IT'S IN THE MIX: OPTIMIZING AUDIO TRACKS

Have you ever visited the control room of an audio studio? Gadgets everywhere. Patch bays, wires, racks populated with cold, black steel control panels festooned with knobs, pushbuttons, LCD readouts, and blinkity lights. Ooh, ooh, ooh! It gives one a testosterone high just thinking about playing with these gadgets. But, do they really do anything (beyond impressing clients)? Do only audio studios need them? Can mere mortals run them without a doctorate in vibratory science? Let's explore some of these toys and see what use they are to videographers and audio-for-video specialists.

Equalizer

This box is a fancy tone control. With it you can reduce one or more sound frequencies or boost one or more frequencies. Since the world is filled with sounds you want to hear and others that get in the way, by using an equalizer you can filter out the unwanted squawking crows, blowing wind, or humming fans while boosting the sounds of a narrator's voice or a bass fiddle.

Equalizers come in two types: the graphic equalizer which has a number of sliders, each slider representing a different frequency to be boosted or cut, and the parametric equalizer with a knob that dials the frequency you wish to adjust and another knob that determines whether that frequency is reduced or increased. Audio mixers often have several parametric equalizers allowing you to dial one frequency for boosting, another for reduction, and a third for whatever. Equalizers aren't exact: you can't select 60 Hz on the equalizer and excise audio hum from a bad recording, without touching the 59 Hz and the 61 Hz frequencies around it. Equalizers generally effect a range of frequencies and for this reason are used delicately in the music recording business. Slicing one particular frequency range can leave a musical instrument sounding a bit odd.

To the average videographer, equalizers are helpful for reducing the burble of wind when a microphone is used outdoors. A low cut filter (also called a high pass filter) reduces the low frequencies where you find the sounds of wind rumble. Although using such a filter will reduce the normal bassiness of a man's voice, the effect is minor and leaves plenty of intelligibility to the narrator's words. Since speaking voices generally range between 150 Hz and 2000 Hz, there are a lot of highs and lows that can be cut without losing the words. Don't cut them unless you have to, however, because you'll be removing fidelity and ambiance from the sound. You don't want your narrator's voice to sound like it came over a telephone (telephones also have a narrow audio bandwidth). In short, unless you have a nasty noise in the background that you simply must remove so that it doesn't distract from your desired audio, leave the sound spectrum alone and deactivate your equalizer.

Equalizers are sometimes handy for "tuning" a room's public address system. If you have set up your mikes and speakers for a speech or a play and you hear a ringing or whoop or wail as feedback envelops the room, it may be that certain frequencies are bouncing
back to your microphone stronger than others. If you can detect which frequencies those are and reduce them a little, you can reduce the feedback without lowering the volume and effectiveness of your loudspeaker system.

I particularly enjoy the chest shaking depth and power that comes from heavy duty bass, but there are times when low frequencies are our enemies. Low frequencies involve huge amounts of power. They saturate (over magnetize) audio tape leaving no room for the other frequencies. Bass frequencies sometimes suck the power from your amplifier, also leaving little room for the other frequencies, either drowning or distorting them. They pin our VU meters (push the needles off scale) sometimes without adding much useful to the mix. You don't want to waste precious audio resources, be it wattage, magnetism, or digits, on noise or thumping or rumble or hum of equipment, wind, foot stomping, and other non-melodic sounds. Cardioid and other directional microphones tend to exaggerate bass sounds of people speaking into them, and often turn percussive consonants like p's and b's into mini explosions. Fine mesh screens in front of the mikes (called pop filters) may be our primary defense against such phonetic fireworks, but our second line of defense is the equalizer or low cut filter. In short, cut down on the bass frequencies unless there's something useful happening there.

Next consider the musical bass frequencies you are recording; these are the legitimate low frequency sounds of various instruments. You may wish to record these frequencies, but not at their full volume to avoid saturating the tape or exceeding the ceiling on your digital recorders.

If you are recording or retransmitting a mix of sounds, you again may wish to reduce the low frequencies. Low frequencies tend to"muddy up" the sound mix, reducing the clarity of the other instruments. If you have recorded an instrument with full fidelity and are now mixing the tracks together, you may at this time wish to reduce some of the low frequencies to keep the final sound from becoming too weighty, ponderous, or muddled.

Low frequencies are a bit unfriendly to reverbs (described shortly). Again, they confuse the sound space, trampling on the higher frequencies of other instruments.

