Acoustical Dynamic Range of the System

We described the dynamic range of
the program going into the microphone,
and of the electrical signal
through the console and power amps,
but what of the sound coming out of
the loudspeaker system? If you haven't
already guessed, it also must have the
same dynamic range. If the speakers
aren't capable of this range, then
they're probably going to either distort
(or burn out) on the peaks, be incapable
of responding to the lowest power
levels, or experience some combination
of these problems.
What are the actual sound levels
that must be reproduced? That all
depends on the distance between the
loudspeakers and the audience, and
how loud one wants the sound to be at
the audience. Let's assume that we
don't want to shatter eardrums... we
don't want people in the audience to
feel their ears are one inch from the
lead vocalist's tongue during a maximum
shout. The peak sound level we
might accept as a reasonable facsimile
ofthis excitement is 120 dB SPL. Without
going through the math (we cover
some of that in Part I, Section 5), take
our word for it that these particular
speakers must (cumulatively) generate
130 dB SPL in this particular environment.
Well, we know ifthey generate
130 dB SPL on peaks, they're going to
have to generate 40 dB SPL during the
quietest passages, and will have a
90 dB dynamic range.
From this, we also know that ifthe
sound reaching the audience during
peaks was attenuated by air and
distance by 10 dB from 130 dB SPL to
120 dB SPL, the 40 dB SPL generated
by the speakers during quiet passages
will also be attenuated. When the
40 dB drops to 30 dB, it will be below
the ambient noise level in the audience.
This means that the audience
may not hear the very quietest parts of
the show. This illustrates why some
electronic manipulation of dynamic
range is often called for. In this case,
compression of the loudest peaks would
allow the level to be turned up so the
quiet passages are louder.

Electrical Dynamic Range of the Sound System

What is the dynamic range required
of the sound system for such a concert?
The electrical signal level in the sound
system (given in dBu) is proportional
to the original sound pressure level (in
dB SPL) at the microphone. The actual
electrical levels, of course, will depend
on the sensitivity of the microphones
the g?-in in the preamplifiers, power '
amplifiers, and so forth, but these
values, once established, remain fairly
constant so well assume they are
constant and look at the nominal level
(that is, the level specified and
designed for) in the electronics.
Thus, when the sound levels reach
130 dB SPL at the mic, the maximum
line levels (at the mixing console's
output) may reach +24 dBu
(12.3 volts), and maximum output
levels from each power amplifier may
peak at 250 watts (of course, there may
be dozens of such power amplifiers
each peaking at 250 watts, but let's
keep things simple for now). Similarly,
when the sound level falls to 40 dB
SPL, the minimum line level falls to 66
dBu (388 microvolts) and power
amplifier output level falls to 250
nanowatts (250 billionths of a watt).
When the acoustical program from
the mic is converted to an electrical
signa.l at .the mixing console output,
does It still have the same dynamic
range?

Dynamic Range of a Typical Rock Concert

Well describe a concert with about
the widest dynamic range you're ever
likely to encounter. The sound levels at
the microphones (not in the audience)
may range from 40 dB SPL (the audience,
wind, and traffic noise at the mic
during a very quiet, momentary pause)
to 130 dB SPL (beyond the threshold of
pain... but then, the performer is
shouting into the mic, not into someone's.
ear). What is the dynamic range
of this concert? It is obtained by
subtracting the noise floor from the
peak levels:
Dynamic Range...
= (Peak Level) - (Noise Floor)
= 130 dB SPL - 40 dB SPL
=90dB
The concert has a 90 dB dynamic
range at the microphone.

NOTE: We specified the dynamic
range injust plain "dB," not in "dB
SPL." Remember, dB is a ratio and
~n this case we are simply des~bmg
the relationship of 130 dB SPL
to 40 dB SPL; the difference is
90 dB, but that has nothing at all
to do with a sound level of 90 dB
SPL referenced to 0.0002 dynes per
cm2. Dynami.c range I.S nearly
always specified in dB, and should
never be expressed in dB SPL,
dBm, dBu or any other specifically
referenced dB value.

