POWERED MIXER EMX5016CF Owner's Manual

Making the Most of Your Mixer
An Introduction
You've got yourself a mixer and now you're ready to use it.
Just plug everything in, twiddle the controls, and away you go ? right? 
Well, if you've done this before you won't have any problems, but if this is the first time you've ever used a mixer you might want to read through this little tutorial and pick up a few basics that will help you get better performance and make better mixes.

A Place for Everything and Everything in its Place

A Plethora of Connectors-What Goes Where?
Questions you're likely to encounter when setting up a system for the first time might include "Why all these different types of connectors on the back of my mixer?" and "What's the difference?".
Let's start by taking a look at the most common connector types.

The Venerable RCA Pin Jack
This is the "consumer connector," and the one that has been most commonly used on home audio gear for many years. Also known as "phono" jacks (short for "phonogram"), but the term isn't used much these days-besides, it's too easily confusable with "phone" jacks, below. RCA pin jacks are always unbalanced, and generally carry a line-level signal at -10 dB, nominal. You're most likely to use this type of connector when connecting a CD player or other home audio type source to your mixer, or when connecting the output of your mixer to a cassette recorder or similar gear.

The Versatile Phone Jack
The name "phone jack" arose simply because this configuration was first used in telephone switchboards. Phone jacks can be tricky because you can't always tell what type of signal they're designed to handle just by looking at them. It could be unbalanced mono, unbalanced stereo, balanced mono, or an insert patch point. The connector's label will usually tell you what type of signal it handles, as will the owner's manual (you do keep your manuals in a safe place, don't you?). A phone jack that is set up to handle balanced signals is also often referred to as a "TRS" phone jack. "TRS" stands for Tip-Ring-Sleeve, which describes the configuration of the phone plug used.

(Figure)
Stereo/TRS phone plug
Sleeve
Ring
Tip
Mono phone plug

The Sturdy XLR
This type of connector is generally referred to as "XLR-type," and almost always carries a balanced signal. If the corresponding circuitry is designed properly, however, XLR-type connectors will also handle unbalanced signals with no problem. Microphone cables usually have this type of connector, as do the inputs and outputs of most professional audio gear.

(Figure)
Male
Female

Balanced, Unbalanced-What's the Difference?
In a word: "noise." The whole point of balanced lines is noise rejection, and it's something they're very good at. Any length of wire will act as an antenna to pick up the random electromagnetic radiation we're constantly surrounded by: radio and TV signals as well as spurious electromagnetic noise generated by power lines, motors, electric appliances, computer monitors, and a variety of other sources. The longer the wire, the more noise it is likely to pick up. That's why balanced lines are the best choice for long cable runs. If your "studio" is basically confined to your desktop and all connections are no more than a meter or two in length, then unbalanced lines are fine-unless you're surrounded by extremely high levels of electromagnetic noise. Another place balanced lines are almost always used is in microphone cables. The reason for this is that the output signal from most microphones is very small, so even a tiny amount of noise will be relatively large, and will be amplified to an alarming degree in the mixer's high-gain head amplifier.

To summarize
(Table)
Microphones:	Use balanced lines.
Short line-level runs:	Unbalanced lines are fine if you're in a relatively noise-free environment.
Long line-level runs:	The ambient electromagnetic noise level will be the ultimate deciding factor, but balanced is best.
(Table End)

How Do Balanced Lines Reject Noise?
Skip this section if technical details make you queasy.
Balanced lines work on the principle of "phase cancellation": if you add two identical signals out of phase (i.e. one signal is inverted so its peaks coincide with the troughs in the other signal), the result is ? nothing. A flat line. The signals cancel each other out.
While the desired audio signals in the hot and cold conductors are out of phase, any noise induced in the line will be exactly the same in both conductors, and thus in phase. The trick is that the phase of one signal is reversed at the receiving end of the line so that the desired audio signals become in-phase, and the induced noise suddenly finds itself out of phase. The out-of-phase noise signal is effectively canceled while the audio signal is left intact. Clever, eh?

