Daking Audio Gear: Mic Pre IV and 3 FET II Compressors

I currently have three of these units in my studio right now and I have had a chance to really put them through their paces.

First, I should say that the sound quality on these units is pristine. There is very little coloration of the sound even when using heavy compression. Many compressors seem to roll off high end when they attenuate heavily, but this is not the case with the FET II. The FET II uses Jensen transformers both in and out of the unit and the pc board is extremely clean and well designed. The FET is in a socket so if it were ever to go bad, it is easy to replace.

The FET II excels at transparent compression and is easily used on bus or program material where lesser compressors really start to sound yucky. The attack times vary between 250 micro seconds to 64 milliseconds and it’s fast enough to be used effectively as a brickwall limiter if desired. The release characteristics are I think what really set the compressor apart though. You have some standard settings of .5 – 1.5 seconds, but also some really nice dual time constant releases designed to mimic some of the nicest compressors in history. The idea behind dual time constant release is this: the compressor releases a little fast at the beginning and then slows down. This effectively eliminates the “pumping and breathing” sounds associated with more abrupt release times.

I have also been able to get some really nice vocal distortion (think Flood’s production techniques) out of it by using the fastest attack and release times and a very high ratio (20:1). Then I drive a very hot signal (over +20) and get a very pretty sounding harmonic distortion very appropriate for alternative rock vocals like NIN, PJ Harvey or Smashing Pumpkins.

I recommend using only XLR cables in and out of the unit, you can use a 1/4″ input but it boosts the signal 14 dB to make up for the -10/+4 difference in operating levels between consumer and pro gear. Another odd thing is the power supply (external, but not a wall wart) uses a DB25 connector which looks pretty weird, but works perfectly well. Just make sure your intern doesn’t try to run the power supply into the DB25 input on an audio interface or multitrack….Bad intern! Bad intern!

You can link two units together to work in stereo with a 1/4″ guitar cable. The sidechaining connection uses DC summing to tell the linked unit when to compress and does not send audio. The FET III does audio summing, but it’s in stereo and is geared more towards working in stereo anyway.

All of the knobs on the unit are switches so you can set two or more compressors exactly the same way and repeat your settings later on. The knobs are really heavy and feel like you’re really working with pro gear.

All in all this is a great compressor with excellent transparent compression that doesn’t color the sounds you are working with. You can use it to chase the waveform to create harmonic distortion with the fastest attack settings to add a little crunch to vocals, bass or drums.

I can’t recommend it more highly.

Their is difference between someone who makes beats – meaning composing and performing (or programming) original instrumental music, someone who is really a producer, and a recording engineer that specializes in hip hop tracking and production.

The fastest way to learn to beat making is to make beats with whatever you have available. I have worked with a couple of heads who were complete geniuses with the Playstation software from MTV. Their music was simply amazing. Software that is highly under-rated is FL Studio or FruityLoops. The step sequencer is the easiest way to make music quickly. Read the manual! Watch videos online.  Start working with as many other beat makers that you can find on the net, in your home town. For me, competition made me write stuff that was much better than working by myself in a vacuum. The three big instruments to learn would be keys, drums, and bass. You did not need to work in a studio to do this kind of work. You need a computer, a decent audio interface (Not an M-Box), and a couple of nice monitors. If money is a factor, don’t get a Mac. You get a lot more computer in the PC world and there’s tons of software available.

A real producer puts the whole show together. They hire everyone, often write songs with the artists, choose the studio to work in, find live musicians to fill out the sound. Sometimes that means doing everything yourself. A lot of the time the producer FUNDS the project and gets the biggest share of the profit (if any).  A producer is a big picture person usually with an excellent understanding of the psychology of creative people, motivation, fear, competition and excellence. This is something that comes with lots of experience, a strong musical background, charisma and usually fame or money.

An engineer deals with the tiniest details of tracking and mixing. Moving a mic a half inch, rotating a mic off axis, how to attenuate the peaks of the kick to get it to sound bigger, without making it wimpy. Attack and Release time minutia for compressing drums, bass and vocals. How the sound stage can be used to the best advantage, how to either avoid masking or use it to create new timbres. You need to learn this either in a studio as an apprentice, in a good audio school that has great facilities (I teach at New England Institute of Art in Boston and at U. Mass Lowell both have great facilities) and then leverage that into getting good internships.

Sometimes there are people who really are all three. Sometimes you will find yourself in one role or the other depending on who you’re working with.

The best job to get to learn audio engineering is working for live sound companies as a grunt. You will carry the bass bins, mic stands and a 43 foot console. But you will get to watch the FOH and monitor guys throw down. Live is good because it forces you to learn to do things quickly and it puts you around dozens of musicians every weekend. Not wanting to be embarassed is a very powerful way to learn.  You are always on stage being watched from the time you load in, to the time you strike the stage.

(posted to GearSlutz 7-4-09)

One of the hardest audio processors to understand is the compressor.  Even after several years of using compressors many of my students and readers still have lots of questions about how to dial in the sound that they are trying to get.  Compressors are in the Dynamics Processors family which also includes limiters, expanders, gates and noise reduction.  Dynamics processors work in the Amplitude Domain.  Compressors work on the amplitude of an audio signal, which is basically the loudness of the signal.  Look at a waveform view of an audio signal:

Graph of a Sine Wave with Amplitude and Frequency

The vertical axis shows Amplitude, which in analog (electrical)  audio refers to the amount of voltage in an analog signal. When the wave is above the center line, then the voltage is positive and when the wave is below the line the voltage is negative. Audio (in the electrical analog sense)  is AC or Alternating Current which means the voltage goes from positive to negative and then back again. The further away from the center line, the higher the voltage and the louder the wave will sound.

The Dynamics of music is generally thought to be the differences between the loud parts of music and the quiet parts of the music.  The dynamics of audio includes all of the differences in amplitude along the waveform.  In most pop music, for instance, the loudest parts of the music are the snare drum hits, followed by the lead vocal, then the background music. Notice in the following image the red dots above the waveform.  They are marking the locations of the snare and kick drum hits in the music.

The red dots mark the locations of the snare and kick drum hits.

Notice that there is audio in between the loud hits as well, but that it just has a lower amplitude. Compressors and all dynamics-based effects work on the amplitude of the audio, to adjust and change the differences in voltage.  The loudest level in digital audio is 0 dB Full Scale or (0 dB FS) which means that anything above that level will be distorted or simply just an error.  We can’t change the loudest possible level, but we can change everything that is below that level.

What a compressor does:

A compressor attenuates (decreases amplitude) audio that is above a threshold by a ratio.  The attack time is how quickly the compressor starts to attenuate the signal after the threshold is exceeded and the release time is how quickly the compressor stops attenuating the signal when the audio drops below the threshold level.

Probably the most common use of a compressor is to make an audio signal sound louder without peaking out the signal and causing clipping and distortion.  In a nutshell, the loudest parts of the audio signal (the peaks) are made a little bit quieter so that all of the signal can be boosted by the amount that the peaks were attenuated.