Ultimate guide to setting up patches on a guitar multi-effects processor.

My prediction is that multi-fx units will get better, bigger and more ubiquitous. Obvious, isn’t it? Sure, a POD can never replace the grandiose aesthetics of a stage decorated with amplifier heads and cabinets. However, the big factor that works in favor of multi-fx units is convenience. Since this article is not supposed to address the ‘tube vs digital’ debate, I will not go into further details. However, if you hold any dogma against digital gear then this article is certainly not for you.
A multi-fx unit, typically, tries to simulate the different components of an analog signal chain that connects a guitar to its final stage: the P.A system. One particular thing that is common among a lot of new multi-fx users, particularly those who are dissatisfied with their units, is poor understanding of the functionality of the individual components and the relative order in which they are to be placed. I will only discuss standard practices. These are not strict rules, and there are plenty of guitarists who have managed to achieve tonal-nirvana without strictly following these practices.

The interesting thing to note here is that most multi-fx units (in fact all that I know of), are designed to simulate everything in a standard signal chain, excluding of course the guitarist, the guitar and the PA system. In this article, I am going to describe the individual components in a guitar signal chain and their functions, and the different stages of the signal chain.

Stages in the signal chain:

The guitar signal chain has an amplifier, manufactured by big names like Marshall and Mesa Boogie. An amplifier’s job is very intuitive; an amplifier amplifies.
The amplifier is arguably the most important component in the chain, and its position in the chain, with respect to other components in the chain, is vital.
Anything that is connected between the guitar and the amplifier is in the pre-amp stage. Typical preamp effects for a guitar are Wah, Volume, Distortion, Compression and a Noise Gate.
Anything that is connected to the output of the amplifier is in the post-amp stage. Typical post amp effects are delay, chorus and reverb.
There are plenty of guitarists who like chorus at the preamp stage and as many who like it in the post amp stage. However, some effects have a more fixed place. For eg. What would be the point of sticking a distortion/boost pedal after the amp?
So how does one know what goes in front of an amp, and what comes after? Well, it depends on what you want. Let us take the example of the delay effect which is ideally placed post-amp. So what does delay do? It simply clones the signal and mixes it with the original signal with a time lag. The effect of placing the delay pedal, in the pre-amp stage is significantly different from placing it post amp. Normally, one should be able to hear the difference and use his/her ears to make a judgment. For those who are more technically inclined, refer to this diagram.

Components in the signal chain:


Pre-amp :

Noise gate: The noise gate is usually the first component in the chain. Place it somewhere else only if you know what you are doing.
All signals have some noise, and the signal from a guitar is no different. A Noise gate is used to reduce this nuisance. A noise gate normally has a threshold, above which it is open (and below which it is closed). Thus, part of the signal that is above the threshold passes through the gate, and that which is below the threshold level is not allowed to pass through. The important thing to note here is that a noise gate does not remove noise; i.e. below the threshold, when the gate is closed, it doesn’t allow anything to pass through, indiscriminating between the desired signal and the unwanted noise.
Thus, if the threshold is too high, you would lose desired nuances which are at a lower volume than the rest of the signal. If you go higher, you might lose the actual signal itself.
You set it too low, and noise passes through.
The human skill is in setting the threshold at the optimum level.
Some noise gates have additional parameters that offer more control:
Range/Attenuation: This denotes how far the gate will close. A higher setting will fully close the gate, not letting any sound to pass through below the threshold. A lower setting will allow some sound below the threshold to pass through.
Attack: This lets you control how fast the gate closes when the sound falls below the threshold. A lower (faster) setting immediately closes the gate, once the signal is below a certain level. However, a very fast setting can introduce artifacts in the form of clicks and pops. A higher (slower) setting, gradually closes the gate (much like a fade-out).
Release: This is the opposite of attack. The ‘release’ parameter gives the user the control of how soon the gate re-opens, once it is closed.
Some gates have an additional control called hold, which lets the user control the duration for which the gates needs to be open or closed.

Distortion/Boost pedal : A distortion or a boost pedal, before the amp, can do wonders to your tone. Such an effect before the amp enables one to send a louder signal to the amp. The pros of doing this is that you get more sustain, and more distortion without pushing your amp to 10. A distortion pedal normally has a drive knob and a gain knob. Typically, one sets the gain anywhere between 20-70% and drive between 0-50%. The more gain you add, the more distortion you would get out of the amp. Drive does almost the same thing, but in addition colors the tone with the characteristics of the pedal/effect. So if you have a tubescreamer set at 60% gain and 0% drive in front of a JCM 800, your tone will sound like a JCM800 that has been pushed really hard. If you have gain at 40% and drive at 20%, your tone will sound like a JCM800 that has been pushed really hard, but will have some characteristics of the tubescreamer as well.
A lot of multifx boxes have digital models of popular pedals. However, some fx pedals don’t offer pre-amp distortion models and often have a single control called pre-amp gain. The cheapest multifx boxes have inbuilt preamp gain in the amp models themselves offering the guitarist little control.

