SERIAL COMPRESSION (a guide for dummies)

Hello and welcome to this week's article!
Today we're going to talk about a common problem during mixing, specifically when mixing instruments with a very high dynamic range, like the Bass guitar and some drum parts.
The idea behind the Compression, as we have seen on the dedicated articles, is to make the sound "stable", to avoid it to move around in the mix balance: this is particularly important for the Bass, since its low-oriented sound generates a lot of energy, therefore it needs to be "tamed" with a particular strenght.
Compression, when set with a high ratio (e.g. 10:1) works almost like a Limiter: it attenuates the peaks that surpass a set threshold with a very strong reduction (in our example every db over the threshold is reduced to 0,1db).
A Strong Attenuation generates obviously a pretty drastic cut on the compressed wave, and it may damage the Transient (click here for a dedicated article) in a very bad way, for example if a Snare track is overcompressed, we will notice that the sound will lose its "snap" and will become very unpleasant.

So How do we tame an highly dynamic sound preserving the transient from being excessively cut?

We can try using a serie of two Compressors, one after the other (thus the term "Serial Compression"): the first one will just tame part of the peaks, moving the whole sound towards a narrower range, the second one will peel off the remaining peaks, and if we set both compressors for example at a ratio of 5:1 we will notice a perceivably better transient preservation rather than using a single, 10:1 ratio instance.
The result will be a Bass sound that will never move from where we set it, and this will allow us to raise its volume without worring.

About the Drumset, the whole situation is a bit trickier: every drum part needs a different compression, but what is important it's that usually all the drum tracks are routed to a drum bus, in which is often set a buss Compression (click here for a dedicated article), therefore it's important to balance the single track compressions with the buss one, in oder to not squeeze the sound too much.

Sometimes the tracks are even compressed on a global mix buss and then compressed again in the Mastering Phase, before getting Limited.
It's EXTREMELY important to have clear in mind all the stages of compression that will be applied to the single tracks, and lower the single compressor settings accordingly, or the final result will be very unpleasant.


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The difference between Vst 1, 2, 3, and between 32bit and 64bit Plugins! a guide for dummies

Hello everyone and welcome to this week's article.
Today we're going to understand the difference between 32 and 64 bit Vst plugins.
First off let's say that the older computer generation features a 32 bit processor, a 32 bit OS and a 32 bit Daw, and this is the way it has been for the last (at least) ten years, and in many studios it's still today the standard.
A 32 bit Os and Daw can only handle less than 4gb of Ram, which means that if you add more memory slots, they will not be detected from the computer.
With the increasing of the computing power needed to run the latest plugins, 3 gb of RAM weren't enough anymore to handle big projects, so lately the market has seen the blooming of 64bit cpus, often with multiple cores, that can handle much more than 4gigabytes of Ram.
With a 64bit cpu we can run a 64bit Os and a 64 Bit Daw, and this chain will let us use all the Ram we have installed, with a lot of benefits in terms of speed and stability (actually a 64bit cpu has many more upsides, but we're musicians, not computer techs so it's not our concern :D).

If we can, it's very important today to have a 64bit processor and Os, in order to be able to use all the ram we can, that is one of the most important things needed in digital music production.

What if we have a 32 bit Daw and it doesn't see our 64bit Vst plugin, or we have a 32 bit plugin that is not being loaded on our 64 bit Daw?
There is some Bridge program, like Jbridge, that helps us in this transition phase, until we will all have only 64 bit software (some Daw already features a bundled bridge program, though).

The 32 and 64bit plugins topic is also a obviously connected to the Vst standard.
Vst plugins have evolved in terms of optimization from their first standard, to the version 2, up to the latest Vst 3 version, which is specifically optimized for the 64 bit computers, and offers more routing options, along with other upgrades that makes this standard more stable and cpu-friendly.

(the video on the top of this page is a song mixed and mastered by me for Subcortical Inertia, using 64 bit Vst plugins).

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WAH WAH and the other ENVELOPE FILTERS! with Free Vst Plugins Inside!

Hello and welcome to this week's article!

Today we're going to talk about the Wah Wah, and more in general about the envelope filters!

Envelope filters are basically Equalization filters that can work both automatically, through fixed or dynamic time and amplitude parameters, or can be controlled manually, to bring in and out in real time certain frequences from a sound.

The Wah Wah, which is the most famous application of an envelope filter, is a foot-controlled filter that takes in or out certain treble frequences, while a note is sustained.

The name Wah Wah is onomatopoeic, which means that the device recreates the sound effect produced, in facts if we apply it to a trumpet or a distorted guitar, the result is similar to the modulation of vowels made by a human voice (in facts one of the most famous wah wah pedals is called "Cry Baby").

One of the musicians that contributed to make the Wah Wah pedal famous is without any doubt  Jimi Hendrix (impressive how this artist has been taken by us many times as an example for the use of effects: the truth is that after half a century, the guitar effect standards set by him has never really gotten old), with songs like "The Burning of the Midnight Lamp", or Guns'n Roses guitarist Slash, with classics like "Sweet Child o'Mine".

