Crossovers

Basically, crossovers are little electrical devices that receive a fullrange signal and divide it into separate outputs of midrange frequencies, lowrange frequencies, and highrange frequencies. That way, the highs are sent only to the speakers designed for the highs, the lows are sent only to the speakers designed for the lows, and the mids are sent only to the speakers designed for the mids. Passive crossovers do this by dividing the signal after it leaves the power amp while active crossovers do this by dividing the signal before it gets to the power amps.

       Passive crossovers (located inside full range speaker cabinets) are good in that they make it possible to provide a full range of sound using only one amplifier, but they are a little inefficient. Since all the speakers are working from the same source, the low speakers, which require more power, will tend to rob power from the higher frequency speakers and horns.

       Active crossovers (plugged in-line before the amplifiers) are good in that they make it possible to power the mids, lows, and highs from different amplifiers. This way you can use a big super-duper amplifier for your lows, and use a smaller amp for the mids and highs. This is a much more efficient use of power, and it gives you the ability to acquire a much more powerful and full sound. The catch is that using an active crossover requires a lot more equipment and expense.

       Using an active crossover in a system is sometimes called bi-amping or tri-amping. Below is an example of tri-amping the mains in a mono system.

Tri-Amping the Mains

       The effects loop and the monitor system should be connected in the same way as before, but now the mains have been turned into a tri-amped system. To do this, a crossover, three separate amplifiers, and three separate sets of speaker cabinets must be used. Each of these amplifiers as well as each set of speaker cabinets must be designated to a specific audio frequency. Which frequency goes where is determined by the outputs on the crossover. The "low out" should go to the input of the amp with the highest wattage because the low end speakers will require the most power, and the "high out" should go to the amp with the lowest wattage because the horns will need the least power. Just remember that it takes a lot more energy to vibrate the great big cone on a fifteen or eighteen inch speaker than it does to move the tiny diaphragm in a midrange horn.

Warning: Never plug a high end speaker or horn into the amp that is plugged into the "low out" of the crossover. These speakers are not designed to handle such low frequencies and will be damaged very quickly if hooked up incorrectly.

To hook up the system in example , follow steps 1 through 15:

Monitors and Effects

1. Connect these together as described in example.

Mains (Keep in mind that even though the signal flow splits inside the crossover, it still flows from the mic toward the speakers)

2. Plug a high impedance cord into the main "output" of the mixer.
3. Plug the other end of this cord into the "input" of the main equalizer.
4. Plug another high Z cord into the "output" of the equalizer.
5. Plug the other end of this cord into the "input" of the crossover.

Lows

6. Plug a high Z cord into the "low output"of the crossover.
7. Plug the other end of this cord into the "input" of the highest powered amp.
8. Plug a speaker cord from each "speaker out" of this amp into the "input" of each low speaker (one cord to each speaker).

Mids

9. Plug a high Z cord into the "mid output"of the crossover.
10. Plug the other end of this cord into the "input" of the middle powered amplifier.
11. Plug a speaker cord from each "speaker out" of this amp into the "input" of each midrange speaker (one cord to each speaker).

Highs

12. Plug a high Z cord into the "high output"of the crossover.
13. Plug the other end of this cord into the "input" of the least powerful amplifier.
14. Plug a speaker cord from each "speaker out" of this amp into the "input" of each Midrange/high horn (one cord to each horn).
15. Take a break. That was a lot of work.

       To see how a large super-duper stereo PA system with multiple effects loops, monitors, and compressors is hooked up, simply click on "next page". Be patient though. The diagram is a little bit on the large side and will take a little time to load up.

Source (thefxcode.com) 

Small Practical PA

Small Practical PA

       Here we have added monitors and effects. The monitors are the speakers that face back toward the stage so that the people there can hear themselves singing. They require a separate equalizer and amplifier and are hooked up in the same configuration as the mains (mic to mixer to EQ to amp to speaker), except that the cord running to the input on the monitor EQ is coming from the "Monitor out" channel on the soundboard rather than the "Main out". Basically what has happened is that the signal that has left the microphone has been split by the internal electronics of the soundboard into two separate signals. One signal is then routed through the "monitor out" into the monitors while the other is routed through the "main out" into the mains. This makes it possible to adjust the sound coming out of the monitor speakers separately from the sound coming out of the main speakers.

Note:        If you are using a stereo equalizer, you can run the mains through one channel of the EQ (say channel "A") and the monitors through the other (channel "B"). Just remember which channel you assigned to the mains and which you assigned to the monitors.. That way you know which sliders and knobs adjust each part of the system. Doing this avoids the need to purchase a separate EQ, and allows you to fully utilize the equipment you already have. This same principle can be applied to any stereo components in the system such as amplifiers, compressors, and crossovers, etc.....

       The effects help to thicken out or modify the sound that is going through the PA system. There are many different kinds of effects including such things as delay (echo), reverb, and chorus. These can be hooked up "in line" or directly in the path of the signal, but are much more versatile when hooked up in an "effects loop". An effects loop is created when a signal is sent out of the soundboard into and through whatever effects you are using and then "loops" back into the soundboard. Once this loop is set up properly, the effects can then be adjusted individually for each input channel (microphone, keyboard, etc...) on the soundboard. This means you could put a lot of echo on one guys vocals while adjusting another guys to have almost none. On the other hand, a digital delay (echo effect) in line with the mains....say, between the mixer and the EQ....would affect everything coming out of the main speakers equally, and everyone would have the same amount of echo.

Now that we have examined the concept, here's how you plug it all in:

Mains

1. Connect everything together as described in example 2. This is essentially the signal path for the mains.

Monitors (Remember, signal path flows from the microphone toward the speaker.)

2. Plug a high Z cable (patch cable) into the "Monitor out" of the mixer.
3. Plug the other end of this cord into the "input" of the monitor equalizer.
4. Plug one end of a high Z cord into the "output" of the monitor equalizer.
5. Plug the other end of this cord into the "input" of the monitor power amp.
6. Plug two speaker cords into two speaker "outputs" on the monitor power amp.
7. Plug the other ends of these cords into the "inputs" of the monitor speakers.

Effects loop

8. Plug a patch cord (usually high Z) into the "effects send" or "effects out" of the soundboard. This is where the signal leaves the mixer (you are sending it out of the mixer).
9. Plug the other end of the cord into the "input" of the effects unit.
10. Plug another High Z cord into the "output" of the effects unit.
11. Plug the other end of this cord into the "effects return" or "effects in" on the soundboard. This is where the signal returns to the mixer thus completing the effects "loop".