DyNAMic RANGE

The difference, in decibels, between
the loudest and the quietest portion of
a program is known as its dynamic
range. Sometimes, the quietest portion
of a program will be obscured by
amb~ent noise. In this case, the dynarmc
range is the difference in dB
b
' ,
etween the loudest part of the program
and the noise floor. In other
words, dynamic range defines the
maximum change in audible program
levels.
Dynamic range also applies to sound
systems. Every sound system has an
inherent noise floor, which is the
residual electronic noise in the system.
!he dynamic range of a sound system
IS equal to the difference between the
peak output level of the system and the
electro-acoustic noise floor.

What is a sound mix?

Audio mixing is the process by which multiple sounds are combined into one or more channels. In the process, the source signals' level, frequency content, dynamics, and panoramic position are manipulated and effects such as reverb may be added.

What does it mean when you master a song?

Mastering, a form of audio post production, is the process of preparing and transferring recorded audio from a source containing the final mix to a data storage device (the master); the source from which all copies will be produced (via methods such as pressing, duplication or replication).

What is mastering a song?

A mastering engineer can unify your album with skillful use of EQ, gain, and compression to give it a consistent sound from track to track. This process also allows the mastering engineer to pump up the volume of your overall album so it's as hot as can be and make it sound unbelievable.

What is the difference between mixing and mastering a song?

Basically, mixing is the step before mastering that involves adjusting and combining individual tracks together to form a stereo audio file after mixdown. The stereo file is then mastered, which ensures that the various songs are clearly polished and form a cohesive whole on an album.

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EQ Or Not To EQ

Equalizing or tone shaping is clearly a very important tool in our audio arsenal. By the way, wouldn’t Audio Arsenal be a good name for a band? Of nerds?
Anyway… I believe that there are two reasons for using EQ. The first, and perhaps most important in the overall scheme, is to properly tune the system for the type of response that is wanted or needed.
Some frequencies that might excite the room too much can be reduced, and tailor the low-frequency response to match the volume of the room, etc.
Of course, don’t forget that there are acoustical problems that cannot be solved with electronics. I know, call me a heretic.
But the other kind of EQ I’m thinking of more is the “color” applied to individual channels. A lot of this is already determined by microphones and where they’re placed. Certainly a snare sounds different when mic’ed with a dynamic vs. a condenser. And really, it does start at the microphone.
But let’s say you’ve chosen the best mic for the source, and put it in the right place, and now you need to do a bit of final tweaking on the channel EQ for the sound to be “perfect.”
I forget who said it, but “people don’t go to the concert to hear the kick drum.” Whoever it was has never been to a Metallica concert. I was at one back in the mid-1990s (the “Black Album” tour). All three of the opening bands had already played, and we were in the middle of the break before the headliner came out. Anticipation was growing… And growing.
And then at one point, the drum tech came out, sat down, and stomped on the kick pedal, sending a thunderous sound through the audience. And everyone went wild! Who would have thought that a single-note kick drum solo would have brought the crowd to its feet!
But I digress.
My point is that each instrument may sound great on its own, but may not sit properly in the mix.

Complete Loudspeaker Management System does all that the PA+ can do plus it features:

• An Ethernet connection so you can connect your wireless router and use your portable device (Android, IOS, Mac, and Windows) to control the PA2 from anywhere in the room. The larger graphics and touchscreen control of your portable device make it easier to operate, too.
• An input delay so you can align the front PA speakers with the amps and instrument speakers from the back of the stage, for total band clarity.
• The Auto-EQ uses an 8-band parametric equalizer and plays short frequency sweeps for its setup, which is a lot less annoying than using pink noise, like the other DriveRack models do.
• There's a dual 31-band graphic equalizer on the input section that includes Quick Curve options so you can instantly dial in an EQ curve that'll work for you.
• The PA2 has 75 factory presets and 75 user presets.

