for each sample recorded, and this influences the dynamic range of the interface. Having more bits means being able to record with higher dynamic ranges (therefore with a lower noise floor). 
16 bit: it's possible to achieve a dynamic range (theoretical) of 96 db
24 bit: it's possible to achieve a dynamic range (theoretical) of 144 db

Dynamic range: it's the ratio between the loudest signal that can be recorded by the interface and the noise floor, and it's measured in dbA (decibel A-weighted). 
The more dynamic range there is, the lesser the risk (when for example compressing the signal) to raise the noise floor to a point that will ruin our sound. 
In order to avoid these problems, it's suggested to use an interface with a dynamic range of 100 or more dbA.

Gain: often in the interface specs there is written something like "Gain 0-60db", which means that the gain knob can amplify the incoming audio signal up to 60db. If the preamp applies a minimum of +10db to a maximum of +60db, the Gain Range (the distance between the minimum and the maximum) is 50db.

EIN: it's the equivalent input noise, a way of stating the preamplifier noise of a recording device. When recording from a source such a microphone, the signal needs to pass through a preamp in order to raise to the desired gain level, and each preamp has a certain amount of intrinsic noise, which should be as low as possible.
Ein is measured in dbU (which are always negative numbers), and the lower the number, the lower the noise.
EIN is a good system to compare the noise level of two different interfaces, to see which one is better (for example -130 dbu is considered a very low noise preamp, while from -120 dbu up the noise starts becoming noticeable).

I hope this was helpful!

Saturday, July 25, 2020

Review: Audio Assault aIR Impulse Rack (with video sample)

Hello everyone and welcome to this week's article!
Today we're going to check out a new cabinet simulator (click here for a dedicated article about them) that loads impulse responses: Audio Assault aIR Impulse Rack!

aIr Impulse Rack is a nice little stereo plugin that acts as a Swiss army knife for impulse responses: it lets you load as many IRs as you want and mix them through a mix knob, then modify them through a serie of rack modules (an Equalizer, a High Shelf and a Low Shelf module, a Cab focus unit, a Phase/delay, a Volume/Pan unit, a Frequency analyzer, a Hi/Lo pass filter and a Normalizer).

The amazing thing of this plugin, that sets it aside from all the other similar products, it's the fact that you can create your stereo chain freely, save it in the internal preset manager, and even export your own custom ir with all the processing applied.
This is a feature that almost no other cabinet simulator has and that actually opens to an infinite landscape of possibilities: instead of having to load your favourite IRs and having to mix them, eq  them etc, you can export all the chain already processed into a single impulse and save your cpu and time for the following sessions.

I had a lot of fun experimenting with this plugin, and considering also that it comes already with a set of good impulse responses (those present also in other Audio Assault amp simulators), in few minutes I've been able to come up with the tone in the sample you can hear (only using free plugins: NaLex Gerbert and Ignite TSB-1).
Considering the features and the price, which is very good, I can only suggest you to try it out and have some fun.

Thumbs up!

- Unlimited IR Slots

- Fully Parametric EQ

- Serial/Parallel IR Processing

- 10 IR Processing Modules

- IR Export Function

- IR Folder Shortcuts

- Presets store any IRs used

Saturday, July 18, 2020

Free VST guitar amp sims article updated with 23 new plugins!

Hello and welcome to this week's article!

Today I have reviewed and updated one of our most popular articles: the one about free amp simulators, adding 23 new free Vst amps (all of which sound very good).

You can check out the updated article here!

Let us know what you think about it and whether are there more good free Guitar Amp VSTs we haven't mentioned yet!

Saturday, July 11, 2020

The history of Korg Pandora

Hello and welcome to this week's article!
Today we're going to make another review of a legacy product that is on its way of blowing 25 candles and that yet is still on the market in its latest version: the legendary Korg Pandora!

During the '90s digital effects for guitar were seen like something almost exoteric, something that belonged to the complicated racks of the guitar heroes and that costed a fortune, but then Zoom came and started offering digital multi effects for dirt cheap (and very low sound quality), and basically revolutionized the market letting anyone have all the most common tones and effects in the world.

Korg is a Japanese brand famous for making music instruments since the '60s, but it's today most known for being one of the main producers of synths and samplers (they even invented the synth+sampler hybrids in 1988), and in 1996 they decided to propose a small tool that in some way revolutionized the concept of playing guitar: the Pandora.