Multiple effects

       You will probably want to run several different effects at the same time. This can be done by either using a multiple effects unit that will run many effects simultaneously within a single unit, or by putting several different effects units in line within the same effects loop. "In line" simply means hooking them up in a row such as in the following example.

       In this case the signal flow is coming from effects send and flowing toward return. Remembering that the inputs are always on the upstream side of the flow, the "inputs" in this situation will always be coming from the effects send jack, and the "outputs" will always be going toward the effects return jack. Thus we have a signal path like this:

Effects send (from the board) to Input (on the delay) to Output (on the delay unit) to Input (on the Reverb unit) to Output (on the reverb unit) to Effects return (on the board)

       Some soundboards are equipped with more than one effects channel. All you need to remember is that if you used the effects send from channel "A", you have to use the effects return for channel "A". If your board has more than one effects channel, you can set them up totally independent of each other. This gives even more control when adjusting the sound to the individual mixer inputs (microphones, etc...).

- Source (thefxcode.com)

Noise Gates

A Guide to Noise Gates in PA Systems

A Noise Gate is to an expander much as a limiter is to a compressor. Essentially it is an expander that mutes the signal when it falls below the threshold, rather than simply reducing the gain. A downward expander with a ratio higher than 8:1 is effectively acting as a noise gate.

What it is

Usually a noise gate is a 19" 1U box with knobs or buttons (and more often than not a couple of LED meters) on the front. Some compressors have a noise gate function included in the channel facilities.

What it does

A noise gate mutes the signal when it falls below a threshold (the threshold is generally user-determined).

How it works

It senses the input level, and closes the channel when the input level falls below the threshold. It can have controls for:

• Threshold.The level below which the signal is muted.

• Attack. How quickly the gate opens once the signal reaches the threshold. On most sounds (particularly percussive sounds), the attack should be very fast to avoid cutting off the beginning of the sound.

• Release. Certain important parts of the sound (decaying resonance & natural reverb tails) may fall below the threshold. If the gate is closed too soon, the cut off may be audible (& unnatural).

• Filter. If a noise gate is used on a small rack-mounted tom, low-frequency sound from the kick drum may exceed any usable threshold. A filter enables the user to tune out (or sometimes tune in) a frequency range, so that the gate only opens when sounds in a particular frequency range exceed the threshold.

• Stereo/Mono. Most multi-channel noise gates allow coupling of each pair of channels for stereo use. If gating is applied to left and right channels independently, drop-out of one or other side can result whenever the signal on one side is above and on the other side below the threshold.

How do you use it?

If all else fails, read the manual!

Noise gates are usually connected on channel inserts.

Expanders and noise gates can have a disastrous effect if set up badly, and are at best problematic in live performances: spill from other instruments or from monitors can make it impossible to set effective thresholds. With care they can deal with very situation-specific problems (for example, buzzing from a bass guitar can be silenced between songs by careful use of a noise gate with a filter). However, they are most commonly used on drums, either to reduce spill between drums, or to reduce the ringing of undamped toms. In this application, they are usually used on channel inserts.

To use a noise gate on a drum:

1. Set the channel gain.

2. Set the attack and release to their fastest settings.

3. While the drummer hits that drum (and only that drum), raise the threshold until the drum starts to cut off.

4. Reduce the threshold a couple of decibels, so that the beat always exceeds it.

5. Increase the release time gradually, so that the drum is able to ring on a little after each beat.

Although the cut-off will still be audible, this will be much less noticeable when the whole kit is being used and the whole band is playing. A little reverb on the drum will help make the cut-off less noticeable during solos.

Other than on drums, leave well alone.

Do you need one?

If the drums are well tuned and damped (or if ringing toms are an intentional part of the drummer's sound), you probably don't need one. Otherwise, one channel for the kick drum and one for each tom can be useful. Most compressors include a basic noise gate on each channel, which is useful if you need to gate and compress the same signal (kick drum is sometimes a good candidate for this).

What sort do you need?

One that has at least all the controls described above. Four channels is usually enough (which might mean you need more than one unit).

-  Source (astralsound.com)

Dynamics Processors : Compressors / limiters

2.1. Introduction

    The aim of a compressor is to reduce the level of the loudest signals. Typical reasons for compressing are:

• Controlling the energy of a signal. The human ear detects energy changes on signals. We can express the energy of a signal mathematically as its RMS value (roughly  its average value excluding the sign). The human ear is very sensitive to energy variations, so changes should always be smooth and subtle so as not to be evident to the ear. Alternatively, abrupt or excessive compression maybe used as an effect, though this is normally used for recording applications and not for live sound.
  Thus, we could keep a singer's voice under control, compensating for higher levels at the microphone due to shouting or getting too close to the mic, and therefore making the voice's levels more even.
• Controlling the peak levels of a signal. Very often, our equipment is limited by its peak signal capacity. Amplifiers in different parts of a mixer's signal path may saturate. A power amplifier may clip. Loudspeakers maybe in danger of getting damaged by excessive excursion. In these cases, we are concerned about controlling the peak levels of signals, such that the needed processing tends to be some form of limiting rather than compression.
• Reduce the dynamic range on a signal. If we attenuate the peaks out of signal, we are reducing its dynamic range. Since many devices are peak limited (power amplifiers, recorders), this allows us to increase the RMS level of the signal.

    Other than compressing RMS or peak levels, the detection circuit may also be RMS or peak based. Some compressors provide the ability to select between compressing based on the detection of average (RMS, the most common option) or instantaneous (peak) levels. The way to detect RMS levels may also vary: higher quality compressors detect real RMS, while cheaper ones only approximate it.

    Which brings us to defining what a limiter is. A limiter is really just a form of compressor. We could say that compressing is smooth attenuation, whereas limiting is doing it in an abrupt manner. Often we will come across compressors that feature dedicated limiters, thus offering simultaneous compression and limiting from a single unit. Typically, the term limiter is also associated to faster times, particularly for attack, so as to avoid exceeding a specific signal maximum at all times. Standard compressors will normally have a range of ratio values that allow performing both compression and limiting, which is the reason why they tend to be referred to as compressor/limiters.