Advanced Feedback Suppression (AFS) wizard

The feedback suppression wizard further tweaks the equalization to prevent the howling noise that happens when sound from the speakers is picked up a microphone and then re-amplified.
The wizard walks you through several steps:

• After all your microphones have been set up and activated on stage, you start by turning the mixer's master volume down.
• Then the AFS Wizard will ask you how many of the 12 available notch filters you want to be fixed and how many you want to be live. The fixed filters stay put after being set, the live filters will change and move to a new frequency if needed, which comes in handy if mics get moved during a show.
• Next you choose how narrow you want the notch filters to be. You can choose a wide 1/5 of an octave band, for a speech-only application, or narrow it all the way down to an incredible 1/80 of an octave for precise feedback control that won't adversely affect the music.
• Then you turn your mixer's master volume beyond normal to try and induce feedback. The AFS will instantly set the filters to prevent any ringing or howling, and you're set to return to normal levels and go for the show.

Auto-EQ wizard

To use the auto-EQ wizard you first choose a target response curve from a list of thirteen suggested equalization curves. Then, using the built-in RTA, noise generator, and the reference microphone, the Auto-EQ sets the DriveRack's equalizers to match that curve as closely as possible. It only takes a few seconds, but since it's done at performance volume, you'll probably want to do it while wearing ear plugs, and when nobody else is in the room.

Wide range

The RTA-M features flat response from 20 to 20,000 Hz, allowing you to get reliable measurements and make your "pinking" more precise. The RTA-M runs on the phantom power provided by your DriveRack, and comes with a case and microphone clip

Perfect pickup

dbx designed the RTA-M reference microphone to measure room acoustics in conjunction with their DriveRack Series processors. When used with the System Setup and Auto EQ Wizards, its flat frequency response and omnidirectional pickup pattern make it ideal for optimizing system response, even in the most difficult environments

Setup wizard

The setup wizard matches the levels of your left and right speakers, and sets the limiter for your amplifiers and speakers so no amount of raising the volume of the mixer will ever damage your speakers.

All DriveRack systems include:

• Advanced Feedback Suppression, which senses whenever a particular frequency starts to “ring out” and then reduces the volume at that frequency to stop the noise
• Auto-EQ that automatically adjusts your system's frequency response to attain the best-sound possible
• Setup wizard, which balances the levels of the system's amps and speakers and sets a peak limiter to protect the speakers from being overdriven
• LCD arrays that indicate input and output levels, and a large selector knob to scroll through menus and enter data.

What's a speaker management system?

A speaker or PA management system is a multi-function signal processor, connected between the mixer and the amplifiers of a PA. Once it has been properly set, it automatically keeps the final mix sounding good and the speakers safe from damage.
Features include limiting and equalization to match the equipment and music to a venue's space. A PA management device may also provide crossover and signal-routing functions for a large multi-speaker system.

Anyone new to vocal eq’ing should remember the following points:

1. Using some foundational eq’ing to get started.
2. EQ in a way that matches the style of music you are mixing – listen to the same song from a professional recording to hear it.
3. EQ to match what you want to hear.  Don’t ask the question “does this sound good?”  Ask the question “does this sound like I want it to sound?“

Get Smoothness”

Smoothness” – much of the natural freq’s of a voice are in the mid-range freq’s.  By cutting or boosting in the mid-range, we can optimize the sound so it sounds best.  We can also boost or cut to separate it out in the mix from other vocals and or instruments that might be vying for the same frequencies.  Think of it like this, a bass is low end.  A flute and even a drum kit’s high hat are on the high end.  You want to fill up the sonic space (freq’s) with as much as you can over the whole range.  When you get a bunch of stuff in the same place, that gets you a muddy sound.

Same with bass, boost a little or cut…or not.  Here’s the thing…the best thing you can do is get a solo track of a vocal on CD (or do this during practice).  Move the EQ dials, one at a time, to an extreme.  Once you hear what is bad, it’s easier to then move the dial until you hear what sounds good.  We just need to know the bad to help identify the good.

Additionally, if you have singers with slightly wavering voices or young singers – teenagers, you can add a little vocal reverb effect that will even out their vocal fluctuations.