The Pandora is a small device, more or less of the size of a smartphone, battery powered, that offered 60 effects (up to 4 at the same time), various amp tones (from clean to extremely high gain), cab simulator, a guitar input, a volume knob, some control and a headphone out.
This little thingy was so small, light and revolutionary that became hugely popular, every guitarist imagined himself rehearsing on the headphones in some public transport, or the night in the bedroom, and the price was not particularly high.

Sure, the sounds were bad, even worse than the ones in the cheaper Zoom 505, but the size and weight were surpassing any logical reasoning.
The following year a second version, the Pandora PX-2 came out, featuring 32 drum patters with adjustable tempo, and this one was the real killer application, the ultimate guitar exercising machine.

Three years later, in the year 2000, Korg expanded its amp simulator array launching a serie called Toneworks (click here for the review of the Korg Ampworks), which was intended to be a large family of products for guitarists and bassists, and that featured the Pandora PX-3, a version for guitar and one for bass.
The only problem of this serie was the timing: the Toneworks serie was still sounding like in the '90s, when in 1998 the Line6 POD came out, changing the game forever.

In the following years there have been even more iterations of the Pandora, such as the PX-4 and then they started incorporating the Pandora tones and effects in their portable recording stations.

The current version, which is still on sale on their website, is the 2011 iteration, the Pandora Mini, which is even smaller, lighter and features 200 presets (some also for bass), 158 effects and tones (7 of which usable at the same time), 100 drum patterns, tuner and so on, and it's still today a great idea if you want to have an awesome little practice gadget always in the pocket of your guitar!!

Thumbs up (if you don't care too much about the sound quality)!

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Saturday, July 4, 2020

Everything you need to know about effects part 4/4: Distortions!

CLICK HERE FOR PART 1/4: Fx routing, in studio and live!

CLICK HERE FOR PART 2/4: Reverb and Modulations 1

CLICK HERE FOR PART 3/4: Reverb and Modulations 2

Once we have gone through all the most common application of modulation effects, it's time to pass to what is considerable the most important type of effect that we can apply to a tone, an effect which has taken an instrument, the electric guitar, and made it the most important, influential and beloved one of the 20th century: the distortion, in all of its forms.

The overdriven guitar tone is born by mistake: bands were using tube amps in the '50s and '60s, and back then there wasn't even a real PA to amplify them, so guitarists, especially those that were playing in big arenas, had to crank them at the maximum volume.
The signal passing through the tube circuits ended up saturating them, and the result was a distorted tone, with lots of additional harmonics and some compression that the guitarists found out to be exciting, to make the tone edgier and even easier to play: the tube overdrive sound was born.
The audio enhancement provided by tubes is still today often used in studios, to add a track, especially if digital, the liveliness typical of the vintage hardware.

Not too long after the discovery of the tube overdriven sound, guitarists (especially the pioneer Jimi Hendrix) started to ask themselves "where do we go from here? How can I make my tone ever more extreme?", and producers started working tirelessly to replicate the distorted sound of a tube amp full volume in a stompbox, and came up with an infinity of solutions, which can be divided in 3 macro groups: fuzz, overdrive and distortion.

The first models to hit the market (end of the '50s/early '60s) was the fuzz: it is a type of pedal that takes the original sound and distorts it, but in a very unique and specific way, by taking the sinusoidal wave and turning it into a squared one.
The result is a gritty tone with lots of sustain, impossible to obtain with a regular amp, and it was the staple of riffs that made the history of rock, such as Rolling Stones' (I can't get no) Satisfaction and most of the Jimi Hendrix and Pink Floyd solos, just to name a few.

An overdrive is a pedal that recreates (sometimes also by using a tube) the saturation of a tube amp full volume without the need of pushing it (thus without ruining the tubes): the effect is a slight distortion, more attack, less low mids, lots of extra harmonics and some extra compression; this is a great pedal to boost our amp and even out its frequencies, and it's often a solution used when recording.

Another effect that can be considered a more extreme evolution of the overdrive is the distortion.
The distortion is a pedal aimed not to boost an overdrive channel of an amplifier but to replace it, going directly into the clean one, since it's capable of producing high levels of gain, or bypassing the preamp altogether and going direcly into the power amp.
Distortions have seen their moment of maximum popularity in the '80s, with many guitar heroes (such as Steve Vai) and classic heavy metal bands (like Judas Priest) using them regularly, and still today they are very popular in certain genres such as grunge or alternative rock.

