Behind The Mountain
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Behind The Mountain - Recording Tips II


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A Compression Kickstart
by Ken Lanyon (Slider)

Compression has to be one of the most confusing and elusive effects out there. It's easy to know you need it just by watching your meters, but what does each knob and button really do and how does it all work? This article should answer those questions, and will touch on the "whens" and "whys" of compression.

Let me first start by explaining the basics of dynamic ranges in recording. First, we have the noise floor. This is the lowest level, where tape hiss and electrical hum reside at. Next we have the nominal level, which is the level that is best for recording your incoming signal in order to minimize distortion and overcome the noise floor. The distance between the noise floor and the nominal level is called the signal-to-noise ratio. Next is the maximum level, which is where distortion occurs at when your incoming level reaches it. This is the highest level in the total dynamic range. Distortion is something that you definitely want to avoid unless you are versed in the skills of good tape saturation (sometimes engineers will try to slightly distort the signal by pushing it over the maximum level because this will give a stronger sound to an originally weak one. However, in digital recording, any distortion due to overpeaking is distasteful.). Now the difference between the nominal level and the maximum level is referred to as your headroom. This is your safety zone, and this is needed to account for some stray peaks here and there without hitting the maximum level. And to wrap this up...the whole thing, from noise floor to the maximum level is called the dynamic range.

Okay, lets cover how compressors work. Imagine a recording scenario where you are starting to record some tracks on your multitrack recorder. You have set a good recording level for your instrument which is at or near the nominal level, but you notice that the incoming signal occasionally jumps up into the red. That is typically going to be the nature of either the instrument, your playing, or both. So, you don't want those distortions going to tape and ruining an otherwise fine performance. This is where the compressor comes in handy.

The Alesis Company a while back issued a brochure on how compressors work, and it gives the analogy of the compressor acting like your own dedicated engineer for that one track. It will monitor all the incoming signals and then act like it is pulling down the fader the instant that high volume peak occurs. In a more technical explanation, what the compressor is actually doing is reading the incoming signals, and then according to the compression ratio that you set, it knocks the hot signal down by that ratio. This allows you to keep the level down to one that is manageable and recordable, without the wild peaks.

Compression ratio you ask? Well, let me explain the 5 main controls. First, we have the threshold. Think of this as the decibel level where the compression will start working. I visualize the threshold as a line that is lowered onto the offending noise peak, and the lower the threshold level, the more the incoming signal will be compressed. This is because more of the noise peak is now ABOVE the threshold level so there is more to squash.

Next we have the ratio settings. This knob has different ratios on it like 2:1, 3:1, 4:1, and usually any combination in between. Okay, assume you set your ratio to 3:1. What this does is that for every 3dB your incoming signal goes over your threshold line, the compressor will allow only 1dB to pass. The level still goes over the threshold, but assuming that you set the threshold low enough and used an appropriate ratio, the peak will never reach the maximum level and distort. This is also due to the amount of headroom you have. Typically, I set my ratio first, and then use the threshold knob to find the point that the incoming levels are being compressed. This is done while watching the meters on the mixer, and you will see the offending peaks all falling within the same lower range which is nearer to the nominal level. Keep in mind that if your incoming signal is lower than the threshold level, (or the threshold is set too high), then none of the signal will be affected.

Next we have the attack parameter. Think of this as how fast the compressor acts on the peaks once they pass the threshold. Some instruments have a really quick attack sound as soon as they are played, and most peaks arise from this attack. Therefore, on instruments like bass and kick drums, you would want to set a quick attack.

The release parameter works by setting how fast the compressor lets go of the incoming signal once it has gone below the threshold level (where the signal doesn't need to be compressed anymore.) You could set the release to fast and cut off a signal quickly, or set it to slow which results in a longer sustain. Many guitar players use this to sustain their notes.

The last main function is the output setting. Typically, when you lower the threshold and the compressor kicks in to squash the signal, your nominal level will be lowered slightly depending on the amount of compression being used. You can then use the output knob to bring the input level back up to nominal. Be careful though, because by raising your signal back to the nominal level, you are also increasing the noise floor due to added noise from within the compressor itself. You may want to increase the trim on your channel or master fader so more pure signal is getting to the compressor instead. Everytime you patch your signal through another pathway (such as another processor), you are also adding the inherent noise of that pathway.