Maybe it’s something deep within our minds that says “if there is a problem with the sound then we need to boost the problem area.”  However, when it comes to EQ and even cross-channel balancing, this is not always the case.  Cutting frequencies is often the cure.  For example, if two instruments are sharing common frequencies and you want one to stand out, don’t boost the frequency for that instrument.  Cut the frequency of the other.  Lowering other channel volumes can bring the boost to the single channel that you need.  Louder isn’t always better.

One last VERY HELPFUL TIP!  If you are having troubles with cleaning up a male vocal, take a 3-6 dB cut in the 325 to 350 Hz range.  this is where a lot of the muddiness in a vocal can be found.

Lastly, vocal eq is where the science of audio manipulation is surpassed by the art of audio manipulation.  The above tips I mentioned might get you exactly what you want to hear.  But more than likely, they will only point you in the right direction that will eventually lead you to the sound you want.  Listening to several genres of music, you can hear the different types of vocal EQ for that style of music.  Then you add in individual taste in EQ.  You might think that a singer’s vocal EQ is perfect but they think it needs more breathiness or more brightness or more bass.  It’s quite subjective, sorry to say.

Get Brightness, not harshness”

As for “brightness,” much of your high frequencies control how bright and airy a vocal can sound.  For example, crank the high EQ all the way up during a practice on a vocal mike.  It will be very airy and then you can reduce it to where it sounds good.  So much of what sounds good comes with having a good ear and knowing your music.

Treat Harsh Vocals: To soften vocals apply cut in a narrow bandwidth somewhere in the 2.5KHz to 4KHz range.”

This is where a lot of what is being done is dependent on the type of mixer you have.  For example, if you run an analog mixer, you most likely have a semi-parametric EQ.  This means you EQ via knobs on each channel with control for gain (amplitude) and the center frequency, however, you can’t control the width of the affected frequencies – the bandwidth.  Thus, your EQ adjustments affect a wide range of frequencies at once – like moving a mountain peak back and forth – it means you have to move a lot of the mountain with it.

Some EQ’s allow the user to work on EQ like a surgeon, making freq cuts/boost in very specific ranges. Harsh vocals can be reduced by sweeping over the mid/ high-mid frequencies until you hear the harshest vocal sound.  Then you cut (reduce) those frequencies via the EQ.  This would be the case with a parametric EQ where you can control the center frequency, the gain/amplitude cut or boosted, and the bandwidth, sometimes known as the Q. 

General: Roll off below 60Hz using a High Pass Filter.”

Each channel on a mixer usually has an HPF (high pass filter) button.  By pressing this button, we are dropping all audio frequencies below a certain level.  As an example, I’ve got a Yamaha mixer with a “/80” button – which means HPF and drop all freq’s below 80 Hz.  Freq’s this low are typically your low bass notes and kick drum.  If any low frequencies seep into the vocal microphone, they can muddy up the sound.  So, it’s good to use a HPF on any channel that’s not dealing with low-end frequencies.  With experience, you might find some vocals sound better without the HPF but if you are new to sound, HPF is a good place to start.

The Reasons For Using Vocal EQ

When a voice is recorded through a microphone, we need to add a bit of EQ to the voice to bring out its natural qualities.  For example, when you hear me talk in a room, you hear some natural reverberation in the room.  In EQ’ing, you can add that natural reverb back in because the microphone might not pick it up in your particular recording environment.

Additionally, vocal EQ’ing is performed to enhance the vocals so they sound best in our environment as well as within the band and within the song.  And this is where most of your work is focused.

EQ Vocals: The Five Primary Areas of Modification

1. General: Roll off below 60Hz using a High Pass Filter. This range is unlikely to contain anything useful, so you may as well reduce the noise the track contributes to the mix.
2. Treat Harsh Vocals: To soften vocals apply cut in a narrow bandwidth somewhere in the 2.5KHz to 4KHz range.
3. Get Brightness, Not Harshness: Apply a gentle boost using a wide-band Bandpass Filter above 6KHz. Use the Sweep control to sweep the frequencies to get it right.
4. Get Smoothness: Apply some cut in a narrow band in the 1KHz to 2KHz range.
5. Bring Out The Bass: Apply some boost in a reasonably narrow band somewhere in the 200Hz to 600Hz range.