Finally, a sound can be further degraded or modified creatively in an infinite amount of other ways, and there are today pedals to do practically everything, but there are still a couple of effects that are worth to be mentioned here: the harmonic exciters, which takes certain harmonics of our tone and emphasizes them, which is a cool way for example to add brightness to a sound without messing too much with the eq (click here for an article about harmonic exciters with free VST plugins) and the bit crusher, which on the other hand takes a sound (usually vocals, or drums, or a bass) and degrades it by reducing the bit depth, creating the feeling that the sound comes through some vintage gear.

Saturday, June 27, 2020

Review: Evertune Bridge

Hello and welcome to this week's article!

The Evertune bridge is a type of bridge that requires a specific casing from the guitar, which consists in a wide hole on both sides (similar to the one for a Floyd Rose but larger) and even by removing quite a lot of wood it adds a large amount of weight to the guitar. 
Said like this it sounds quite bad, but the truth is that if you go to see the guitars that are actually used on stage or in studio from many of the most famous professional guitarists of today, odds are there is an Evertune.

Why? Because this type of bridge delivers an impossible promise: to not having to tune your guitar (almost) ever again.
This magic is achieved by a system that uses a concept similar to the one of the Floyd Rose but that reverses it: with a Floyd Rose you move the bridge to alter the pitch of the strings, and then a set of springs placed in the back of the guitar tries to put the bridge back into its original position, while with an Evertune, once you set perfectly everything, there will be a spring that will pull independently each string, forcing it to stay to its original tuning.

The result is a guitar that takes around 60/90minutes to set up perfectly (instead of the 15/30 of a normal one), but that stays in perfect tune indefinitely, and that even by changing strings, as long as the gauge doesn't change, doesn't need any additional mantainance.


What are the practical outcomes of this? Is it worthy? It is, definitely: in my case, even if my guitars are usually set up and intonated by a luthier or by myself after every string change (because I like to use different gauges every now and then), I had the feeling (and you notice it mostly when recording) that for the first time I have used a guitar without any micro-pitch alteration throughout the whole fretboard (I have it in my LTD MH-1007 ET).

How does it work? It's easier done practically than explained, and for the details I suggest you to check out the videos in the Evertune website, but the concept is this: for each string there is a saddle in the bridge, and the saddle can be in 3 positions: Zone 1 (when there is not enough tension), Zone 2 (when the string is in tune but not bendable, because every bending is compensated), and Zone 3 (when the note starts to become sharp and the string is bendable). 
After you have found the right pitch and intonation, you should turn the peg in the headstock until the string arrives to Zone 3, and then loosen it a bit until it goes back to the perfect pitch: this way you will have the ideal pitch AND the string will be bendable, it's called "the sweet spot".

You can even decide which string can be bent and which not, for example you can set it so that the lower strings are unbendable, and that are basically used only for rhythm part with perfect pitch, while the higher strings will be freely bendable (you can even adjust the bending sensitivity!), and this versatility is really stunning, so my bottom line is that this bridge is probably the first real useful guitar innovation since decades.

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Saturday, June 20, 2020

Everything you need to know about effects part 3/4: Reverb and Modulations 2

CLICK HERE FOR PART 1/4: Fx routing, in studio and live!

CLICK HERE FOR PART 2/4: Reverb and Modulations 1

CLICK HERE FOR PART 4/4: Distortions!

We have started digging into the modulation world, but the reality is that it's a rabbit hole with infinite variants, so what we're going to do today is to finish covering the modulation macro areas that are most known and used.

Let's start today with the one that can be considered probably the king of all modulations, the most used and in my opinion more useful and pleasant sounding: the delay.
In the '50s sound engineers were looking for a way to recreate the echo effect that one would obtain by yelling for example inside of a cave, with the soundwave bouncing around and coming back after a certain amount of time: the voice gets reflected and comes back to the listener summing up to the original one.

This effect is called echo, but with time the use of this echo in music became more and more creative, stemming many different types of echo, and today we refer as this type of effect as delay because we refer to the time distance from the original source and its reflection. 

One of the coolest ways to use a delay when mixing is the so-called "slapback" delay, which is a delay with just one repetition of the original sound: this type of delay is particularly cool because it lends itself to being used in creative ways with the stereo field, for example sending one repetition left and one right in 2 slightly different moments or following 2 different time signatures, and this creates a sense of depth and 3d-like sound which can make our vocal or solo tracks really stand out.