The amp : The guitar signal, which has been through the pre-amp components runs into the amp. Amp settings: There isn’t a lot of explanation to be done here. There are just 3 rules.
(1) Gain: Less is more. Don’t push it up like a maniac.
(2) Mids: You can never have enough mids. The more the mids the better.
(3) Bass, Treb and Presence: Again, less is more. Only have as much as you need.

Cabinets: Choose the cabinet model as you please. If in doubt, use the cabinets that your favorite guitarists use. 4 X 12 is the most common size.

Microphone: Not all multifx boxes offer mic models. In real life, the sound that comes out of a cabinet is always captured through a microphone. Normally choose a dynamic mic like the SM57 (which is almost the standard), and works really well for almost all forms of music and works especially well for metal. If you want more shimmery stuff, you would want to use a condenser.

*For the technically inclined; A cabinet/speaker convolves something similar to a multiband dynamic compressor to the signal that is played through it. This makes the guitar sound a lot more musical. Try playing your guitar through an multi-fx unit with the cabinet simulation switched off and you should be able to hear the difference.

Post-amp :

Delay : Refer to the section above. Delay simply clones the signal and mixes it with the original signal with a time lag. Delays have a time control, which lets the user control the time lag; i.e. once the original signal is played, how long should the pedal wait before playing the cloned signal. Another important control is often called ‘mix’. This controls how loud the cloned signal is when compared to the original signal. Some delay pedals let the user control some other parameters including the equalization on the cloned signal.


Effects that can be placed Pre or Post-amp :

Wah: Wah in technical terms is a spectral glide. Of course, as a guitarist, one doesn’t need to know that. However, it is one of those cool concepts that you might want to check out. Wah settings vary from guitarist to guitarist and completely depend on the playing and accentuation styles of the guitarist. Where to place the wah is controversial as a lot of people like adding the wah in the post-amp stage. There are two distinct camps, and one needs to figure out their preferences, himself/herself.

Compression: Compression plays with the dynamic range of the signal; i.e. the difference between the loudest and the softest parts of the signal. One can use a compressor to decrease (or increase in the case of an expansion) reduce this dynamic range. Thus, once a signal is compressed, the softer parts appear as loud as the loudest part of the signal.
Compressors can be used in the pre-amp stage, or post amp. There is a sonic difference between both the cases because in the first case the amplifier sees a compressed input.
A compressor has a threshold control. What it does is that when the signal breaches this threshold, it reduces the level of the signal; reducing the entire dynamic range. Typical controls on a compressor are:
(1) Threshold: level above which signal is to be reduced.
(2) Attack: once the threshold is breached, how long should the compressor wait before reducing the signal level.
(3) Ratio: The amount by which the signal is to be reduced once it crosses the threshold.
(4) Release: Once the signal is reduced, how long should the compressor take to come back to it’s original level.
Say a compressor is used with the following settings: Threshold: -26db, Attack: 30ms, Release: 150ms, Ratio: 4:1
A signal reaches –20db; i.e. 6db above threshold. The compressor will wait for the Attack time to get over, before acting; i.e. 30ms. After 30ms, the compressor will reduce the additional 6db by a ratio of 4:1. Thus, it will reduce 6db to 1.5 db. So the compressed signal now is at a level of –21.5 db. Once the compressor does this, it waits for the release time (150 ms) before it activates itself again. After 150 ms, it waits for another peak in the signal to cross the threshold. Please note that when below the threshold, the signal is not affected by the compressor at all.

EQ : One uses EQ to cut/boost particular frequencies. It is always advisable to cut, as opposed to boost (boosting often results in phase anomalies).
Some important guitar frequencies ranges are:

<140-150>8kHz: Hardly any frequencies will exist in this range.

When playing live, referring to something called the Fletcher-Munson effect is essential. The human ear perceives different frequencies differently at different volume levels. At live-gig levels, humans perceive bass and treble more prominently than mids. Thus, before playing live, one should cut bass and treble and boost mids.

I normally prefer a post-amp EQ because of the way an amplifier works. When an amplifier boosts a certain frequency, it also effects other frequencies. So, if 100 Hz is notched out at the pre-amp stage, the amp will have less to work with. Also, an amp can introduce the very same frequencies that I had cut earlier with the EQ. So, for more control I prefer the EQ to be post-amp.
Please note that this is just how I like to work, and not a strict rule. For instance, to change the sound of guitar pickups, one would want to use pre-amp EQ.