2.2. Controls

Compression is a difficult task that may require very different characteristics depending of the type of signal. Numerous controls are therefore needed. The drawing below shows a compressor with the most common controls.

The most common controls provided on compressors are given below. You may not always find all of them, or you may get additional ones:

• Threshold. When this level is exceeded, the processor starts compressing (i.e., attenuating, reducing volume).
  The illustration below shows resulting levels (in dBs) of a signal being compressed with a higher and a lower threshold level. In the first example, the third signal peak passes through unaltered.

• Attack time. It's the time it takes for the signal to get fully compressed after exceeding the threshold level. Minimum attack times may oscillate between 50 and 500 us (microseconds) depending on the type and brand of unit, while maximum times are in the range from 20 to 100 ms (milliseconds). Sometimes these times are not available as times, but rather as slopes in dB per second. Fast times may create distortion, since they modify the waveform of low frequencies, which are slower. For instance, one cycle at 100 Hz lasts 10 ms, so that a 1 ms attack time has the time to alter the waveform, thereby generating distortion.
  Specially for mastering and FM radio broadcast applications, where low dynamics are desired, there exist multiband compressors (also know as split-band compressors) that divide the spectrum into several frequency bands which are compressed separately with different compression times (faster for high frequencies, slower for low frequencies), and summed again into a single signal. This minimizes compression induced distortion while achieving very high compression, and avoids dulling of the sound, a compression side effect that will be explained later.
  In limiter applications where we want to avoid speaker damage, the longer the attack time, the higher the risk of damaging the equipment. However, too fast an attack time will generate distortion... we start to see the difficulties of selecting the correct times.
  
• Release time. It's the opposite of attack time, that is, the time it takes for the signal to go from the processed (attenuated) state back to the original signal. Release times are much longer than attack times, and range from 40-60 ms to 2-5 seconds, depending on the unit. Sometimes, these times are not available as times, bur rather as slopes in dB per second. In general, the release time has to be the shortest possible time that does not produce a "pumping" effect, caused by cyclic activation and deactivation of compression. These cycles make the dominant signal (normally the bass drum and bass guitar) also modulate the noise floor, producing a "breathing" effect.

Although it is not commonplace on compressors (it is on gates), some models may provide a hold time control. This can be useful to avoid low frequency distortion when fast release times are needed, by setting the hold time to a time longer than a cycle of the lowest frequency. For instance, 50 ms for 20 Hz. That way the compressor waits for a cycle to be completed, thereby avoiding distortion of the shape of the waveform.

• Compression ratio. This parameter specifies the amount of compression (attenuation) that is applied to the signal. It normally ranges between 1:1 (which is read "one to one", and represents unity gain, i.e., no attenuation at all) and 40:1 (forty to one). The ratios are expressed in decibels, so that a ratio of, for instance, 6:1, means that a signal exceeding the threshold by 6 dB will be attenuated down to 1 dB above the threshold, while a signal exceeding the threshold by 18 dB will be attenuated down to 3 dB above it. Likewise, a 3:1 (three to one) ratio means that a signal exceeding the threshold by 3 dB will be attenuated down to 1 dB. With a 20:1 ratio and above the compressor is considered to work as a limiter, though a theoretical limiter would have a compression ratio of infinity to one (whatever the input level, it would always be attenuated down to the threshold level, so that output would never exceed the threshold once the attack time has elapsed). We could say that a ratio of around de 3:1 is moderate compression, 5:1 medium compression and 8:1 strong compression, while over 20:1 (or 10:1, depending on who you ask) would be limiting.
  The illustration below shows original and compressed signal levels for ratios ranging from moderate to maximum compression (limiting). The ratios, from left to right, are 3:1, 1.5:1 and infinity:1 (note the slight overshoot as it takes a finite attack time to clamp the signal down to the threshold level).

In a way, compression ratio and threshold are related, since both increasing the ratio and lowering the thershold will result in more compression being applied to the signal.

A more scientific way to show compression is through input versus output diagrams. We will find this type of graph in the user's manual of our unit. The 45 degree straight line represents the absence of dynamics processing, i.e., like a (loss less) cable. Above the threshold (which we have arbitrarily set to 0 dB), the 45 degree line deviates and forms another straight line with a slope that is lower the higher the compression ratio is. The line for the infinity:1 ratio shows a zero slope, since we are forcing the output signal to never exceed the threshold level, no matter what the input level is.
NOTE : If you find the graphs difficult to understand, look for an input level (horizontal axis) and follow it upwards in a straight line until you meet one of the compression lines. Take that point all the way to the left in a straight line to the output levels (vertical axis) and check that the level is lower. The example dotted gray line in the graph shows how a +10 dB input level becomes +5 dB a the output for a 2:1 compression ratio.

 

• Knee. On compressors that have it, it is a control that allows the selection of the transition between the processed and unprocessed states. Typically one would get an option between a "soft knee" and a "hard knee". Sometimes the control allows the selection of any intermediate position between the two types of knee . Sometimes soft knee compression is referrer to as "OverEasy" (can't start to even figure why, i do not see a connection to eggs over easy), as used by DBX branded compressors. The soft knee allows for a smoother and more gradual compression.

• Stereo link. In general, when dynamics processors are used to process a stereo signal, we need to be able to link the processing on both channels such that it takes place on both channels at the same time. Otherwise the imaging will be confusing as it will change from the center to one of the sides or the other. Monophonic compressors often feature a link connection to be able to send a cable to another unit and synchronize the compression action.
   
• Output gain. Since compression introduces attenuation, this can be compensated by raising the output volume and in fact this control is often referred to as "makeup gain", as it makes up for the compression-induced attenuation. Or, given that a compressor reduces the dynamics or a signal, we can raise the output gain to make use of all the available headroom of the equipment to which the compressor is connected, though that would also mean raising the background noise that was present in the signal. To avoid the latter, compressors are often utilized in combination with noise gates, which may also come built into the compressors themselves.
   
• Automatic mode. It has become increasingly common to control some the compressor's parameters (typically attack and release times) automatically based on the signal's characteristics. This control enables that working mode. In general, automatic compression works well when one is looking for subtle compression, while the manual mode would be used for special effects.
   
• Side Chain Listen. Compressor that feature a Side Chain function (explained later) often provide a switch that routes of the side chain signal to the output of the compressor, which permits listening to it, which helps troubleshooting and setting the compressor.
   