Finally, delay is one of the most beloved guitar effects because it can seamlessly give depth and atmosphere to a clean guitar part or smoothness to a hi-gain solo, and countless pedal producers have put in the market their interpretation of this effect.

Proceeding with our walk in the world of modulations, the more we dig the more we can find interesting uses, and in this the guitarists of the '60s (for example Jimi Hendrix) have been exceptional innovators, looking always for new and creative ways to play with their sound. 
A classic example of this research is the tremolo, which emulates a fast, rhytmical opening and closure of the guitar volume knob, or the vibrato, which is a fast, rhytmical alteration of the sound pitch, or the rotary effect (also known as Leslie, from the name of its inventor), which consist into passing the sound through an actual speaker which rotates inside a box.

Now it's time to move to another group of modulation effects which are a bit less common, because they are more particular sounding and are used for quite specific purposes.
Let's start from the ring modulator: this effect would deserve a lot of space because its actual uses are infinite, but let's just define it broadly, by saying that it takes two different sounds (for example a guitar and a voice) and puts them in correlation, producing a sound that is the result of the interaction between the two (for example the famous "talking guitar" of Peter Frampton).
Then we have the huge world of pitch shifters, which are the processors that take one sound and produce a copy in a different pitch; among these, the most used are the octave, which creates a copy of the original tone shifted one octave below, and the harmonizer, which creates a harmony of the original tone, and usually it features an intelligent system which respects the right interval between the notes to make it musical.

One last modulation that is worth to be mentioned is the vocoder, which is a tool that takes a sound, usually a vocal track, and "passes it through a synth": this was a very popular effect in the '70s and '80s, used for example by Earth, Wind and Fire, but also more recently by Daft Punk (e.g. in the song Harder, Better, Faster.   

Saturday, June 13, 2020

How to use the marquee tool (or smart tool) for editing and automating

Hello everyone and welcome to this week's article!

Today we are talking about an useful tool for editing and doing automations that can make our workflow faster, which in some daw is called "marquee tool", in some other "smart tool", but it's the same thing, and we're going to show it through the Presonus Studio One interface.

Let's start with the editing: usually to edit a track we select the cut tool, click with the cursor in the beginning and ending part of the section we want to cut and then we drag it around wherever we need.
With the smart tool we need to select the symbol in the red square in the image, and to keep selected the arrow tool. This way if we click in the upper half of our audio track it will become a select tool, so we can highlight a certain part of the track, then we just double click on the selection and it will automatically be cut at the beginning and the end of the selection. If you sum up the time saved using this tool when editing a whole song, it will add up to minutes, or sometimes hours.
By dragging vertically the selected part from the lower half of the audio track, we can also automatically move it to another audio track, and all these things works also with multiple selections at the same time.

Moving to the automations, usually we open up our automations panel and start drawing points wherever we need, for example to raise and lower the volume of certain parts of our track.
In order to do the automations via the smart tool first we enable the automation we need, then we select the area we want to automate (we can do it also with multiple tracks at the same time), hover with the mouse in the upper part of the track until it turns into the shape of a bracket like this l-l and then simply drag the automation up and down in the selected area, it will automatically move it without the need of drawing points.

This marquee tool is very useful and time saving, give it a try!

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Saturday, June 6, 2020

Everything you need to know about effects part 2/4: Reverb and Modulations 1

CLICK HERE FOR PART 1/4: Fx routing, in studio and live!

CLICK HERE FOR PART 3/4: Reverb and Modulations 2

CLICK HERE FOR PART 4/4: Distortions!

Now that we have seen the basic rules of how effects work is time to start talking about some of the most important of them.
Let's start talking about envelope filters: those are basically a creative form of equalization that can be controlled in real time, moving through the spectrum while the sound is playing and creating a very interesting effect which shows one of its most famous uses in the Wah, a classic guitar effect, click here for a dedicated article with free vst plugins.

Besides the eq-based effects, it's definitely worth mentioning the reverb, one of the most important effects of all: it's an effect that mimics the bouncing of a sound in various ambients, which can be from small rooms to huge caves, and it's essential to add smoothness and realism to any tone and to make it less harsh; click here for an in-depth article about reverbs.
Reverb can be used not only to recreate ambience, but also for creative purposes, such as specific effects like a sound that is incoming (the inverse reverb effect, or preverb, like in the horror movies) or even to recreate a very accurate reproduction of the interaction between a cabinet and a microphone (click here for an article about impulse responses).