And finally, what to do and what NOT to do:

Do’s:
(1) Have more mids
(2) Keep your cabinet on
(3) Try to understand when you would want a particular effect on
(4) Do not forget Fletcher-Munson
Don’ts
(1) Never scoop out mids, keeping bass and treble high, unless you want to be inaudible.
(2) Never have too many effects like chorus/flange on, without knowing what they do.
(3) Avoid lots of delay.
(4) Never turn off the cabinet simulation, unless you are using an external cabinet.

So, that is it. Have a good time setting 'that' tone.

Gear whoring begins…

Since I am royally bored at work, I decided to put together a basic home-recording setup. Of course, the idea is plagiarized from Tweakheadz. Nevertheless, I’ll try to ensure that things here remain different than Tweakheadz and that I have something different and useful to offer.

The setup that is seen here, is really simple and meant for the guy who wants to record himself, or simply put some ideas on tape. It would be difficult to produce a quality mix on this setup, but the capabilities of cheap/affordable gear has always surprised me. I won’t feel flummoxed if I hear a pro-quality mix out of this setup.

What’s the setup all about?

Soundcard: Lexicon Alpha; A very basic USB soundcard. Capable of handling 24 bit/48 Khz. The unit has one XLR mic input and one instrument input, the adequate number required by the musician on a tight budget. The preamp sounds surprisingly good, for a unit at this price. It doesn’t add anything to the sound, but doesn’t take away much either.

This is a substantial upgrade from a stock-consumer soundcard. One of the cool things are the two RCA outputs. Lexicon guessed, and rightly so, that a lot of their prospective customers would be beginners who do not own studio monitors. The RCA outputs are useful in case you need to connect the soundcard to stock computer speakers.

Monitors: M-Audio BX-5A; Definitely the most important piece of equipment in the signal chain. Now, these are not my favorite monitors. However, this definitely is one of the cheapest monitors available that can do the job. There are a few monitors in this range that in my opinion are better but, as I have mentioned here, opinions are subjective. Other monitors, in this range, that one can check out are Tascam VLA5s, Tannoy Reveals, Yamaha HS50, KRK RP5s etc. One can also push his/her budget a little and go for slightly better monitors like the Yamaha MSP5. I don't recommend a pair with speakers less than 4''. That just won't be adequate, unless one has special needs. If one has a small room, going for something too big is not a great idea either.

Mic: Shure SM58; One cannot go wrong with this. There exist very few studios who do not own an SM58 or SM57.

**One caveat though; if you want to record music that is centered around vocals, a la Enya, you would want to pick up a condenser mic and a soundcard with phantom power.

Headphones: Sennheiser HD280 Pro: A very good pair of headphones and a must have.

MIDI controller: M-Audio 49; Basic MIDI controller, and does exactly what it says. Don’t expect more or less. If you don’t like the idea of keyboards, you may want to look into other ways to programme MIDI. One method that I can recommend is using a software, with a basic midi editor, like GuitarPro.

Overall, I just love the simplicity of such setups. Simple, but capable of a lot. The best part is that all of this combined costs around 50k INR in India. In US, the setup would cost less than 900 USD. If I leave out the monitors for now (not recommended but most people do so anyway), your costs would come down to 30.5k INR and 600 USD in India and U.S. respectively.

P.S. I am not a lawyer. I do not own the images published here. If anyone has any problems, related to copyrights of the image posted, please get back to me and I will remove them.

Ambiguity

Audio engineering is widely accepted to be the art of producing the “best” audio signal. Not really, because there is nothing called the “best signal”. As audio engineers we try to produce a signal that, in our opinion, is most appealing to a target audience. Of course, this is where most of the ambiguity begins. What an audience may or may not like, is very difficult to define.

The human ear, of course, is a lot more than a simple “20-20khz” acoustic wave receiver and interpreter. Frequencies in this range have a variety of palpable effects on us. We interpret different frequencies in different ways, and we generally do not like any frequency “missing” even if that, what is missing, has merely a subtle effect. However, if our requirements were as simple as ensuring the presence of every frequency in a given range, audio engineering wouldn’t have been a skill with substantial demand, and neither would you be reading this article.

We do not like “too much” of anything, and the margin that defines “too much” is so narrow that even the best audio engineers make mistakes. Of course, what adds to the complexity is that an audio engineer is also required to address the fact that the human perception of different frequencies varies with levels of loudness, and that people have vastly different speakers on which they would reproduce a mix.