• Bypass. Allows comparing compressed and uncompressed signals.

2.3. Meters

    Typically, compressors would feature at least some form of attenuation (compression) meter, which is normally implemented as a row of LED indicators. It informs the operator of how much attenuation in being applied so that he or she can evaluate whether the signal is correctly compressed or not (it could be over compressed or under compressed). The meter should show 0 dB (i.e., no compression) at some point when the signal is present, otherwise some of the compression is just continuous gain reduction that is best achieved with a volume control.

2.4. Side Chain

    Normally, the detection circuit uses a copy of the signal being compressed to check whether it exceeds the threshold level or not. However, many compressors allow using an external signal that is feed to the detector via the Side Chain (sometimes also called key) input. That way it is the external signal that triggers the compression, though it is the main signal that gets compressed. There may be a switch that toggles the detection signal between the main and the side chain signal, or sometimes, if the side chain input uses a 1/4" connector (often wrongly referred to as jack in many non-English speaking countries!), it is the connector that enables the function when the 1/4" plug is inserted. This 1/4" connector is an insert type connector that carries both a send and a return signal, the send carrying a copy of the main signal to facilitate its connection to a processor (e.g., an equalizer) and then feeding it back to the detector through the return part of the side chain connector.

    The most common use for this is using an equalizer for the side chain, So much so that some compressors already provide EQ facilities for the detector so that an external equalizer is not needed. For instance, we could reduce the high frequencies on the signal feeding the detector to avoid cymbals triggering the compressor. Or boost the sibilance frequencies to compress them on the main signal, a process which is referred to as "de-essing".

2.5. Setting a compressor depending on the application

    First of all, we need to decide whether we need compressing at all in the first place. Commercially available recordings are already compressed, so that it is seldom necessary to add further compression. In sound reinforcement applications, it is not common to use compression in a creative way to achieve specific effects, since it is the musicians that are responsible for their own sound character through effects units or amplifier combos. One must also bear in mind that compressing allows for increased average energy to reach amplifiers and loudspeakers, which could also increase the possibility of acoustic feedback, since a kind of sustain effect is generated.

    Before using a compressor, we need to connect it in the right place. If we use it in combination with a mixer, we will connect it to an insert point. The insert outputs are always pre-fader, which means we do not have to change the compressor's threshold every time the fader position is changed. Since attenuation of the higher volume signals produces a kind of sustain effect, compression may worsen some situations where feedback is a problem. On the other hand, if we apply compression to reduce the dynamic range and then add an amount of gain such that peak levels of compressed and uncompressed signals are the same, we are raising the average energy of the signal that gets to the amplifiers and speakers, which may be useful if we are short of equipment for the application, though it can potentially create thermal failure on the speakers (i.e., we may burn a voice coil) or trigger the thermal protection of the amplifiers (particularly if we are driving low impedance loads), which will mute to protect the amplifier. If we have an oversized system for the application, it's not a bad a idea to keep compression to a minimum on the instruments to a minimum and thus preserve their natural dynamics.

    Another side effect of compression is dulling of the sound, which is perceived as having less high frequency content. The reason for this is as follows. The frequency content of music has a lot more energy on the low frequencies than on the high frequencies. Which is why VUmeters move following bass drum and bass guitar. When a bass drum is compressed in the context of a full mix, we are also compressing the cymbal hits that may happen at the same time and which is a lot lower in level. The result of that is the aforementioned dulling of the sound. This effect can be minimized with slower attack times that let the percussive transients through. Some degree of high frequency boost is also often applied to counteract the dulling effect. Alternatively, some compressors automatically boost the high frequencies automatically during compression phases to avoid dulling.

        If we are looking to limit the output signal to a set level to protect a piece of equipment or avoid distortion, we will use a compressor (acting as a limiter in this case) just before the device (such as an amplifier or recorder). For instance, between the master mixer output and the amplifier. If the amplifier already features a built-in limiter that works as a function of the amplifier clip, it's probably best not to use a compressor and let the amplifier do it. If the speaker system is active and there is an active crossover with independent limiters per band, it would be advised to use these, as their attack and release times would normally be adequate for the frequency band being reproduced (quicker for high frequencies, slower for bass). I like clean sound system with some headroom to spare, so i would only occasionally active the limiter as a form of protection.

    In general, the criteria in this article are given as overall guidelines and starting points, but they will depend on the specific compressor model and they may have to be fiddled with by ear.

Limiters
For the compressor to work as a limiter, we will adjust the compression ratio to 20:1. Unlike compression, limiting is utilized as a brick wall that avoids signal peaks causing damage to speakers or overloading amplifiers (or recording devices), so limiters should only activate occasionally. Otherwise the effect will be very audible and sound quality will suffer. Attack times need to be fast to ovoid overload or over-excursion (on the speaker). Since there is always some degree of limiter overshoot (the limiter takes a finite time to provide full limiting, so some transient peaks may escape the limiting action), the threshold level may have to be set 2 or 3 dB lower than the level we do not want to exceed, so as to allow for some time for the limiter to be able to clamp the signal down.

Depending on the speed of a limiter's attack time, some limiters may distort the signal, working as abrupt wave form clippers. As mentioned earlier, some compressors are equipped with dedicated peak limiters. If so, we will make use of then as they are specifically designed for the job.

A specific type of limiter is the one that may be integrated into a power amplifier's channel to prevent continuous clip. If they are correctly designed, the compression (limiting) threshold is not fixed, and compression is only activated when the amplifier channel is actually clipping. The output voltage at which the amplifier clips may vary as a function of the type of signal and the mains power supply voltage, so the limiter would use a "floating" threshold to get the limiter to track the amplifier clip, avoiding unnecessary limiting when the amplifier is not clipping, or avoid the amplifier clipping when the mains voltage is lower than nominal AC power levels. In the case of the limiters in a crossover or controller, ideally they receive a "sense" signal from the amplifier to determine whether the amplifier for a given band is clipping or not, though the additional cabling makes it somewhat cumbersome for live sound applications. It the crossover unit is taking care of the limiting, in practice we have a multiband compressor and, if compression attack and release times are user selectable, we will need to chose faster timer for the high frequencies and slower ones for the low frequencies, thus optimizing the compromise between protection and audibility.

