BACKGROUND Once upon a time there were only two signal processors for PA, spring reverb and tape echo. Both suffered from mechanical frailty and the echoes had a tendancy to overload with amazing ease. This remained the case throughout the nineteen-sixties, up until that fateful summer of 1969 and the Woodstock festival, in the wake of which PA technology took a huge, faltering step forward. Suddenly 4 and 5-channel PA "heads" were being replaced by multi-channel consoles and separate power amplifiers. Speaker "columns" and "cubes" began giving way to jumbo bins with horns stacked ontop. The wedge-shaped stage monitor was born along with the dreaded term "monitor feedback" (Which is it - the main stacks or the monitors?!).
Signal processors now included passive crossovers and crude graphic equalizers, neither of which were well understood. In the mid-1970's digital delays made their entrance along with 31-band EQ's, active crossovers and bi-amping. Power amps became stereo then they got bigger and then people learned about "bridging" them. The smoked drivers, the blown amps, the cursing - it was truly something to behold.
But the dust finally began to settle in the mid-1980's when bin/horn stacks started giving way to advanced all-in-one speaker systems with smooth frequency response and superior passive crossovers. This reduced the need for bi-amping and its inherent dangers for the inexperienced. Meanwhile, previously astronomical prices on good quality compressor/limiters had come down and continued to do so. Now they're available to club-size PA users. And the list of PA signal processors continues to grow. This writing will cover the major ones used for PA - guitar effects will not be covered.
GENERAL Signal processors all have two things in common:
GRAPHIC EQ Try to imagine a set of "volume" controls wherein each one only affects a certain segment of the overall 20Hz to 20,000Hz range of sound frequencies. That' s a graphic equalizer - more or less. Actually the faders can't turn things up to "10" or down to "0", but they can turn them up to about "7" and down to about "4", far enough to have a marked effect on the overall volume level, depending on how many are pushed up or down from centre, and how far. It's hard to imagine that merely changing the comparitive volume levels of various frequency bands could make such huge changes in the way things sound, but that's exactly what happens.
The actual tone which identifies something, let's say a trumpet note, depends on the comparitive volume levels at which the harmonics of that note occur. Harmonics are incidental vibrations which occur whenever an instrument or voice is producing a note. In this case, the trumpeter's lips can be vibrating at 440Hz to produce middle A, but ripples in the skin also produce tiny notes at higher frequncies, these are the harmonics. The same sort of thing happens along a vibrating string or a reed or anything which vibrates to produce sound . When you adjust an equalizer you increase or decrease the amount of audible emphasis on any harmonics occurring in that range, hence altering the tone.
Feedback is simply a sound that's gone wild. What's at work here is a phenomenon called sympathetic vibration. Basically, two or more things which are physically prone to vibrating very freely at a certain frequency can get each other in motion by "remote control". One of them sets the air in motion and the vibrating air then sets the other thing(s) in motion.
{ TIP - When connecting the EQ directly to an input channel via the Insert jack, remember that on some if not most mixers, the Insert jack is post-Gain. As a result the "return" part of it is not regulated by the channel's input Gain control. Thus, if you increase the EQ's Gain by too much and distort the channel, you should reduce the EQ's Gain, not the channel Gain.}
Below 50Hz you enter the realm of very long wave forms and their ability to set all kinds of things in motion - bass drums, bass strings, microphones, speakers, furniture, walls, ceilings, you name it. This may cause feedback problems and sound colouration. Additionally, woofers reproducing these very low frequencies at high sound-pressure levels consume huge amounts of amplifier power which leaves less for the important 60-100Hz range where audiences get treated to the "thumps" they like so much.
{ TIP - In a live sound situation, it's generally a good idea to activate both filters. They cost your audience no real sonic performance and they offer a valuable insurance policy.}
PARAMETRIC EQ Unlike the graphic EQ, the parametric most often has rotary controls rather than sliders. You will find controls for Frequency and Q accompanying each cut/boost control so that you end up with three knobs for each frequency. This provides you with the ability to pinpoint a range of frequencies and either boost or cut the gain there.
The Frequency control is self-explanatory - it moves the cut/boost across the audio spectrum. The Q control enables you to broaden or narrow the band of frequencies to be cut or boosted. As a rule, the cut/boost will have a less noticeable effect on a high Q setting because Q reflects the ratio of gain change to frequency band, ergo a high setting represents a high gain-change-per-octave ratio, hence a narrow band of frequencies. Together the three controls create a potentially ideal cure for problems like feedback. You set the cut/boost for a -6dB cut, set the Q up about half way, then rotate the frequency control until the feedback stops.