Another artist that has built his fame also around the sound of his wah is Kirk Hammett of Metallica.

There are many different types of Wah, that changes according to how many and which frequences are filtered, but the idea behind them is always the same.

Talking about the envelope filters in general, instead, we can see a massive use of them in todays electronic music, for example artists like Skrillex shapes their tone by filtering in and out certain frequences from their lead synths, in time with the song to emphasyze the groove.

Most of DAWs already features a bundled filter or wah effect, but if you want to try some cool vst downloadable for free, here's our suggestions:

- Samsara Cycle Audio Wahzi: an interesting autowah effect with many editable parameters.

- Mtg Wahwactor: a very simple wah emulator, Voice Controlled.

- Coyote Electronics WahGT: a wah emulator that can also be controlled with a midi pedal.

- Fretted synth WAHT: a wah designed for the guitar.

- GSI EFFECTIZER: a single vst that features all the modulation effects, included the wah.

- Admiral Quality Naive LPF: a midi controllable envelope filter.

- Topaz Dyna Filter II: an AutoWah and Dynamic Filter.

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THE PERFECT BASS EFFECT CHAIN ORDER

Hello everyone!
Today we talk about an interesting bass related topic, that we have already seen for the guitar (Click Here for the perfect guitar effect chain order!).
The rules aren't totally different, actually they are very similar, but we need to adapt them to this other instrument's characteristics.

So, as for guitars, here is our bass chain by areas: BASS -> PRE GAIN AREA -> GAIN RELATED AREA -> POST GAIN AREA -> AMPLIFIER

- PRE GAIN AREA: Starting from the instrument itself, it's a good idea to set at the beginning of the chain the WAH or the othter envelope filters (such as the Electro Harmonix Bass Balls), then we can set an Octave, and, if we need it, a Compressor.

- GAIN RELATED AREA: in this area we can use any kind of Overdrive, Fuzz or Distortion, and in this case the Post Gain effects goes after this area, or we can use the Amplifier's Preamp section as Gain Related Area, and the Post Gain effects goes into the amp's FX Loop.

- POST GAIN AREA: in this area we can set all the post-gain processors, such as modulation effects (chorus, delay..) and last the Reverb.

Here is the effect order: BASS -> WAH/FILTER -> COMPRESSOR / LIMITER ->OCTAVE / SYNTH -> OVERDRIVE / DISTORTION / FUZZ -> CHORUS / FLANGER / PHASER / TREMOLO -> DELAY / REVERB -> AMPLIFIER

As for the guitar, the EQUALIZATION can be set anywhere, from the top of the chain to right after the gain related area; the thing to keep in mind is that if we cut, we can go before the compressor, if we boost we should place the eq after the compressor, otherwise it would cut our boost.

Keep in mind that this is just one of the many chains usable, and it's made to understand the basic rules of bass effects, but we can obviously experiment modifying the effect order to see what happens, and often the results will be very cool and creative. 

Just keep an open mind and let us know if you find some interesting variation!

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REVIEW: 1973 GIBSON SG STANDARD

Hello and welcome to this week's article!
Today we're going to review an interesting guitar, owned by the father of a friend of mine.
This guitar is one of the very few Gibson that featured a Bigsby bridge, the type of bridge that allowed the player to use the whammy bar whithout having to dig a hole in the guitar body, and this feature disappeared almost completely from Gibson guitars around the half of the '70s.
The body is extremely resonant, yet not too heavy, this may be also because it has been regularly played for 40 years and always kept in dried places, allowing the wood to dry out and become resonant at its best.

The pickups were also different from today SGs: they were still humbuckers but with a lower output (some sort of "p.a.f."), still very creamy and mid-rangey, but with more clarity and less bass than the contemporary models.
These same pickups, treated with tar (the way they used to make them a few decade ago), today are equipped only in some dedicated vintage reissue.

Tech Specs:

Body: Solid mahagony
Neck: Mahagony
Fretboard: Rosewood 22 frets / block-inlays
Pickups: 2x ori. Gibson "Pat. Pending-stamped" tar humbuckers
Electronics: 2x Volume, 2x Tone, 1x 3-way toggleswitch
Pickguard: Small ´60s style 5-ply pickguard (blk/wht/blk)
Tuners: 6x Grover metal
Bridge: Bigsby


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REFERENCING and MIXING!

Hello and welcome to this week's article!
This article can be considered an extra part of our Ear training section, and explains practically how to learn something about a professional record's song and to translate it in our mix.
Let's picture this scenario: we have our mix project in our Daw, we add a new stereo track and import our favourite song, lowering its volume until it matches the one of our project.
Now we can investigate that song, asking it the typical questions that we have already seen on the second and last part of our ear training aricle:

"how much room is reserved for rhythm guitars? Where is located the snare? How deep the bass goes? how much room is taken from the vocals?" ...and so on, and we use the tools and the methods explained in those articles to find out how our favourite mix is composed.