Ducking
Ducking refers to reducing (like a duck lowers its head) the level of a signal when another signal is being played. The standard example would be that of music being lowered when a DJ or presenter starts to talk. To achieve it we would use a copy of the presenter's voice fed into the detector circuit via the side chain (key) input.
Ringing out a system
A compressor can be used to aid setting up a system when it is being ringed out, i.e. its main feedback frequencies are being removed with an equalizer or a feedback elimination type unit. The compressor will have a low threshold level and infinity-to-1 ratio with hard knee characteristics. With no signal present, we will gradually increase the volume until the first feedback frequency rings. The compressor will catch it and keep it at a constant safe level, making adjusting the equalization an easier task. The process will typically be repeated until the third or fourth feedback frequency has been ringed out.
De-essing (compressing sibilance)
Certain singers exhibit excessive essing, which causes obvious sibilance. The side chain can be used to feed the detector with a signal that has the sibilance frequencies boosted such that the compressor is most sensitive to them. An equalizer is inserted in the side chain that would apply about 10 dBs to the 3.5-8 kHz region. That way, compression will take place 10 dBs before on sibilant sounds. The "s" sounds should trigger about 5 dB of compression, which will be set to be relatively fast. Normally the manufacturer provides a side chain output, which is just a copy of the input signal, but makes it easier to carry it to the equalizer or other gear. Sometimes the output and input for the side chain are in the same 1/4" stereo connector, like on a mixer insert. The illustration shows the configuration for de-essing.

For live sound this is quite a cumbersome configuration, so it would probably only be worth doing it if de-essing was built into the compressor.

"Pop" compression
Basically the same thing as de-essing, but the "popping" frequencies (around 50 Hz) would be boosted on the equalizer to compress microphone handling pop sounds.
Voices
In live sound applications, the singers often place the microphone very close to their mouths. This generates very large volume changes from small changes in distance to the microphone. Sometimes, the singer may have a tendency to shout. For those reasons, some compression will help us to achieve more uniform levels. On the other hand, human hearing is very sensitive to manipulations on the voice, so compression should be as transparent as possible. Compression for the voice would normally use a soft knee setting and a compression ratio between 3:1 and 6:1, depending on the application. Attack time should be fast, and release time around 0.4 seconds. Level reduction should be about 5 to 7 dB on the loudest passages. For more rock type voices, we can use heavier compression with up to 10:1 ratio, a hard knee setting and attenuation levels up to 15 dB.
A benefit of compressing is a certain feeling of warmth as the artist's whispers can be heard. However, other low level vocal noises such as breathing and lip smack are also emphasized, so a noise gate (if the compressor has a built-in gate, this can be used) is sometimes needed to eliminate or attenuate them.
Acoustic guitar
(These settings are also valid for acoustic sounding electric guitars). Attack times should be in the 5-40 ms range, with around 0.5 s release. Slower times allow the percussive attack of the string to pass through. Ratios should be between 5:1 and 10:1, with around 5-10 dB level reduction.
Electric guitar
In general, the sound of the electric guitar does not need compression in sound reinforcement applications, since the much needed sustain is provided by the guitar amplifier and/or a compression pedal. If necessary, though, attack time should be in the 2-5 ms range (slower if some emphasis is to be preserved), and some 0.5 s release. Ratios should be around 6-10:1, with 8-15 dB compression and a hard knee setting.

For funk type sounds, compression should be higher, using a low thresholds and ratios around 6:1 with a soft knee settings.

Bass drum and snare
By and large, quite substantial compression is applied to the drums, particularly if the drummer's technique is not very consistent. Ratios should be around 4:1, with an attack time somewhere between 1 and 10 ms, closer to the latter if we want to emphasize the attack, which is particularly useful for adding presence and depth to the bass drum. Release times should be between 20 and 200ms; and in any case shorter than the time between drum hits. The threshold should be set such that the compression meter shows just a little compression in the softest parts and up to 15 dB on the loudest beats. Hard knee.

Pre-recorded drum sounds from a drum machine or samples from a drum module triggered by an acoustical or electronic drum set will require less compression that a real drum set picked up with microphones.

Bass
Like electric guitarists, (electric) bass players will normally provide an already compressed signal to the sound guy, given that compression is an integral part of their sound. In any case, bass is a the foundation of rock and pop music, so it is important that its level does not vary too much. Try attack times between 2 and 10 ms (slower times will emphasize the slap), with 0,5 s release. From 4 to 10:1 hard knee compression, meter showing 5-15 dB attenuation.
Brass
1 to 5 ms attack and around 250 ms release. Hard knee compression with 6 to 15:1 ratio and 7-15 dB level reduction.
Synthesizers
In general, these sounds do not have a large dynamic range, so they do not need much compression. For live sound, we can skip the compressor, though sometimes different sounds can have widely different signal levels. A 4:1 ratio may be enough to provide compression on the loudest sounds.
Instruments in general
We will use automatic times, or, if not available, fast attack times and around 0.5 s release. Around 5:1 ratio (soft knee) and about 10 dB compression.
Complete mixes
There are opposite lines of thought with respect to whether compression should me used on the main signals or not. Some compression could be used to generate a slight "pumping" effect and increase perceived signal levels, making it more exciting. Ideally one would use a multi-band compressor for this. If not available, we can use a fast attack time (around 5 ms) and the fastest release that does not create excessive "pumping". 
- Source (doctorproaudio.com)

Dynamics Processors. A tutorial

1.1. Introduction

    It not uncommon to have the need to control the volume (dynamics) of a signal in an automated way.

    We may be trying to avoid too high a level that will clip an amplifier or deafen the audience or send a speaker cone excursing to hyperspace. Or we may just want to regain control on the voice of a singer that will alternate shouting and whispering. Sometimes we will want to avoid background noise when no signal is present.

    To perform all those functions, dynamics processors come to our aid. These are commonly used in live sound reinforcement as well as multi-track recording, while they are not used as often for pre-recorded sound, which is assumed to have canned controlled dynamics. They are also not frequently used on fixed sound reinforcement installations (unless they do live sound), even though volume control on these is sometimes critical.

1.2. Defining dynamics processing

    The concept of a dynamics processor is really not that different from that of a person changing the volume by moving a mixer fader. For instance, if we have a singer that starts singing too loud or getting too close to the microphone, we will reduce the volume on that voice's channel. In that case we are proving compression. When the singer is not singing, we may move the fader all the way down to avoid background noise leaking into the main outputs, thereby acting as a noise gate. There is basically a process whereby someone is listening to the volume changes of a signal and taking the decision of whether the volume needs to be changed or not. The graph below illustrates the process : the auditory system detects the volume changes, and the brain commands the hand to bring the fader up or down as a function of the signal volume.