There is one other feature that not all compressors have, and this is the option to compress with "hard knee" or "soft knee". Hard knee is where the signal is compressed the moment it goes above the threshold to the full extent of the ratio that is set. Soft knee is where the compression is applied more softly so as not to sound so abrupt. This is similar to using the attack knob, and I use hard knee compression on signals like bass and kickdrum.

Hooking up a compressor is a simple procedure involving an insert cable. This is a Y configuration cable with one 1/4" TRS connector that splits out to two 1/4" connectors. One of these connectors is an RS and the other the TS. (I should mention here that TRS stands for TIP -RING-SLEEVE, with the tip being the send and the ring being the return. This way, the TRS connector allows signals to go both ways, and the TS connector allows on signals to send FROM the compressor to the mixer while the RS connector returns the signal from the mixer to the compressor.) The TRS end is plugged into the insert jack on one channel of your mixer, the TS to the compressor send, and RS to the compressor return. This creates a loop where the original signal leaves the mixer, goes to the compressor, is then compressed, and finally returns to the mixer.

As for using compression, that is a matter of personal preference. I use it only when needed. Unless I am going for a certain type of effect by heavil y compressing the signal, then I use it only for stray peaks, since putting a signal that isn't peaking through a compressor will only introduce more noise. Some people think that even though the signal is peaking out during recording, they can compress the signal in the mix and it will be the same. I used to think that myself but I realize now that when you put a distorting signal to tape, the damage is already done to that signal's sound. The track is already saturated with distortion and no amount of compression during the mix will make it sound as if it were compressed during tracking. That is why you should definitely fix stray peaks with the compressor when recording. Also, final mixes may also need a little compression even if you used it on tracks during recording. This is due to the summation of all the track signals.

The following are just suggestions of where to start setting your parameters for certain instruments. As I mentioned earlier, how YOU want to use compression is your personal preference.

Bass: Try starting out with a ratio of 4:1, and a fast attack and release. I usually use the hard-knee type of compression here since bass is such an attack-oriented instrument. But if you were playing smooth jazz bass, then you may want to try soft-knee. It depends on the sound you are trying to get.

Guitar: This depends on the type of sound you are using, but a good general place to start is 2:1 on acoustic, and maybe 3:1 on distorted guitar (although you may need 4:1 here.) To get a good sustain, try a 4:1 ratio, use a fast attack and slow release. Then play the note you want to sustain, and raise the ratio until the sustain is as long as you want it.

Drums: Drum signals are often compressed due to their hard-hitting attack volumes. If nothing else, compress the snare drum, because each hit will likely peak higher than other hits. Try starting out with a ratio of 3:1, and use a fast attack and release. If the signal is still peaking, try using a ratio of 4:1. This method could also be applied to the toms. As for cymbal hits, try starting with a 2:1 ratio (moving to 3:1 if needed), using a fast attack and a slow release (to preserve the natural decay time of cymbals).

Vocals: As with drums, compression is also commonly used on vocals. The ratio to start at varies for each singer, since some may be very strong and loud singers, and others quieter, having a smaller dynamic range. Try starting out with a 2:1 ratio, with a fast attack, and medium to slow release. Keep increasing your ratio until you get your peaks under control.

Compression is not typically something that can be heard. You can hear it if you really spank all the knobs to full-on, but usually that technique is used more for an effect, rather than to control the level of the individual signal. Compression should be applied and monitored by using the peak display meters on your compressor or mixer. As I mentioned earlier, compression is something of an art, and you will have to play with it to find your personal preferences, so don't be afraid to tweak all the knobs to find out how they affect your sounds. Just remember that mastering compression techniques will help to make all of your recordings sound more professional.

(c) 2000, Ken Lanyon,
All rights reserved.


(You are allowed to copy and use this essay for your own non-professional use. You are prohibited from distributing copies to others for a fee or for no-charge. You may not publish or quote this essay without obtaining the written permission of the author.)


An Introduction To Mastering
by Stephen J. Baldassarre (Silent Bob)

As a mastering engineer, many people have asked me about the importance of mastering. However, in order to thoroughly describe the importance of mastering, I must first describe some of the equipment and processes available to a typical mastering engineer.

