Review: Seymour Duncan '59 neck

Hello and welcome to this week's article!

Today we're going to review a legendary humbucking pickup (click here for an article dedicated to humbuckers), which ended up also in our top 10 pickups list: the Seymour Duncan '59!

The '59 is a "p.a.f." pickup, which means "patent applied for", and indicates pickups that try to recreate the tone of those mounted on '50s and '60s Gibson guitars, with medium output but with a good, musical mid range, and with a present but not "scratchy" high end.
This style of pickup is still today extremely popular, and it inspired a wide production of clones from many manufacturers, to the point of almost becoming a standard.

The neck version of the '59 is probably the most popular Seymour Duncan neck pickup, and for good reasons: it's warm and full sounding thanks to its alnico 5 magnet, but it has a precise attack and the mediums are not too pronounced, resulting in a perfect balance both for clean parts (btw there is also a coil split version) and for high gain parts, since it sounds good also for high gain solos.

For my genre (I play mostly metal) a pickup like a p.a.f. would be too weak for the bridge position, I need higher output ones especially for a matter of low-end tightness and definition, but in the neck position a pickup like this is great, because it captures all the nuances and it's fantastic also when lowering the volume knob.

In conclusion this is the pickup that I usually suggest when someone asks me which passive humbucker to put in the neck position, since it sounds very balanced and pleasant for basically any genre, from jazz to metal, and you should check it out too, if you have the chance.

Thumbs up!

Specs taken from the website:

- DCR: Neck 7.60k

- Magnet: Alnico 5

- Cable: 1c Braided Shield, 4c Shielded

- Type: Passive

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Clean guitar tone from a distorted one (and why you should use it)

Hello and welcome to this week's article!

Today we dive back into the guitar world talking about tone, and specifically about a trick that has been kept secret for quite some time: the perks of getting a good clean tone by rolling off the input level that enters a distorted channel.

Usually when dialing in a clean tone we start from the clean channel of an amp, whether it is a real one or a plugin, and we work our way around from there, but there is an alternative way that in some case can sound even better: in some amp, in facts, the clean channel can sound a bit too weak, too cristalline, too "cold", while sometimes the type of clean we are looking for is richer in harmonics, at the limit of the breakup (the point in which it starts saturating and entering almost in the overdrive territory); we can be looking for a tone that is more full in the lower-mid range, and not always this is obtainable from the clean channel.

So what do we need to do? Do we lower the volume?

No. That one can stay fairly high, for example at noon: what we need to do is to lower the input level (if we're using a plugin) or lower the guitar volume by moving the volume knob in the guitar, and this way the signal coming into the amplifier will be weaker (but make sure to turn off any noise gate, or else it will fall below threshold and be cut).

Once the guitar signal will be lowered, the amp will not have anymore enough signal to produce a distorted output, and will give you a very warm tone, much fuller and warmer, and almost "out of focus" compared to the "zing" of the clean channel, in which you can really hear the brightness of the string. 
Now it's time to adjust the master volume in order to compensate for the lower input volume, and this (especially if we're talking about tube amps) will add an additional layer of saturation and harmonic richness, so dose it carefully.

Finally, note that this method of obtaining a low gain/clean tone will need eq adjustments in order to restore some brightness, and maybe to tame some low mids, which will become quite prominent.  

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Review: POD 2.0

Hello and welcome to this week's article!

Today we're going to make a long awaited legacy review from one of the most important pieces of guitar hardware of the latest 20 years: the Line 6 Pod 2.0!

This revolutionary piece of gear (that we have described along with its other iterations in our article "The Battle of PODs") came out in the year 2000, and (along with the first model, which came out 2 years before) contributed greatly in making digital amp modeling a viable option not only for beginner guitarists but also for professional ones.
How did they do? Well I guess the technology was starting to be mature enough to allow the guitar player to obtain sounds that were a bit richer in harmonics and fuller in terms of frequencies, compared to the '90s digital processors, which often sounded like fingernails on a whiteboard and were not a viable option if you wanted tone to do anything more than doodling in your bedroom.

This 2.0 version has been the one that exploded, in terms of public recognition, and became very popular both in studio and live environment, because it was user friendly and yet packed several interesting features, such as 32 amp models which could be used to drive a real power amp or mixed and matched with the integrated cabinet models, 16 effects, several inputs and outputs, and 36 presets.

The amp models were covering the whole guitar spectrum, from the classic Fender cleans to a high gain Mesa Boogie Dual Rectifier (plus there was also some custom Line6 model such as the "Line6 Insane"), the cabinet simulations were available in several brands and sizes (for example 1x12, 4x12, 4x10 etc...) and the effects were the most common ones and there were also several effect combinations.

How does it sound? As I have said, compared to the previous digital amp sims such as the '90s Zoom ones there is a huge leap forward: for once the high gain tones are usable, and even if the distortions are still quite scratchy, there is already a decent amount of musical mids, and also the effects are more than usable.
Probably the biggest bottleneck today is represented by the cab simulators, which today sound quite dated, but already someone on Youtube has shown that by using this unit together with modern impulse responses you can still obtain sounds which are competitive today, and this tells us a lot: the cab simulation is still today probably the most important part in a digital guitar rig.

Does it deserve still today to be bought?
Actually the technology of POD has been used, with some update, for basically 20 years, for example in the Spider Combo lineup, or in the Pocket POD and so on, so the hardware has proven to be good and reliable, but today the latest iteration of the POD is the POD GO, which features a completely different hardware and software: it is based on the one of the Line6 Helix, which is much more advanced.
If you are considering of buying anything from the Line6 family, probably the POD GO today is the bang for the buck to check out!

Thumbs up!

Specs:

- Amp Models: 32, customizable

- Digital Effects: 16

- Factory/User Presets: 36

- Headphone and Direct Out: 1/4 in. TRS

- Mono/Stereo: Stereo

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What is the right buffer size to record and to mix?

Hello and welcome to this week's article!

This article is a look more in depth to a topic that we have already seen here: buffer size and latency.

Basically the idea is to record at lower buffer size and mix at higher buffer size.

Why? Because the lower the buffer size, the lower the latency, therefore if you are using for example a virtual amplifier it will sound more "real" under your fingers, but it will be heavier on the CPU, while with a higher buffer size there will be more latency, but if you're mixing this is not a problem: it doesn't matter if it takes even half a second or more between when you press play and when the reproduction starts.
When playing, instead, having half a second of delay between when you play and when you hear it makes it literally impossible to play an instrument.

Is there a sweet spot between the settings? It depends on 2 factors: 

1) how powerful the computer is

2) how heavy is your session: the more plugins you have, the harder it will be for the CPU to process everything in real time.

That's why it's suggested to use different settings when recording and when mixing: when mixing you can keep it around the maximum settings (1024, or 2048), which makes the project a lot less CPU intensive.
When recording, instead, the aim is to keep the latency at around 10ms or less, and usually this is a setting that you can obtain with 128 or 256 buffer size. 
There is no real need to get too low, like 64 or lower, because this way unless the project is basically empty, you can start experiencing clicks and pops here and there, because the CPU struggles too much.

And you? What buffer size do you use when mixing and mastering?

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