    This human dynamics processor has its limitations. It can only control one channel, it is slow, and its actions are not repeatable. We might want to use a robot with an articulated arm that would ride a mixer's faders. In practice, though, we choose an electronic device that performs an equivalent function. The electronic version is not very different from a philosophic point of view, though it does away with its limitations.

    As show below, the input to a dynamics processor is split into two. One of these signal copies will be processed by a gain changing element, which will typically be a voltage controlled amplifier (VCA) or a digital equivalent. The other copy goes to a detection circuit that rides the VCA. For the volume changes to be smooth, an envelope generator is used to ramp volume changes and thus avoid abrupt audible changes. The slope and shape of the ramp can be modified, as we will see later. Often we can choose to feed the detector with the input signal or, alternatively, an external signal which is referred to as the "Side Chain" or "Key" signal.

    One of the side effects of using analog volume control elements (VCAs) to process the signal is noise. The quietest VCAs tend to be expensive, and therefore only included in highly professional equipment. As well as a good VCA, a quality dynamics processor also needs a good detection stage, which is by no means easy to design. Which would explain why only a handful of brands on both side of the Atlantic enjoy a reputation for quality dynamics processors and are used for serious sound reinforcement. A good dynamics processor should make it easy to control dynamics transparently, avoiding any kind of audible undesired "pumping" and "breathing" effects. Newer digital units, often built into digital mixers, do not suffer from noise problems, though quality dynamics processing algorithms are not easy to come by, and therefore it will probably be unrealistic to expect quality compression or gating from inexpensive digital products.

1.3. Types

The more commonly used dynamics processors are :

• A compressor / limiter attenuates or limits signals exceeding a pre-defined signal level. There exists an special version of a compressor/limiter called "de-esser", which tames excessive levels of the portion of the frequency spectrum where sibilance occurs.
  A limiter is only a form of compressor.
• A noise gate mutes or attenuates signals below a pre-defined signal level. If it allows the selection of the attenuation level (as opposed to just providing total attenuation, i.e. muting), it is referred to as a "downward expander". 

    There also exists the "true expander", though in practice there are no commercial devices that perform true expansion, which would entail amplifying signals above a specified level and attenuating those below it, therefore expanding the dynamics of a signal.

    We could also speak of digital (those that process a digitized signal) and analog devices. In reality, (good) digital devices can work like their analog counterparts, though normally one uses their processing power to increase manipulation possibilities. For instance, for recording and other non-real time applications, we could compress a signal using a "look-ahead" buffer to make compression/limiting decisions based on what is still to come, so that, for instance, we could start compressing a peak before it exceeds the threshold, avoiding the transient overshoot that would occur on an analog compressor and doing so in an inaudible way. 

1.4. Controls

Different dynamics processors will provide differing sets of controls and indicators. In general, the controls we will find on dedicated units (built-in ones may obviate some of the controls) are :

• Threshold. When the signal goes above or below this level, the processing starts.
• Attack time. It's the time it takes for the signal to get attenuated/limited/muted/amplified. In general, low attack times work better with low frequency signal and, conversely, faster attack times do a better job with high frequency signals. When processing a full range signal, attack times are generally based on the lowest frequencies present in the signal.
• Release time. It's the opposite of attack time, i.e., the time it takes for the signal to go from a processed state to not being processed. Release times are normally longer than attack times.
• Hold Time. Specifies the minimum time that a compressor or gate will process the signal for.
• Ratio. Defines the amount of attenuation or gain that will be applied to the signal. On noise gates, it may be pre-set so that it is just a muting effect.
• Stereo Link. Used to process a stereo signal such that both channels are always processed at the same time, even if one of them has not triggered the processing. This avoids confusing image shifts from the center to one of the sides when only one of the channels is being compressed or gated.
• Automatic. It is becoming more and more common to control some of the parameters defined above (typically attack and release times) automatically based on the signal characteristics. This control activates or deactivates that feature.
• Bypass. Allows comparing of the original versus the processed signal.

1.5. Meters and indicators

The most common visual indicators provided on dynamics processors are given below. You may not always find them, or you may get additional ones:

• Gain or attenuation meter. It is normally implemented as a row of LEDs and indicates the amount of attenuation or gain being applied, to visually judge whether we are over processing or not processing at all. On noise gates we will normally only find an activation light.
• Activation LED. Shows when processing is taking place.
• -Source ( doctorproaudio.com)

Setting Up A PA System

PA Systems come in many different shapes and sizes, ranging from the very elaborate systems used in large stadiums all the way down to a simple microphone patched into your home stereo. Listed below are several of the most common setups.

The Bare Bones

       The above example is about as basic a system as you can get. Hooking it together is relatively simple. The most important thing to remember when hooking up any size PA is the direction of the signal. This is indicated above by the red arrows. The signal starts with your mouth (or drum, or horn, or whatever), then goes through the microphone into the system, then routes its way through the amp, and finally into the speaker where it leaves the system as a much louder sound. A good rule of thumb is to remember that when plugging something in (like an amp), whatever you plug into the input should be coming from the direction of your mouth while whatever you plug into the output should be heading toward the speaker.

       An easier way to think of it might be to think of it as a river, the microphone being upstream, the speaker being downstream, and the amp being a reservoir in between. As the water (the signal) flows from upstream (the Microphone), it must enter the reservoir (the amp) through an input, and then exit the reservoir through an output until finally, it reaches the downstream side (the speaker).

       So, that's the theory (complete with a picturesque metaphor). Now here is the reality. In the example above, you would plug things up in this order.

1. Plug the mic cord into microphone (There is only one place to plug it in. Technically it's an "output").
2. Plug the other end of the mic cord into the "input" of the amplifier (remember, input is coming from the microphone).
3. Plug the speaker cord into the speaker "output" of the amplifier (the signal is flowing out of the amp toward the speaker).
4. Plug the other end of the speaker cord into the "input" on the speaker (the signal is coming from the microphone through the amp to the speaker).