The Q control is especially valuable in this regard because once you have located a feedback frequency range using a medium Q setting, you can increase the Q setting thereby narrowing the range of cut frequencies so that fewer "innocent" frequencies are affected and the basic sound quality does not suffer needlessly. Like the graphics, parametric EQ's may feature high and low-pass filters. See above for details.
Controls usually include:
Perhaps more common is the need to ensure that speakers do not receive more than their maximum rated power. A limiter is the best means by far of providing this insurance. Fuses, for instance, will always fail after a certain amount of usage because amplifier output signals cause the elements to flex rather like an inchworm, causing the metal to fatigue then break (fuses also add current-variable resistance to the speaker circuit which affects the amp's damping factor). A good, sonically transparent limiter can be set so that the maximum mixer signal the amplifier receives is only sufficient to drive it to the pre-determined output power level.
A watt meter attached to the output of the amp can be used to measure power while test signals are put through the system. With the comp/limiter's Output Level on full, the Ratio on 4:1, Attack and Release at low settings, establish the right Threshold setting so that the desired power level is not exceeded according to the watt meter. Now "peg" that Threshold setting with a dot marker. It is worth noting that a setup lke this to cover monitors as well as main speakers has the added benefit of protecting against accidental transients from mics being dropped, cables being yanked out, turn-on "snaps" when the mixer goes off then comes back on (somebody tripped over the power cord) or A.C. surges when the lighting system dims right down.
A few other applications for comp/limiters and settings for them are listed below (from the ART model CS-2 manual); Note: "# o'clock" simply refers to the general control setting
Crossovers do this by putting the audio signal through a series of filter stages which separate the lows from the mids and/or highs. What makes Active crossovers preferable to the passive variety is their lack of phase alignment problems, their frequency variability and their ability to control the volume level of the different frequency ranges.
Basic features include the following;
For example, if you were to set the high/low frequency of a 2-way, 12dB-per-octave crossover at, let's say 1,000Hz (1,000 cycles-per-second), the high-frequency output signal level from the crossover will be attentuated by -12dB one octave lower than that. As a result, power signals from the amp to the horn or tweeter occurring down at 500 Hz, i.e. one octave below 1,000Hz, will also be attenuated by -12dB or roughly 75%. Thus, if your horn or tweeter is rated at 100 watts at 1kHz. and a 100-watt power amp is connected to a 12dB/octave crossover set at 1kHz., the component may be receiving output down at 500Hz at a power level of 25 watts - not likely to cause a problem in this case. But if the amp puts out 200 watts, the HF components could be receiving 50 watts down at 500Hz which is dangerous.
Keep in mind that low frequencies kill horns and tweeters even at low applied power levels. Hence, when you are bi-amping high-frequency components in a 2-way system, do not over power them and be sure about the crossover frequency.
During setup when there are no audience members present, plug the pink noise into a mixer channel with all EQ set flat including the main graphic - the system will be roaring like Niagara Falls. Now set up the mic in the middle of the audience area and adjust the crossover Level controls until the RTA 's response graph display looks as flat as possible. Now you can adjust the main EQ slightly to reduce the size of response peaks which look big enough to pose a feedback threat; just remember the rule, never over-equalize, not even to make an RTA look flat.
Although the RTA provides a handy visual guide, it should not be taken too literally. You still need to use common sense and your ears to a certain degree; but remember that this is a sound job, not your living room - take it easy with the "boom and sizzle" or the system will start to sound muddy when you increase the level later on. And if you are in doubt, the safest thing to do is to set all level controls on the crossover at maximum and, in so doing, send equal amounts of signal to all power amps. The maximum settings will compensate for potentiometer tolerances - max is always max no matter what the tolerances might be}.
Aside from that, it's worthwhile mentioning that your reverb unit would normally be connected to the mixer's EFX Send and Return jacks - Send going to the reverb's Input and the reverb's Output going to Return. However you may only want a particular type of reverb or echo on one channel, possibly the lead vocalist. Then you would patch the reverb unit into that channel's Insert connection (See >>"INSERT" under MIXER INPUTS). Note that this channel should probably not be put through any additional reverb as the clarity of that channel could be muddled. In general terms, reverb can add a nice "dimension" to your sound and even a bit more if it's a stereo reverb and you're running a stereo main PA. Just don't overdo it.
