Showing posts with label Compressors. Show all posts
Showing posts with label Compressors. Show all posts

Friday, 31 August 2012

Setting Mic For Drums

Miking the Drumset in Your Home Recording Studio

If you're like most musicians, getting great-sounding drum recordings seems like one of the world's great mysteries. You can hear big, fat drums on great albums, but when you try to record your drums, they always end up sounding more like cardboard boxes than drums. Fret not — here are some solutions for you.

The room

The room influences the drums' sound more than it influences other instruments'. If you're looking for a big drum sound, you need a fairly live room (one with lots of reflection).
You may be thinking, "But I just have a bedroom for a studio and it's carpeted." No worries, you can work with that. Remember, you have a home studio, so you potentially have your whole home to work with. Here are a couple of ideas to spark your imagination:
  • Buy three or four 4-x-8-foot sheets of plywood and lean them up against the walls of your room. Also place one on the floor just in front of the kick drum. This adds some reflective surfaces to the room.
  • Put the drums in your garage (or living room, or any other room with a reverberating sound) and run long mic cords to your mixer. If you have a studio-in-a-box system, you can just throw it under your arm and move everything into your garage or, better yet, take all this stuff to a really great-sounding room and record.
  • Set up your drums in a nice-sounding room and place an additional mic just outside the door to catch an additional ambient sound. You can then mix this in with the other drum tracks to add a different quality of reverberation to the drums.

Kick (bass) drum

The mic of choice for most recording engineers when recording a kick drum is a dynamic mic. In fact, you can find some large diaphragm dynamic mics specifically designed to record kick drums.
No matter where you place the mic, you can reduce the amount of boominess that you get from the drum by placing a pillow or blanket inside the drum. Some people choose to let the pillow or blanket touch the inside head.
That said, you can place your mic in several ways (all conveniently illustrated in Figure 1):
  • Near the inside head: If you take off the outside head or cut a hole in it, you can stick the mic inside the drum. Place the mic 2 to 3 inches away from the inside head and a couple of inches off center. This is the standard way to mic a kick drum if you have the outside head off or if a hole is cut in it. This placement gives you a sharp attack from the beater hitting the head.
  • Halfway inside the drum: You can modify the preceding miking technique by moving the mic back so that it's about halfway inside the drum. In this case, place the mic right in the middle, pointing where the beater strikes the drum. This placement gives you less of the attack of the beater striking the head and more of the body of the drum's sound.
  • Near the outside head: If you have both heads on the drum, you can place the mic a few inches from the outside head. If you want a more open, boomy sound (and you have the drum's pitch set fairly high), point the mic directly at the center of the head. If you want less boom, offset the mic a little and point it about two-thirds of the way toward the center.
Mic setup for the kick drum

Figure 1: There are several places that you can place a mic to get a good kick drum sound.
The kick drum responds quite well to a compressor when tracking. For the most part, you can get by with settings that allow the initial attack to get through and that tame the boom a little. A sample setting looks like this:
Threshold: -6dB
Ratio: Between 4:1 and 6:1
Attack: Between 40 ms and 50 ms
Release: Between 200 ms and 300 ms
Gain: Adjust so that the output level matches the input level. You don't need much added gain.

Snare drum

The snare drum is probably the most important drum in popular music. The bass guitar can cover the kick drum's rhythm, and the rest of the drums aren't part of the main groove. A good, punchy snare drum can make a track, whereas a weak, thin one can eliminate the drive that most popular music needs.
Because the snare drum is located so close to the other drums, especially the hi-hats, a cardioid pattern mic is a must. The most common mic for a snare drum is the trusty Shure SM57. The mic is generally placed between the hi-hats and the small tom-tom about 1 or 2 inches from the snare drum head (see Figure 2). Point the diaphragm directly at the head. You may need to make some minor adjustments to eliminate any bleed from the hi-hats. This position gives you a nice punchy sound.
Mic setup for Snare drum

Figure 2: The proper placement for the snare drum mic.
Adding compression to the snare drum is crucial if you want a tight, punchy sound. There are a lot of ways to go with the snare. The following settings are common and versatile:
Threshold: -4dB
Ratio: Between 4:1 and 6:1
Attack: Between 5 ms and 10 ms
Release: Between 125 ms and 175 ms
Gain: Adjust so that the output level matches the input level. You don't need much added gain.