At the other end of the spectrum are troublesome sizzles, whistles, and hisses, often associated with the letter "s". Some sibilant speakers and some sensitive mikes conspire to emphasize "s" sounds. Screeching s's are about as appealing as fingernails on a chalkboard, so a little high frequency trimming may help. There are also specialized devices called "de-esssers" that, like audio antibodies, seek out and vanquish just that one consonant alone. Equalizers go upstream of most other audio processors. In other words, once your mike signal is preamplified to mixer level or line level, you pass it through the equalizer before sending the signal to reverbs, delays, and other gadgets. This is because those other devices are more likely to respond well to "tailored" sound, sound they can manipulate easily without dealing with troublesome chaff such as booming bass and sibilant highs. The entire signal is fed through the equalizer, treated, and output to the next device, unlike the signal path applied to reverbs and delays. For reverbs and delays, part of the sound passes untouched to its destination while another part of the sound is sidetracked and passed through the reverb or delay device, and recombined with the original sound. We'll hear more about this shortly.

Variable gain amplifier

Variable gain amplifiers (VGAs) modify the dynamic range, the ratio between loudness and softness of sounds. Examples of VGAs are noise gates, expanders, limiters, and compressors. A variable gain amplifer is like a genie with his hand on your volume control. As the genie hears the sound coming into the VGA, he raises or lowers the volume of the output sound nearly instantaneously, and only for a moment. Depending upon the settings, the genie could lower the volume for a millisecond to take the edge off a certain percussive sound. Working just the opposite, the genie could raise the volume for just a millisecond, increasing the punch of the percussive sound. In the first case, the VGA can tame the hardness of a drumbeat. Adjusted to an extreme, the VGA can turn the tap-tap of a snare drum into the chug-chug of a steam engine. In the second case, the VGA accentuates the drumbeat bringing it to the forefront of your audio making it more pronounced.

Adjusting the VGA another way, the genie would ride audio like any audio technician, lowering the volume slightly whenever it peaked above a certain level, and raising it slowly when incoming audio volumes seemed a bit low. This is what we have grown to know as automatic volume control. And just like the real audio person, the circuit can be "fooled". If you were to speak very softly for a minute or so, the automatic volume controls would rise bringing your voice to the right volume. The increased amplification would also increase the strength of ambiant sounds in the room such as echoes, hum, hiss, shuffling of paper, whatever. Then if you suddenly sneezed, the loud sound would knock the volume control down instantly (but not soon enough to save the listeners from a blast from their speakers), and now the volume would be so low that your voice would be barely audible. Slowly the automatic control would allow the volume to creep back up to normal.

Automatic gain controls are handy when you don't have enough hands. You simply switch them on and then go ahead to do other things. They don't do too bad a job; most home VCRs have automatic gain controls in them to adjust sound volume. A little more subtle, however, is the limiter. The limiter is a VGA that effects only high volumes. It's only job is to keep the surprise sneezes from blowing everyone's ears out. It also keeps an occasional verbal outburst from distorting badly. It doesn't crank down the entire volume control leaving you with semi-silence after an outburst; it just cuts off the peaks leaving the valleys and mid-mountains untouched and natural.

Noise gates work at the other end of the spectrum. When things are quiet, they turn the volume down more. This is great for reducing hiss and unwanted background sounds. When music is playing and people are speaking continuously, you can't hear these weak noises. When the music or narrator is silent, however, these sounds become audible to your ears. By turning down the volume during the silent moments, you don't miss any of your wanted sound, but the noise is reduced during the silent passages so that it cannot be heard during that special time when your ears are most sensitive to noise. Noise gates have to work quickly to turn themselves back up the instant there is
legitimate sound. Perhaps you have heard a noise gate that was over used. Someone would be talking and between every word there would be an uncanny silence, and as they spoke, you would hear the whoosh of some other background sound mixed with their words.

In short, noise gates make quiet sounds disappear. They also can accelerate the decay of reverberation changing how you perceive the size of a room. Used in reverse, they can increase the attack of sound making drums and musical instruments sound more percussive. Compressors compress the dynamic range of sound. If a sound were to double in loudness, a compressor could make it increase in volume by only 50% Weak sounds are hardly touched at all. Loud sounds and extremely loud sounds end up having nearly the same volume. Compressors make it possible for sounds that vary excessively in volume, like the voice of someone shouting, to remain within the range of what the mixer, recording media, radio transmitters, and loudspeakers will bear. A typical example of highly compressed sound is one of those annoying pitchmen on AM radio; he seems to be shouting in your ear, but the VU meters would show that his actual volume level rarely exceed 1 dB.