       And there you have it. You have successfully hooked up your first basic PA system. Of course, although it will amplify the sound, this particular system won't be of much practical use to you in any real life playing situation. It still lacks three essential ingredients.

The Essentials

       With the addition of a mixer (soundboard), an equalizer (EQ), and a set of full range speaker cabinets, we have created a small PA system that can be used both for rehearsal and for some gigs. The principle of signal direction stays consistent. As the arrows indicate, the signal again starts at the microphone passing through each component in turn until it reaches the speakers where it exits the system as an audible, much louder sound. It is important to note the order in which the components are hooked up. No matter how many more components (such as effects or compressors) are added, these basic building blocks should always line up in this order relative to each other. The EQ should always be connected somewhere between the output of the mixer and the input of the power amp, the microphone should always be on the input side of the mixer, and the speakers should always follow the amplifier.

Keeping in mind signal direction, the system shown in example 2 should be hooked up like this:

1. Plug the mic cord into the microphone (only one end of the cord will fit).
2. Plug the other end of the mic cord into any "input" channel of the mixer (input comes from the microphone).
3. Plug a high Z cable (patch cable) into the "main out" of the mixer (the signal is flowing out of the board toward the speakers).
4. Plug the other end of this cord into the "input" of the equalizer (the signal is flowing from the microphone).
5. Plug one end of a high Z cord into the "output" of the equalizer (the signal is flowing out of the EQ toward the speakers).
6. Plug the other end of this cord into the "input" of the power amp (the signal is flowing from the microphone).
7. Plug two speaker cords into two speaker "outputs" on the power amp (the signal is flowing through the amp toward the speakers).
8. Plug the other ends of these cords into the "inputs" of the speakers (the signal is coming from the microphone to the speaker).

Warning: Never plug anything other than a speaker into the output of a power amp. A "speaker out" connection carries a very strong signal that can and probably will cause damage to the other components.

 - Source (thefxcode.com)

How to EQ a Room

1. Place a dynamic microphone with a cardioid pattern at the center of the stage and point it toward where a person would speak or perform.
   
   
2. Ask anyone making noise on the stage to leave so they don’t corrupt the EQ process.
3. Set all of the microphone’s EQ channels on the mixing board to flat.
4. Bypass the compressor-limiters, feedback destroyers and other similar processors so you get a clean signal.
5. Move the microphone and instrument channels in the monitor(s) to about where they need to be for the performance.
6. Adjust the Main House EQ so it’s set to the Center position.
7. Turn the mixer’s Input Gain to “Off” and the channel fader to 0dB or -10dB. The fader setting on the Main Out needs to be at -10 or 0dB.
8. Raise the Input Gain slowly until you hear a ringing noise from the speaker system. Turn it down until the sound stops.
9. Manipulate the channel fader to cause the speaker system to ring at a low, nearly steady tone.
   • A real-time analyzer will show you which frequency is ringing, but it isn’t the most accurate reading. If you have a multimeter with frequency, plug it into the Headset Out and use the Pre-Fade Listen (PFL) Output as a signal. The better reading you get, the more accurate your adjustments when you want to properly EQ a room.
10. Decrease the ringing by -3dB on its frequency, which should eliminate the ringing sound.
11. Raise the fader once the ringing stops.
12. Repeat these steps for each slider until several of the frequencies rise at once or until one of the frequencies hits -12dB.
    • If you hit the EQ’s bottom before this happens, there’s a problem with the room itself or with the system design.
13. Turn the master monitor volume back to your preferred level for the performance.
14. Ask someone to stand at the microphone(s) and instrument(s) and check them so you can adjust the levels of each individual input.

Source (Wikihow)

Graphic EQ

The Graphic EQ is ubiquitous, and almost essential in any PA system.

What it is

Physically, it is usually a 19" rack-mounted box with vertical faders, each controlling a limited frequency range. Generally it will be between 1U and 3U in height. It will usually have two identical channels (although some single-channel graphics are available, and have their uses). Each channel will have either ten, fifteen, or thirty-one (sometimes only thirty) frequency bands. Usually, the centre frequency of each band will be an ISO (International Standards Organisation) standard frequency. For reference, over 31 bands these are:

20Hz, 25Hz, 31.5Hz, 40Hz, 50Hz, 63Hz, 80Hz, 100Hz, 125Hz 160Hz, 200Hz, 250Hz, 315Hz, 400Hz, 500Hz, 630Hz, 800Hz, 1kHz, 1.25kHz, 1.6kHz, 2kHz, 2.5kHz, 3.15kHz, 4kHz, 5kHz, 6.3kHz, 8kHz, 10kHz, 12.5kHz, 16kHz, & 20kHz.

On a thirty-one band graphic equaliser, each band covers one third of an octave (you can work this out from the fact that one octave represents a doubling - or, going the other way, halving - of frequency, and there are ten octaves between 20Hz and 20kHz: on a 31-band graphic there are three steps between each doubling of frequency). Ten-band (octave) and fifteen-band (2/3 octave) graphics are not generally adequate for live applications, as each frequency band is too broad for anything more than approximate tone shaping.

What it does

It boosts or cuts a signal in one or more narrow parts of its frequency range. A line taken across the faders gives a graph-like view of the approximate overall effect, which is why this kind of equaliser is called a graphic EQ.

How it works

Each fader controls the level of an individual bandpass filter circuit, dealing with its own specific frequency range. Moving the fader up boosts that range, and moving the fader down reduces it. The combined effect of the filters is to change the overall balance of frequencies.

How do you use it?

If all else fails, read the manual! You can also find general guidelines on many manufacturer websites.

A graphic EQ can be connected to a PA system in one of two ways: on inserts, or in-line. Using the main (left & right or group) mixer inserts will mean that any changes to the graphic settings will be seen on the channel meters, and heard on headphones or listen wedge. This is considered an advantage by many sound engineers. If a graphic EQ is connected in-line (i.e. between the mixer outputs and the crossover or power amp inputs), changes will only be heard through the main or monitor speaker system, and the mixer's meters may not accurately represent the signal strength at the controller or amplifier inputs.

The main use of a graphic EQ in live PA systems is to correct anomalies in the overall sound, and (to a limited extent) control feedback. Overall tone shaping (largely a matter of individual taste) is another common application.