Tom-toms

The tom-toms sound best when using a dynamic mic. For the mounted toms (the ones above the kick drum), you can use one or two mics. If you use one mic, place it between the two drums about 4 to 6 inches away from the heads (Figure 3 shows this placement option). If you use two mics, place one above each drum about 1 to 3 inches above the head.
Mic setup for drums tom tom

Figure 3: Miking the mounted tom-toms with one mic.
Floor toms are miked the same way as the mounted tom-toms:
  • Place a single mic a couple of inches away from the head near the rim.
  • If you have more than one floor tom, you can place one mic between them or mic them individually.
If you want to apply compression to the tom-toms, you can start with the settings that for the snare drum in the preceding section.

Hi-hats

The hi-hats are generally part of the main groove and, as such, you want to spend time getting a good sound. You'll probably have problems with a few other mics on the drumset picking up the hi-hats, particularly the snare drum mic and overhead mics. Some people don't bother miking the hi-hats for this reason.
Hi-hats often sound too trashy through the snare drum mic. If you mic hi-hats, make sure that the snare drum mic is picking up as little of the hi-hats as possible by placing it properly and/or using a noise gate (a dynamic processor use to filter unwanted noise).
You can use either a dynamic mic or, better yet, a small diaphragm condenser mic for the hi-hats. The dynamic mic gives you a trashier sound and the small diaphragm condenser mic produces a bright sound. You can work with either by adjusting the EQ. Try adding just a little bit (4dB or so) of a shelf EQ set at 10 kHz to add just a little sheen to the hi-hats.
Place the mic about 3 to 4 inches above the hi-hats and point it down. The exact placement of the mic is less important than the placement of the other instrument mics because of the hi-hats' tone. Just make sure your mic isn't so close that you hit it.
Compression isn't usually necessary when tracking the hi-hats unless you have a drummer whose volume level is inconsistent. In this case, try using the same snare drum settings.

Cymbals

You want to know one secret to the huge drum sound of Led Zeppelin's drummer, John Bonham? Finesse. He understood that the drums sound louder and bigger in a mix if the cymbals are quieter in comparison. So he played his cymbals softly and hit the drums pretty hard. This allowed the engineer to raise up the levels of the drums without having the cymbals drown everything else out. Absolutely brilliant.
Because the drums bleeding into the overhead mics is inevitable and the overhead mics are responsible for providing much of the drums' presence in a mix, playing the cymbals softly allows you to get more of the drums in these mics. This helps the drums sound bigger.
Ask (no, demand) that your drummer play the cymbals quieter. Also use smaller cymbals with a fast attack and a short decay. Doing these things creates a better balance between the drums and cymbals and makes the drums stand out more in comparison.
Small diaphragm condenser mics capture the cymbals' high frequencies well. You can mic the cymbals by placing mics about 6 inches above each cymbal or by using overhead mics set 1 to 3 feet above the cymbals.

The whole kit

Most of the time, you want to have at least one (but preferably two) ambient mics on the drums if for no other reason than to pick up the cymbals. These (assuming you use two mics) are called overhead mics and, as the name implies, they are placed above the drumset. The most common types of mics to use for overheads are large and small diaphragm condenser mics because they pick up the high frequencies in the cymbals and give the drumset's sound a nice sheen (brightness). You also may want to try a pair of ribbon mics to pick up a nice, sweet sound on the overheads.
To mic the drumset with overhead mics, you can use either the X-Y coincident technique or spaced stereo pairs. Place them 1 to 2 feet above the cymbals, just forward of the drummer's head. Place X-Y mics in the center and set up spaced stereo pairs so that they follow the 3:1 rule (the mics should be set up 3 to 6 feet apart if they are 1 to 2 feet above the cymbals). This counters any phase problems. Point the mic down toward the drums and you're ready to record. Figure 4 shows both of these set-ups.
Mic set up for the cymbals

Mic Set up for Cymbals / Crashes

Figure 4: Overhead mics capture the cymbals and the drums.

Monday, 27 August 2012

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.
A typical compressor / Compressor 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.
Gráfica comparativa de umbral alto y bajo / Comparative graph for high and low thresholds
  • 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).
Gráfica comparativa de diferentes relaciones de compresión / Comparative graph for different compression ratios
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.

Gráfico de salida/entrada de un compresor / Output/input graph for a compressor
 
  • 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.
Rótula blanda /soft knee
  • 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.
Compresión de sibilancia / 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.
Human Dynamics Processor
    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.
Electronic Dynamics Processor
    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)