I can go on forever with the list of ambiguities in audio. However, I can talk of another form of ambiguity; difference in opinions. Audio engineers, from professionals to the amateur home-studio owners, do not seem to agree on the right kind of gear, or the right method to mix or master. The same person who might otherwise, with conviction, state that gear A is great and gear B is the worst-piece-of-equipment-ever-manufactured, can be found equivocating about the qualities of both A and B, when put under a blindfold test.

Our aural sense is limited, and not as advanced or developed as we would like it to be, and simply inferior to our more reliable senses like our visual sensatory system. This leaves ample room for subjectivity and associated terms like “warm” or ”harsh” or “tubby”.

What I find intriguing about humans, and the way we have evolved is, how at a subconscious level we find ways around our physical limitations. For example, we can sense/perceive frequencies slightly beyond the standard 20-20khz-range and, and at the same time the range that we can clearly claim to hear is less than 20-20khz. Yes, ambiguity kicks in again! As I’ve mentioned earlier, our aural sensitivity varies with frequency. Normally, we are most sensitive to sounds around 1-3khz. As we move away from this range, in either direction, our sensitivity decreases. The variation is so sharp that we only perceive very low (<40hz)>19KHz) frequencies, and do not literally hear them. We, and probably other animals as well, have developed mechanisms to slightly compensate for our limited aural range. Why I find such sensing mechanisms, in humans, intriguing is that an understanding of such workarounds that we humans involuntarily use, can help in producing a better signal. For e.g. a standard way to make a song appear louder, without having to trouble neighbors, is using an equalizer setting that scoops out the mids. Why most people like such an equalizer setting is because it makes a song sound like the way it would have, if it was played louder. Why? Because when a song is played at a louder volume, we tend to hear more bass, more treble and less mids. Clearly, in this case, people exploit a shortcoming in their aural senses.

For more details on this subject, consider googling “fletcher-munson”.

A workaround, at the end of the day, is a workaround and has shortcomings. Audio engineers tend to capitalize on these shortcomings to hide gaps in their signals and thus, try to produce signals that are enjoyable for a wide range of audience.

An interesting and idealistic question that arises here is; “Why should a professional engineer have gaps in his signal at all?” The answer lies in the limitation of technology that we have available to us. Acoustic environment inside studios, in spite of all the effort that goes into treating them, affect the signal, our recording-devices are not perfect, neither are our processing-devices nor speakers. Our rooms, or other enclosures like our cars, in which we listen to our music have an impact as well. An audio engineer needs to compensate for all these factors, and many more. In realistic terms, such a task is virtually impossible. Thus, we resort to workarounds.

How convenient would it be to have a list of all the workarounds, tips and tricks? But unfortunately, there are none. It is simply impossible to compile such an exhaustive list. It is as difficult to find two engineers using the same subtleties, as it is easy to find two of them making the same mistakes.

“Can I have some of your secret tricks?” is a very common question that new home-studio owners ask. I have had a few people ask me “can I have your parameters?” Well, if parameters could help, we would already be seeing people selling them. The truth is that every engineer has his own set of unique tricks and workarounds.

So how does one acquire his personal set of tricks and workarounds?

Let us answer another question first. What does someone, who knows nothing about audio, expect from an audio engineer? Normally, people expect a professional with hi-end-difficult-to-operate gear and an enhanced sense of hearing. Ironically, the primary reason that home recording has become so big is that sound engineers are not mutants with gizmo gadgets. Sure, hi-end studios do stock hi-end, expensive gadgets but it isn’t difficult to understand how they operate. And obviously, no one has any gift for mixing, mastering or recording. Audio engineers do not hear more or less than most humans. They are only able to explain what is right or wrong with a sound and have the requisite tools and skills to enhance or rectify them.

Of course, whatever the artifact, there are certainly more than a few ways to fix them using a variety of tools. The only constant is one’s ears. A professional sound engineer’s skills stem from the hours that he has put in, carefully listening to a variety of music. By variety, I am not trying to imply different genres but albums that are sonically different.

A good listening exercise, to try, is listening to two sonically different albums by the same band, or bands that play very similar music, and noting down the difference in production. The idea here is to tune one's ears so that they can identify subtle differences and draw out plausible reasons for the same.

In this series of articles, I will primarily deal with the philosophy and science that goes into producing a song. In the next article, in this series, I will compare Metallica’s “Master of Puppets” and “And Justice for All” albums and analyze the difference in production styles. These two albums satisfy both the criteria for our listening test. Both are musically similar, at least in my opinion, but sonically very different.