As a corrective measure, cutting a particular frequency is generally more effective than boosting other frequencies. There are several technical reasons for this, but a simple thing to bear in mind is that the peaks stand out, and are therefore more noticeable (imagine a level floor - the theoretical ideal - and think of the difference between stepping on a nail and stepping on a nail-shaped dent in it). Taking out the peaks will have more useful effect (and is easier) than trying to fill the holes. Boosting is the equivalent of creating a more spiky floor, while cutting is the equivalent of creating a more dented one.

The anomalies EQ was designed to address arise from peaks and dips in overall frequency response.

Generally, peaks are caused by resonance. Where resonances arise from instruments or the PA system itself, EQ can limit the damage, but it cannot eliminate them, or remove room resonances (often a major culprit). Also, resonance is a design feature of most musical instruments, and while reducing the most obvious "honk" from a harmonica will help it sit more comfortably in the mix, trying to remove it altogether will rob it of what makes it sound like a harmonica.

Dips in response often result from phase cancellation (over which EQ is completely powerless), masking by obstacles (pillars and walls, over which EQ is relatively powerless), or inefficiency of the sound system in that frequency range (microphones and speakers are the most likely contributors here). The higher frequencies will not reach listeners at the back if the speakers are on tables at waist height, and EQ will be a much less useful solution to this than speaker stands. Try changing the type and position of speakers and mics first.

As a rule of thumb, use any EQ as little as possible. Only resort to EQ if no other remedy is available, and apply it sparingly to the most obvious problem frequencies. A thirty-one band graphic gives you reasonably precise control. If you apply drastic cut to most of the mid-band (the novice's "smile" EQ), you are wasting its precision.

If your experience of using a graphic EQ is limited, try the following (start with all the graphic faders at their mid - 0dB of cut or boost - position):

1. Corrective. Using a CD player or similar source, play some material that has detail throughout the useful frequency range (i.e. 40Hz - 16kHz) through one channel (left, right, or one monitor channel) of the system. Boost each frequency range in turn on the graphic EQ. If the effect of boosting it simply makes that frequency stand out, return the fader to the mid position. If boosting it makes it boom, honk, squawk, shriek or whistle (or if boosting it makes it seem uncomfortably loud), move the fader below the mid position. How far below you move it is a judgment call, and depends on how badly it boomed, honked, squawked, shrieked or whistled. Either repeat this procedure for each channel individually, or copy the settings from the first channel to other channels using the same amplifiers and speakers.

2. Corrective. Using a CD player or similar source (or the mixer's pink noise generator, if it has one) play some pink noise through one channel of the system. Use a calibrated microphone and spectrum analyser to view the output in the listening area (preferably, do this in more than one room position). Use the graphic EQ to reduce the level of any obvious peaks. It is usually unnecessary to get a flat reading (& it might sound a bit grim if you do), so after you have done this check the sound using some material that has detail throughout the useful frequency range, and if necessary reduce the biggest cuts by a few dB until it sounds OK. Either repeat this procedure for each channel individually, or copy the settings from the first channel to other channels using the same amplifiers and speakers.

3. Feedback control. With most of the mics you will be using in place, set up with appropriate gain, & with all relevant input channels open, raise the fader of an output channel until it is on the verge of feeding back. Boost each frequency range in turn on the graphic EQ. If you can get it to the top without feeding back, return it to the mid position. If you can't get it to the top, move it as far below the mid position as it was below the top when it started feeding back. Check the sound using some material that has detail throughout the useful frequency range, and reduce the biggest cuts a little if it doesn't sound OK. If the system feeds back at a lot of frequencies (more than ten, say), you are pretty much at the limit of your usable headroom, and getting it any louder will only be achieved at the expense of noticeable colouration.

3. Overall tone shaping. Using a CD player or similar source, play some material that you know well (preferably with some detail throughout the useful frequency range), and have heard through a high quality sound system. Boost each frequency range in turn on the graphic EQ. Return the fader to the mid position unless boosting that frequency sounds horrible, in which case cut it a bit (in proportion to the horrible).

With any frequency alteration, bear in mind in mind that any change is relative: the effect of boosting or cutting one frequency range will be heard in relation to the overall sound. For example, substantially boosting bass frequencies will make the higher frequencies less noticeable in comparison (so it may sound "duller" or "muddier"). Similarly, cuts in the lower frequencies may make the overall sound "clearer" or "crisper", as well as "thinner".

Most graphic EQs have a master section, with controls that might typically include:

• Input level. Some graphic EQs have meters or overload lights, and allow some attenuation or boosting of the input signal to bring it within the EQ's nominal operating range.

• Output level. Even quite modest amounts of cut or boost in only a few frequency bands can be enough to cause a noticeable difference in the overall volume (both as it is heard, and as it appears on any subsequent meters). An output level control allows you to restore the overall volume, so that the graphic EQ affects only the tonal balance of the sound, not its apparent level.

• High-pass and low-pass controls. Many graphic EQs include shelving EQ, ranging from fixed high-pass and/or low-pass filters (with an In or Out switch), to filters with variable frequency and variable gain. If you want to cut or boost the highest or lowest ranges, use these (rather than the faders).

• EQ in/out. Most graphic EQs allow you to bypass the EQ (on some, high-pass and low-pass controls may also be selected independently). This is useful for instant comparison: it is important that any changes you make actually improve the sound. In/Out comparison is made easier if the output level is adjusted so that switching the EQ in or out has no apparent effect on the overall volume. Sounds can apparently "improve" (or get worse) from changes in volume, so comparison without level matching may be misleading.

If you get the chance, play with a graphic EQ (using a variety of material) until you are familiar with the effect of cutting or boosting different frequencies. You can also improve your frequency recognition by downloading the (free) Simple Feedback Trainer from Sourceforge.

Do you need one?

Almost always.

What sort do you need?

1/3 octave is a must! There are 12 semitones in an octave, so even a 1/3 octave equaliser is relatively coarse when it comes to frequency control. Anything with less resolution - i.e.10-band (octave) or 15-band (2/3 octave) - is only useful for broad tone shaping.

The frequencies below 40Hz and above 16kHz are not vital, so a couple of the standard 31 bands are dispensable. In/Out switches and level controls are useful. High-pass and low-pass switches and/or frequency selectors are useful too. If you have enough rack space, the longer the faders the better.

Other factors (like whether the filters are constant-Q) are more open to debate, but each EQ has its own sound, so - if you can - listen before you buy.

- Source (astralsound.com)

Basic Audio Setup for Office