Moving towards the modulation effects, let's first start with the definition: they are filters that take a given signal and create a copy, with a given delay and pitch modification, to be summed with the original one, creating a wide range of different results.
Said this way it's a very wide definition, but modulations are the core (together with the reverb) of sound effects, and they are many and capable of obtaining very different result.

Let's start with the chorus: the chorus is a type of modulation that doubles the sound creating a slighly delayed (usually around 20ms) copy whose delay will not be stable but will keep on variating, oscillating 5ms more and less, plus the copy's pitch is slighly detuned, giving the impression that the copy is (in case of a Vocal track) another person singing along with the first one: not identical, but very similiar, and this effect is used to make the original sound wider and fatter.

Moving to other types of modulation, other two types which are quite important are flanger and phaser.
These 2 effects starts from the same idea: to split the signal in 2 copies and putting one of the 2 rhytmically out of phase with the other. The fact that this phase changes all the time following a certain tempo creates a very "sci-fi" effect, and the flanger takes this effect one step further by delaying the second sound copy and moving it back and forth in time, creating an even stronger phasing effect.

Saturday, May 30, 2020

Video day 2!

Hello everyone!

Today (after years) I have finished the cleanup: I have taken all the audio samples created for old articles of this blog and turned them into Youtube videos, so now there are 2 new video samples for the following articles:

How to sound like Metallica (with free VST plugins)


How to sound like Jimi Hendrix (with free VST plugins)


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Saturday, May 23, 2020

Everything you need to know about effects part 1/4: Fx routing, in studio and live!

Hello and welcome to our new omnicomprehensive article in which we are going to gather together and put in a coeherent order all the various articles of this blog related to effects!

Let's start very generic: what do we mean in a wide sense when we refer to audio effects?
We talk about any process that artificially modifies an audio signal, which can be recorded or live.

(Small disclaimer: in this article we will talk about everything except Equalization and Compression, because they have a specific tag on their own and are explained more in detail in other articles such as our "how to mix a song" serie).

In the music world, whether we are talking of studio recordings or adding an effect to our electric guitar sound, the basic concept is that we can put any effect in any order, but the reality (and decades of perfecting the craft) tells us a set of rule quite stiff in which to optimize the effect chain order: every effect, in facts, cascades into the others and modifies them, therefore putting the same effects in a certain order can result in a pleasant enhancement, while putting the same effects in another order can just create a messy cloud of noise.

Decades ago, some audio engineer has realized for example that by putting all the effects in front of an amplifier would make the resulting tone quite bad, while putting some of them AFTER the preamp, just before the power amp, would make the sound much cleaner and more polished: that moment was the creation of the effect loop, a tool that still today is considered a staple in audio technology (click here for a dedicated article).

By putting the right effects in the right order in our guitar and bass rig, we can obtain the best tones possible, click here for the perfect guitar effect chain order (with the explaination also of why a certain effect needs to stay there and not in another position), and click here for the perfect bass effect chain order.

The same logic can also be applied in studio, and not only for guitar, but also for any sound source that goes into a mixer, that's why also mixing boards often features an effect loop.
Regardless of the loop, if we're talking about the digital world of recordings, what we need to know is that we don't need (especially if we have projects with a very large number of tracks) to use individual instances of an effect (for example the same reverb) into every track: we can create an fx track (click here for a dedicated article), and "send" this same effect to the various tracks, controlling the single amount desired for each track.

The nice thing of an fx channel track is that we are not limited to one effect at the time if we want, we can create also elaborate effect chains, for example with reverb and delay (click here for a dedicated article), and we can even make sure that only a part of our signal (for example from a certain frequency up) is affected (click here for a dedicated article about how to use the insert of an fx channel, practice also known as "to effect an effect").

Finally, it's important to say that any effect or serie of effects we are going to use in studio to process our tracks not only can be applied only to a certain part of the tone (for example a specific frequency area), but also to the whole tone in an adjustable amount, which is controlled by the "dry/wet" control: if the knob is 100% dry the whole signal will not be effected, if it's 100% wet the signal will be completely effected, and every shade in between will be a blend between effected and dry signal (consider that often a 10/15% wet signal is more than enough to give a track the enhance it needs without making it drown).
If you want to get nerdy it's also possible to be creative with the use of wet and dry, for example with the "wet/dry/wet" trick, which can be seen clicking here.