Once we have clear in mind how the song is constructed, tonally speaking, we need to compare it with ours, keeping in mind that 60% of the tone shaping is in the recording, and that we cannot twist a sound into a completely different one without runining it: we can cut frequences doing relatively little damage to the nature of the sound, but when boosting, the more we boost, the more it will sound unnatural and digitally warped (as it actually happens).

Comparing our song with the referincing one we coud notice for example that our snare drum sound it's completely different from the imported one: that one for example can be high oriented, because the drummer had a small snare with a very high tuning, while ours is dark because the snare was huge and low tuned, so the sound will never be similar, and it will sit on a totally different place in the mix. We cannot completely transform it because it's gonna sound awful, so we got two options: replace our snare with a sample through a drum replacer, or compare our track with another one that has sounds more similar to ours. It has no sense to have for example a grunge song to mix, and compare it with a death metal one only because we like the snare: it will only confuse us and lead us to wrong choices.

What it's particularly interesting is to understand how the audio engineers have treated the 2 most problematic areas: the one in which the vocals and the snare drum lies, and the low end.
The first one it's the zone the human ear is most sensible to, so we can understand by listening to it how a mix is balanced: are the vocals at the same volume of the drums, as in the modern rock?
Are kick and snare louder than vocals (as in death metal)?
Or the opposite, as in pop music?
We can apply the same pholosophy to our mix, and it will make the difference.
This area is also interesting to study, to understand how the engineer has treated guitars: has he cut the guitar frequences here to leave room to vocals and snare, or did he let the guitar be prominent, sacrificing frequences of the other instruments on that area?
About the low end: where does the guitar stops and where does the bass pops out?
The lower the guitar tuning is, the more usually a guitar is low-frequency oriented and "eats" room to the bass.
Is the bass distorted? How much?
To the point of becoming just a complement of the guitar as it happens in many black or death metal albums?
Or at the opposite, it's just lows and it lies on a range reserved to it?
How loud the cymbals are? Usually the louder they are, the more the album acquires a "garage", realistic, alternative rock feeling, and the more it becomes "dirty".
In pop music and modern metal, cymbals are held very low in volume, while the indie bands likes them to be high and resonating, so, again, it's a style choice that we must understand and use to our advantage.

These are all lessons in sound balancing and eq choices that we can absolutely learn on our own from the masters, and apply them on our mixes, keeping in mind that in the end our mix has to sound well, to be euphonic and balanced with what we got, that is totally different from the original sounds tracked for the referencing songs.
We cannot be too stubborn in wanting a sound that we cannot obtain because the raw track is too different, and we should focus instead into making great music with what we got.
In the end, that is really all that matters.

You can learn more about the best uses of a reference track HERE.


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BIT DEPTH and SAMPLE RATE! a guide for dummies

Hello and welcome to this week's article!
Today we continue with the technical stuff made easy, and we're going to find out what are and how to use the Bit Depth and the Sample rate.

First off: why do we need them?
Because we need to choose the right setting before starting to record our tracks: the different settings will affect the dimension of each recorded track, its characteristics and the cpu usage of the project itself, so we must find the right compromise between quality and usability, especially when it comes to large projects.
The types of bit depth and sample rate availabe is variable according to the type of Audio Interface, and can reach 24bit and 192khz (as a maximum), but if an audio cd, an mp3 or a Youtube video can read only files at 16bit and 44.1khz, why do we need to record at settings so different?
The idea behind the availability of those settings is that we can record at the maximum fidelity possible, and then Dither the signal down to the standard format, obtaining a result that can be better than recording straight in the cd format.

- Bit Depth: why does people record usually al 24 bit? Because Bits are all about headroom: with 24 bits we're allowed to record hotter signals without the risk of clipping, or to record normal signals with a much quieter "noise floor".
In facts, in the analog era, it was important to record with as much signal as possible before clipping, because that way there was much signal emerging from the noise floor, which was quite high; in the digital domain, especially with 24 bit interfaces, the noise floor is so low that we can perfectly record with conservative levels without risking anything.
Sometimes the DAW offers the opportunity to record at 32 bit, but since 99% of audio interfaces in the market can record at a maximum of 24 bit, this feature should not be used.

- Sample rate: to understand this parameter, think about the frames of a movie. If a movie is 24 frames per second it's good, if it's 60fps it's much more fluid.
Sample rate does a similar thing for the audio section: it controls how many snapshots of an audio wave are taken in a second.
The reality is that 44.1khz are enough to represent well basically any musical style, so unless we have some particular need that forces us to use different settings, 44.1khz it's perfect, and at the same time it doesn't weight too much both on our Cpu and hard drive.
Obviously if you want to record your album at higher sample rates feel free to experiment, and if you want, let us know the differences you've noticed!

Our suggestion for a project is to record at 24bit and 44.1khz, that way there will be enough headroom and a great noise to signal ratio, and at the same time the project will not be too heavy on the cpu and hard drive.
Click here for a dedicated article about why recording at a too high sample rate can be even detrimental to our project.


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