Compressors are good for keeping control of exuberant narrators and holding the lid on singers and musical instruments that range from a whisper to a blast. Compressors are also used on wireless microphones to assure that loud outbursts don't overdrive the transmitter circuits.

Back to narrators for a moment --- amateur narrators tend to speak in monotones --- very dull to listen to. Listen to some professional announcers and you'll notice how they constantly vary the pitch and volume of their speech to keep the message exciting. This kind of speech pattern pushes VU meters all over the scale, requiring constant attention from the audio person. Enter the compressor, squeezing the whispers and the screams into a 4 dB range. Now the announcer sounds excited and animated, but the audio levels are tame. Incidentally, some announcers can mimic animated conversation and keep within the 4 dB volume range without a compressor. But boy do they sound weird when they do this at the dinner table!

If compressors are like bondage for sound, expanders are the opposite, letting sounds leap to life. Unlike compressors, expanders will do nothing to a weak sound, but will take a loud sound and make it louder, adding more emphasis to it. Expanders often work in cooperation with compressors; compressed sound is a little unnatural, but is the price we pay for squeezing widely varying volumes through recording devices and transmitters that have limited range. Once we are past the recording devices and transmitters, and the sound is ready to be amplified and fed to a loudspeaker, an expander can decompress a sound returning it to its full dynamic range.

Describing this another way, say you had some sounds that ranged between 0 and 100 dB in signal strength. Say that your recorder distorts when it hears anything over 60 dB. You don't want to just turn down the volume of your recording, that would hurt the weak sounds too much. Instead, you use a compressor which hardly touches the low volume sounds between 0 and 40 dB. The incoming sounds between 40 and 80 dB get squashed down to fit in the range between 40 and 50 dB at the compressors output. The incoming sounds between 80 and 100 dB get squashed into the range between 50 and 55 dB. Thus, sounds that ranged from 0 to 100 dB going into the compressor came out between 0 and 55 dB, well within the range of a digitizer, recorder, wireless microphone transmitter, or whatever. We could stop there or we could attempt to restore the sound to its original dynamic range once we have passed our equipment bottlenecks. With an expander, the incoming barely-compressed sounds between 0 and 40 dB would remain untouched. The louder sounds between 40 and 50 dB would be amplified and come out in the range between 40 and 80 dB. The heavily compressed sounds between 50 and 55 dB would be expanded to the range 80 to 100 dB, thus reconstituting the full dynamic range of the original sounds.

Like equalizers, varible gain amplifiers are usually connected early in the sound chain, before signals are sent to reverbs, echo, delay, and other processors. Also like equalizers, VGAs have all of the signal pass through them; they don't sample a bit of the signal, change it, and then add it back to the sound stream.

Delay

A delay is a repeat of the original sound just milliseconds after the original sound. It generally adds thickness and richness to a sound, and can also be used to artificially construct a mental image of the room where the sound was generated. Imagine for a moment that a person were speaking to you while standing on a mile tall pedestal. Presume it's a calm day and you are up there with him. Ninety percent of the vibrations from his voice will go other places than towards your ear. The small amount of sound that strikes your ear will seem thin and strange. Anechoic (soundproof) rooms and "dead" (soundproof) audio studios also sound this way. Music and voice both sound unnatural because, in the real world, we always have a floor and walls around us reflecting delayed strains of the original sound back to our ears.

Now imagine someone with a tall crane hoisting a wall and positioning the wall behind the person speaking. If the wall is merely a foot behind the speaker, you will hear his original words plus a weaker delayed repeat of those words about 1 millisecond later. His voice will sound stronger and more realistic, but you won't quite know why. One millisecond is such a quick delay that your ear cannot tell that it's hearing the sound twice. If a wall is moved slowly away from the speaker, the delay becomes greater. Between 5 and 15 feet, the wall will continue to reinforce the person's voice making him sound natural and normal, but the sound will change character as the wall increases its distance. When the wall is 15 feet away from the speaker, the sound is delayed about 30 milliseconds and your ear begins to perceive it as a discernible delay or a repeat. Moving the wall further eventually turns the delay into your classic
echo...classic echo.

When a delay time is less than 5 ms, some of the delay's electrical vibrations cancel some of the original vibrations. Certain frequencies will be nullified while others remain (a condition called comb filtering). The result is a hollow, through-a-pipe sound, a bit like the voice of Darth Vader. Somewhere between the zone of cancellation and the zone of distinct delays is the desired reinforcement zone where the delays are constructive and thicken the sound. These are the delay amounts you will find most useful for normal audio sweetening.