CLICK HERE FOR PART 2/4: Reverb and Modulations 1

CLICK HERE FOR PART 3/4: Reverb and Modulations 2

CLICK HERE FOR PART 4/4: Distortions!

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Saturday, May 16, 2020

is it better to use guitar plugins in mono or in stereo?

Hello and welcome to this week's article!

Today I have come to this question because I was mixing a project which was quite cpu intensive, lots of plugins involved, and on each guitar track there is a gate, a booster, an amp simulator and a cabinet simulator, then both tracks (left and right) are routed into a stereo buss with eq and compression.

I was looking at the cpu struggling and I have decided to make some trial: to see the difference in cpu load when loading 2 mono instances of a plugin in 2 tracks, or 1 stereo instance into a stereo track.
The logic would suggest that by loading the stereo plugin into the stereo track and routing on it 2 or 4 d.i. tracks there would be less cpu stress, but not always is like this!

I have tried Ignite Amps ProFET, the Tyrant Screamer and Pulse, both in mono and in stereo, and I have seen for example that they respond very well: 2 mono instances are not too heavy on the cpu, 1 stereo instance is even lighter (this means it's good code!), but with other amp simulators (one of which is one of the most praised in the forums) I have noticed a 20% cpu usage per each mono instance, which skyrocketed to a 50% for a stereo one, basically jeopardazing my project.
Needless to say, I couldn't use that plugin in my project (even if my pc is not that bad).

What is the lesson to learn from this?

That we cannot tell how much a plugin is cpu intensive until we load it, and that sometimes there is no correlation between how heavy it is in mono or in stereo.
We just need to test it ourselves.

If the plugin drains in stereo as much as the sum of the single instances or less, then it's suggestable to run it in stereo, so that we can control the various rhythm guitar tracks faster and all with one fader (if we have an impulse loader that lets us load 2 impulses and use them as dual mono we can also use different irs for the 2 sides), but if the stereo instance of the plugin drains more cpu than the sum of the individual mono ones, then let's stick to the mono ones.

And let's note that it's not a good plugin to be run in stereo.

I hope this was helpful!

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Saturday, May 9, 2020

Review: Audio Assault Signature Ir Cab Pack (with video comparison sample)

Hello and welcome to this week's article!
Today we are going to review another Audio Assault product: Signature Ir Cab Pack.

This is a very rich IR pack, the first paid one produced by SeaCow Cabswhich features more than 1500 phase coherent impulses, divided in 13 cabinets, each one with separate impulses for single microphones (the most used in the studios: Shure SM57, Royer R121, Neumann U87, Shure SM7B, Sennheiser MD421) and combined ones, both in Wave file and Line6 Helix format.

The cabinets are divided by artist, meaning that they are those used by famous rock and metal artists (in my sample I compare 5 ot the 13: those used by Kirk Hammet of Metallica, Dave Mustaine of Megadeth, Synyster Gates of Avenged Sevenfold, Misha Mansoor of Periphery and Jim Root of Slipknot), but the thing that I have appreciated the most is the fact that for once, the way these impulses were captured and named is tidy and intuitive:

each cabinet has a folder for each microphone, and within each folder there are some irs, each one corresponding to a different position in the speaker, from closest to the farthest from the dustcap.

This intuitive approach, which should be the normality but anyone that has tried different impulse packs knows that usually it's quite the opposite, led me to choose the microphones and the positionings the same way I would do in a real studio, fot example combining an SM57 at the edge of the dustcap with a Royer, or with the MD421, and the result was like I was expecting, sign that these impulses are very realistic and well captured.

What else to say? This is a very complete pack which will provide every rock/metal fan all the tools to dominate the amp simulator world for years, offering a canvas of tonal capabilities which covers all the most common studio microphones and positionings, and at its price range it's probably one of the best if not the best ir pack on the market today.

Thumbs up! 