The equipment used by mastering engineers is very specialized and precise. Most people have dynamic compressors in their studios but the compressors used in mastering are a bit more complicated. For instance, I use compression that can control high and low frequencies independently. It can catch peaks in the audio signal instantly or before the peaks even occur. This compression uses joint stereo operation which means that if a peak occurs on one channel of the stereo mix, both channels (right and left audio channels) with be attenuated equally. This is important because if only one channel is attenuated, there will be a sudden loss in one channel's volume which will interfere with the soundscape. Joint stereo operation also prevents stereo separation from deteriorating as compression is increased.

Most people are also very familiar with equalizers or EQ. The EQ used in mastering can affect both right and left channels independently or identically. This is useful if the right and left channels have significantly different frequency content or if there is an error in one channel and not the other (if it ain't broke, don't fix it). Also, I can use EQ from a ten-band analogue EQ all the way to 2,400 band digital FFT filters. FFT means Fast Fourier Transform, which is a method of processing a digital signal using discrete amounts of delay to control independent bands of frequencies Why so many bands? Precision, that's why. I've mastered songs with high pitched ringing going on throughout caused by substandard equipment or from having a computer too close to the recording gear. Normal EQ could eliminate such sounds but would cause severe interference with the rest of the program material making it sound unnatural. The digital EQ is so precise that it can eliminate the ringing without any audible effect on the program material. It can also be used for split seconds to reduce bum notes or add a little accent to certain instruments without affecting the surrounding material. This is very useful for increasing clarity and overall impact of the sound.

Nonlinear editing tools such as a software controlled hard drive system are also important for removing sections of sound for the purpose of making different versions of songs for radio or album cuts, CD singles etc. Fixing bad "punch-in" glitches, and cleaning up fades are also advantages of nonlinear editing tools. The same tools are used to put the songs or other material in the correct order and set the correct timing between tracks on CDs. Dynamics can also be added with great precision to program material using a nonlinear editing system to increase the impact of the sound. One other real advantage of a nonlinear system is the ability to reduce transients (occasional sudden volume peaks), which prevent the overall volume of the material from being increased. After stray transients have been removed, the signal can usually be boosted 3-9dB louder than before.

Noise reduction is also a very handy tool in mastering. The same FFT filter used for EQ can also be used to remove AC hum, tape hiss (to a limited extent) or other unwanted noises such as clicks and pops. If there is noise in a particular track like AC hum, a segment of the track containing only noise can be sampled in the FFT as a profile for noise reduction. This profile is applied over the entire selection and (hopefully) attenuates the noise. This is incredibly useful for restoring older recordings, but many new projects I've worked on have also benefited from this process.

Mastering engineers also have the ability to widen the stereo field of recordings, even if they were originally recorded in mono. Granted, if you send a mono recording to a mastering house, they cannot, for instance, pan the guitar to the right and the keyboard to the left, but they can add stereo space that was not there originally. If the recording is done in stereo but just does not have the aural space it needs, then the stereo field can be accented, creating an improved soundscape. There are several methods of doing this that can only be done in the digital domain, but some methods are done using specialized analogue processors.

One of the last mastering tricks I should mention is time stretching. A song's tempo can be increased or decreased without affecting the pitch of the song. This is important for making radio edits of songs, as radio programmers have a tendency to speed up songs in order to fit more commercials into the day. The tempo of the song can be decreased so when the radio station speeds it up, it will have the tempo it was originally intended to have. There isn't a large demand for this process, but some people wanting to make their tunes more danceable or to cheat the radio stations like to have this option.

So when people ask me what the importance of mastering is, I could sum it up into just a few short statements. Mastering increases the impact and clarity of the material. It is the final polishing an album as a whole receives before it is released to the public. Final touches on fades, song order and volume are all made here as well as some correctional touch-ups.

Who should have their stuff mastered? Anybody looking for a more professional sound in their work should have their material mastered. Mastering is a key process in bringing recordings up to commercial standards. Home-recorded demos all the way to industrial studio recordings can benefit from mastering, which is why I stress the importance of it so much. Industrial studios have their material mastered religiously to gain that extra edge. Many audiophiles have their material mastered to compete with the industrial studios, and musicians with homemade demos may have it done just to increase the impact of their sound for promotional use. So mastering can serve anybody who is looking for a more professional sound in their music. For audiophiles, it is a great help for achieving the perfect sound. For industrial studios, it is a step all to important to skip.

(c) 2000, Stephen J. Baldassarre,
All rights reserved.

(You are allowed to copy and use this essay for your own non-professional use. You are prohibited from distributing copies to others for a fee or for no-charge. You may not publish or quote this essay without obtaining the written permission of the author.)