In the real world, we generally have more than one wall, so there are several sound reflections picked up by the microphone and recorded. By combining several delays together, one can recreate the perception that a room of nearly any size exists. If the delays are all short, a voice can sound as though it were coming from within a vehicle, a hallway, or a small room. If the short delays are strong (ie. nearly as loud as the original sound), the sound will appear to be coming from a bathroom. Increase the delay time and the room becomes larger. Strengthen the delay volume and the walls become "harder" like in a gym or parking garage.

How can you use this in the video world? Say you made a recording of a person speaking in a room. If while editing you need to dub in new lines, you could bring that person back to that room and record them, thus maintaining the "room sound". If the room isn't available and you bring that person back to the studio or have someone
else record the person in another town, naturally the character of the sound will be different. You would try to use the same kind of microphone at the same distance from the person in order to match as many variables as possible, but often the subtle difference in room reverberation will make your inserted audio sound glaringly different
from the rest of the recording. Here is where you can run the signal through a delay (or several) and attempt to sonically recreate the missing walls.

For a more creative example, we are always trying to make sets seem like the real thing on the TV screen. Sound carries some of the subconscious message to the listener, so if you can make your tiny set sound like a big room, or the metal interior of a submarine, you can transfer the audience to a place that never existed. Some inexpensive cardboard and paint and a few delayed sounds can become a convincing spaceship or foxhole without the cost or cramped quarters.

When working with music, delays are useful for creating the impression that there are more than one instrument. This trick is a flimsy one, however. In the real world no two instruments play exactly the same note at the same frequency in the same phase all the time. One voice cannot be simply duplicated into a chorus of singers.

In the real world, some voices will vary above and below pitch slightly. To more closely approximate the real world, there's a button found on delays and music synthesizers called chorus. The chorus is a swept delay which is a combination of one short fixed delay and one changing delay. The changing delay is like having a wall coming toward you, and then moving away from you. Electronically the swept delay is being adjusted from a small amount of delay to a larger amount and then back to the smaller amount. The frequency varies a little above and below the normal frequency. Other adjustments on your processor allow you to vary the modulation (rate of sweep) of the swept delay creating a slow waaahhhhaa sound or maybe a quick fluttering wowowowo sound. If you listened to it, it would sound a little like vibrato in a singer's voice, or the sound of a train whistle as it goes by you. To make the chorus sound even more natural, the sweep rate is varied, perhaps making a woowowoooowowwo sound. The random or fake-random sweep is similar to what you hear when two voices are singing the same note.

Reverberation

Reverberation is the reflection of sound in a defined space. It is the sound you hear when singing in the bathroom, in a stairwell, or in a cavern. The word is used interchangeably with echo which is technically something different. Echo means decaying repeats of a sound....sound...sound. Echo is what you hear when you scream "hello" to a canyon wall. Echo makes an interesting sound effect, but will turn a voice into cacophony. Guitarists can use echo to turn one pluck of the string into many, adding complexity to the music. For the most part, echo is simply used as an effect, and reverberation is the preferred flavoring for sound. Incidentally, many mixers have controls on them called echo send. This is a mixer circuit that takes the microphone's preamplified signal, and sends it out of the mixer for further processing (by an echo, delay, reverb, or whatever), then takes the result back into the mixer and recombines it with the original sound. You would vary the amount of the two signals with a knob on the "echo send" part of the mixer.

Reverberation used to be made with mechanical devices such as metal plates and steel springs. Audio signals would be transduced into physical vibrations causing the spring to bounce around or the plate to vibrate. At the other end of the spring or on a parallel
plate would be another transducer that changed the mechanical vibrations into an audio signal again. Today most reverbs are digital audio devices which sample a sound, convert it into numbers, manipulate (perhaps repeating) the numbers, then convert the data back to sound. Reverbs are often souped-up delays that feed their own delayed sounds back into themselves in complex manners. Where a delay represents one repeat of a sound (like from that wall behind the guy on the pedestal), reverb is many repeats blended together (as you'd expect from a real multi-walled room).

Reverberation creates a room ambiance, much like delay does. You can think of delay as a couple walls (usually the closest and most important ones) and think of reverb as the entire room. There would be a direct reflection from the nearest wall and perhaps the floor, followed by weaker more numerous reflections from the rear wall of the room, ceiling, and other places. Reverberation forms mental space so that you can tell whether something was recorded in a living room or a gymnasium. Reverb adds natural depth and excitement to music and plays an important role in gluing together independent sounds that have been mixed together from separate recordings. You could feed many microphones into a mixer or have many tracks in an audio recording, and each of them may sound singular and independent even though they are mixed together. By using reverberation, the sounds blend more naturally.