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Stereo link. Output gain. Automatic mode.Side Chain Listen. Bypass.

• 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. Compressors 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. Meaningful comparisons will require level matching between compressed and uncompressed signals (the output gain control can be used for this).

Knee.

• 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.

Compression ratio

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.

Compression ratio.

• 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).
• 
  

Attack time.Release time.

• 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 known 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.

Threshold.

• 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.

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.

Introduction to dynamics processors

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. The dynamic range (when expressed in decibels, as is commonly done) is the difference between the loudest and the softest 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.

EQ Rules of Thumb

When equalizing you must pay attention to some of the most common mistakes that sometimes even the seasoned engineer makes. First of all try always to keep your equalization gain parameter at a reasonable level. As a general rule, avoid cutting or boosting by more than 6 dB unless absolutely necessary. If for some reason you see that some of your EQ settings go over this limit try to question why and see if there is a better solution to the problem. The same can be said for situations where you end up boosting (or cutting) several frequencies at the same time that have the only effect of raising (or lowering) the overall volume of the track without really affecting its sonic content. In this case try to bypass the equalizer and experiment with volume changes instead.

You will be surprised how much a small amount of equalization can change, and hopefully improve your mix. Try to hear the sound in your head that you want to achieve through equalization and avoid playing around with the parameters trying to “find” the perfect sound. 

Application

Remember that equalization is a problem-solving procedure. This means that there's no point in playing around with the settings if you don’t know what you want to achieve and how the final result should sound. A good approach to equalization is to listen carefully to the soloed track and come up with a list of things you might want to improve or correct. If you are using a parametric EQ, the next step is to bring up the gain and sweep across the frequency range until you find the frequency range you want to cut or boost. After that, boost or cut as desired. 

Keep in mind that when equalizing you will have to make small adjustments every time you add tracks to the mix since the frequencies and respective ranges of the other instruments affect the way an instrument sounds. The most important concept here is to be able to emphasize the characteristic frequencies of the track you are working on and eliminate frequencies that do not enhance its sonic features in any particular way. In fact, you should be able to “carve” a small niche inside the audible range for each instrument and section so that it is clearly intelligible and not masked by other instruments. If the mix sounds muddy and cluttered you should start trying to focus on which instruments contribute to the clutter. Try to use the equalizer to add clarity by gently shifting the center of each instrument involved so that they do not overlap with each other. As a general rule it is always better to cut than to boost, mainly because the human ear is more used to a reduction than to an augmentation in intensity of frequencies. 

Equalization: Overview

An equalizer, in its broad description, allows you to boost or cut the volume of specified frequencies. During the mix, equalization can be effectively used in different ways to correct problems that were created during the recording session or from incompatibility among instruments. Equalization can also be used in a creative way in order to produce original effects.

No matter which way you are going to use an equalizer, there are few notions and concepts that you should know before beginning an equalization session. First, equalizers are generally used as inserts on the channel and not as auxiliary sends. Next, you have to be familiar with the most used types of equalizers in a digital audio workstation setting. Among the several types of equalizers available nowadays there are five main categories that have proven to be the most useful in a mixing situation: peak, high shelf, low shelf, high pass and low pass. 