Signal Chain: LTD 1007ET -> Ignite TSB-1 -> NaLex Uber -> Lancaster Audio Pulse 

Specs taken from the website:

- 1500+ IRs

- 13 Guitar Cabinets

- 5 Mics Per Cabinet

- 50+ Mic Mixes Per Cab

- 65 Helix Presets

- Available as WAV files & Helix Presets

- Made by Seacow Cabs

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Saturday, May 2, 2020

Songwriting tips: how to write a good song - part 3/3



Moving on with the construction of our song we have arrived to the moment of the arrangement, a word that means a universe and that we can summarize as "the dressing for our dish of pasta": we can leave the pasta completely without anything, or add just a bit of oil and salt, or we can go crazy with the most complex sauces and dressings, to the point that the pasta doesn't even have its original taste anymore.
All of these choices are completely legitimate, and the arrangement will turn out to be almost a reflection of our psychology, of what we actually want to express beyond the lyrics and beyond the pre-costituted conventions in terms of lenght, structure or chord progressions.
The musical variations that we will impress to the main melody (this is the basic definition of arrangement) will make the song minimal or baroque, will transmit to the listener feelings of happiness or sadness, thrill or relaxation.
Click here to know more about arrangements with our in-depth article.

The concept of choosing, giving an imprint, which is typical of when structuring and arranging a song, is the core value also of creating the tracklist of our record: the sequence of the songs it's a flow that must leave the listener wanting always for more, the various moods needs to be alternated and the record, if we want to make a classic that is listened from the beginning to the end without pauses, needs to follow certain proven rules in terms of distribution of the various dynamics (for example it doesn't make sense to put all the slow songs one after the other and then all the fast ones, because people would get probably bored).
Nailing the perfect tracklist is like dressing up, it's all about trying to draw the attention of the people on the parts that we know are good while "hiding" a bit the parts that we are least proud of, and this can be done also by predicting the parts in which the attention of the listener is higher and when it can be lower, or his/her ear fatiguing, assuming that he's listening to the album from the beginning to the end.
This and other informations can be found in our article about how to build the perfect tracklist for our record.

The final suggestion of this article is something that adds up to the building of the perfect tracklist for our record: building the perfect live setlist.
Once our album is ready and the band is all fired up to take it live, it's important also here to spend some minute in deciding the perfect setlist for the gig, since the live public follows dynamics that are different from someone who listens to the album from the bed, therefore we shouldn't just repeat the same tracklist of the record.
The audience live usually comes to be more entertained, so we need to keep the setlist on an average a bit more energetic, and to be able to regain the attention of the crowd if we see it's declining: Click here to read our article with 5 tips on how to compose the perfect setlist for your live gig!

I hope this was helpful!



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Saturday, April 25, 2020

Review: Audio Assault RVXX (with video sample)

Hello and welcome to this article!
Today we're going to review a new guitar amp simulator from Audio Assault: the RVXX!

The RVXX is an amp simulator based on "an extremely aggressive high gain lunchbox amp with a very big bite"; even if it's not expressely said, I imagine that we are talking about the Revv G20, a 20w all tube little amp, which has a lot of gain and it is indeed extremely aggressive.

Right after you load the plugin you will find yourself in front of the typical Audio Assault interface, which is basically the same seen in the Sigma amp sim: a nice 3d render of the head, a simple one-knob noisegate, a one-switch booster, an IR loader with 12 custom SeaCow cab impulses, and the amp controls.

This head has two channels (clean and distorted) with common eq section, presence, depth, master (which adds also saturation) and two extra switch: aggression and wide, as in the original REVV head.
The wide switch adds bottom end (and some high end in the clean channel), while the aggression one adds some gain and changes a bit the frequency response, basically creating a different sounding channel.

How does it sound? Well, it sounds very good! Compared to the Sigma you can tell this one is modeled on a lower wattage amp: it's more high-mids oriented, which makes it a perfect djent amp (that's why for the sample I have been inspired in making some djent-ish riff), and the gain and attack are really great for modern metal, I would even dare to say that this is probably the best sounding modern metal/djent amp simulator I've ever played, even if it's good also at lower gain settings, being capable of pulling out also credible warm overdriven tones.
Plus, and this is the same also for Sigma, I must say that involving SeaCow Cabs has been a good choice: the impulse choice on these 2 amp simulators is better than ever, probably better than any previous Audio Assault product so far (even if I'm still particularly in love with one of their first ones, the Bulldozer).

Thumbs up!

Specs taken from the website:

- 2 channels

- Aggression and Mid Width switch

3 Band equalizer, Presence & Depth controls

- Master (adds saturation in the power amp section)

- Gate, In & Out controls

- Stomp, Amp & Cab engagement

- 12 Wild Built-in IRs (by SeaCow Cabs)

- IR loader

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