Equalization

20-60 Hz	- Cut to reduce rumble and noises related to electric interferences	It is a good idea to always reduce by 4 to 6 dB this area in order to lower the low frequencies noise
60-80 Hz	- Boost to add fullness to low frequency instruments such as bass and bass drums
100-200 Hz	- Boost to add fullness to guitars, French horns, trombones, piano, snares - Cut to reduce “boomy” effect on mid-range instruments	This frequency range effectively controls the powerful low-end of a mix
200-300 Hz	- Cut to reduce low and unwanted resonances on cymbals - Boost to add fullness to vocal tracks	Be careful not to boost too much of this frequency range in order to avoid adding muddiness to the mix
400-600 Hz	- Cut to reduce unnatural “boxy” sound on drums - Boost to add presence and clarity to bass	This frequency range can also be effective to boost the low range of the guitar
1.4-1.5 kHz	- Boost for intelligibility of bass and piano
2.8-3 kHz	- Boost to add clarity to bass - Boost to add attack and punch to guitars	This range can also be used effectively to add clarity on vocal parts
5-6 kHz	- Boost for vocal presence - Boost for attack on piano, guitars and drums	A general mid-range frequency area to add presence and attack
7.5-9 kHz	- Cut to avoid sibilance on vocal and voice - Boost to add attack on percussions - Boost to add clarity, breath and sharpness to synthesizers, piano and guitars	A mid-high range area that controls the clarity and the attack of the mid-high range instruments
10-11 kHz	- Boost to increase sharpness on cymbals - Boost to add sharpness on piano and guitars - Cut to darken piano, guitars, drums and percussions	High range section that affects clarity and sharpness
14-15 kHz	- Cut to reduce sharpness on cymbals, piano and guitars - Boost to add brightness on vocals - Boost to add real ambience to synthesized and sampled patches
18 kHz	- Cut to reduce hiss noise - Boost to add clarity to overall mix	A delicate high range section that should require drastic positive or negative gain settings only in extreme situations

Principles of Equalization

High Shelf	It cuts or boosts the frequency at the cutoff and all the frequencies higher than the set cutoff point. It has only two parameters: the cutoff frequency and the gain	It is usually used in the mid-high and high end of the spectrum. It can be effectively used to brighten up a track by using a positive gain of 3 or 4 dB and a cutoff frequency of 10 kHz and higher (be careful because this setting can increase the overall noisiness of the track). It can also be used to reduce the noise of a track by reducing by 3 or 4 dB frequencies around 15 kHz and higher
Low Shelf	It cuts or boosts the frequency at the cutoff and all the frequencies lower than the set cutoff point. It has only two parameters: the cutoff frequency and the gain	It is usually used in the low-mid and low range of the audible spectrum to reduce some of the rumble noise caused by microphone stands and other low end sources
High Pass	It cuts all the frequencies below the cutoff point. It has only one parameter which is the cutoff frequency.	It is a very drastic filter. It is often used to cut very low rumble noises below 60 Hz
Low Pass	It cuts all the frequencies above the cutoff point. It has only one parameter which is the cutoff frequency.	It is a very drastic filter. It is often used to cut very high hiss noises above 18 kHz. Use with caution in order to avoid cutting too much high end of the track.

Caution:

 DO NOT ACTIVATE A GLOBAL PHANTOM
POWER SWITCH IF AN UNBALANCED SIGNAL
SOURCES IS CONNECTED TO ANY MIC INPUT.
Because of the voltage present on pins 2 and 3 of the XLR
connector, you will damage your microphone/signal source.

+48v or Phantom Power

Some microphones, known as condenser mics, require
battery power to operate. Alternatively the power may be
provided by the console. This is known as ‘phantom power’
and runs at 48vDC. Simply press “Phantom Power” and
any condenser mics connected will operate without the
need for batteries.

Stereo Returns

These allow signals from external equipment, such as effects
units, to be returned to the mixer and routed to the stereo
Mix or Groups, without using up valuable input channels.

Meters

Normally they show mix output levels. When any Solo
button is pressed, the meters automatically switch to show
the solo level. They provide visual indication of what’s
going on in your mixer.

AFL

Allows monitoring of the actual signal leaving the Aux
Masters.

Aux Masters

These govern the overall output levels from the auxiliary
outputs and therefore the amount of signal going to an
effects unit or a musician’s foldback mix.

Track Tape Returns

Allow you to connect the outputs of your cassette or DAT
player and listen back to your completed masterwork. They
may also be used for playing pre-show music at a gig using
2-Track to Mix switch (not shown in illustration).