(Figure)
Balanced noise cancellation
Phase inversion
Source
Hot (+)
Cold (-)
Ground
Noise
Cable
Phase inversion
Noise cancelled
Noise-free signal
Receiving device
Unbalanced noise
Source
Noise
Cable
Receiving device

A balanced cable has three conductors:
1)	A ground conductor which carries no signal, just the "ground" or "0" reference against which the signal in the other conductors fluctuates.
2)	A "hot" or "+" conductor which carries the normal-phase audio signal.
3)	A "cold" or "-" conductor which carries the reverse-phase audio signal.

(Figure)
Balanced
Hot
Cold
Shield (Ground)
Outer Insulation
Unbalanced

Signal Levels and the Decibel
Let's take a look at one of the most commonly used units in audio: the decibel (dB). If the smallest sound that can be heard by the human ear is given an arbitrary value of 1, then the loudest sound that can be heard is approximately 1,000,000 (one million) times louder. That's too many digits to deal with for practical calculations, and so the more appropriate "decibel" (dB) unit was created for sound-related measurements. In this system the difference between the softest and loudest sounds that can be heard is 120 dB. This is a non-linear scale, and a difference of 3 dB actually results in a doubling or halving of the loudness.
You might encounter a number of different varieties of the dB: dBu, dBV, dBM and others, but the dBu is the basic decibel unit. In the case of dBu, "0 dBu" is specified as a signal level of 0.775 volts. For example, if a microphone's output level is -40 dBu (0.00775 V), then to raise that level to 0 dBu (0.775 V) in the mixer's preamp stage requires that the signal be amplified by 100 times.
A mixer may be required to handle signals at a wide range of levels, and it is necessary match input and output levels as closely as possible. In most cases the "nominal" level for a mixer's input and outputs is marked on the panel or listed in the owner's manual.

(Figure)
Most professional mixers, power amplifiers, and other types of equipment have inputs and outputs with a nominal level of +4 dBu.
The inputs and outputs on home-use audio gear usually have a nominal level of -10 dBu.
Microphone signal levels vary over a wide range depending on the type of microphone and the source. Average speech is about -30 dBu, but the twittering of a bird might be lower than -50 dBu while a solid bass drum beat might produce a level as high as 0 dBu.

Making Better Mixes
Approaching the Mix-Where Do You Start?
Mixing is easy, right? Just move the faders around until it sounds right? Well, you can do it that way, but a more systematic approach that is suited to the material you're mixing will produce much better results, and faster. There are no rules, and you'll probably end up developing a system that works best for you. But the key is to develop a system rather than working haphazardly. Here are a few ideas to get you started:

Faders Down
It might sound overly simple, but it is usually a good idea to start with all channel faders off-all the way down. It's also possible to start with all faders at their nominal settings, but it's too easy to lose perspective with this approach. Start with all faders down, then bring them up one by one to fill out the mix. But which channel should you start with?

Example1: Vocal Ballad Backed by Piano Trio
What are you mixing? Is it a song in which the vocals are the most important element?
If so you might want to build the mix around the vocals. This means bringing the vocal channel up to nominal first (if your level setup procedure has been done properly this will be a good starting point), and then adding the other instruments.
What you add next will depend on the type of material you are working with and your approach to it. If the vocals are backed by a piano trio and the song is a ballad, for example, you might want to bring in the piano next and get the vocal/piano relationship just right, then bring in the bass and drums to support the overall sound.

Example2: Funky R&B Groove
The approach will be totally different if you're mixing a funky R&B number that centers on the groove. In this case most engineers will start with the drums, and then add the bass. The relationship between the drums and bass is extremely important to achieve the "drive" or groove the music rides on. Pay particular attention to how the bass works with the kick (bass drum).
They should almost sound like a single instrument-with the kick supplying the punch and the bass supplying the pitch. Once again, there are no rules, but these are concepts that have been proven to work well.

To EQ or Not to EQ
In general: less is better. There are many situations in which you'll need to cut certain frequency ranges, but use boost sparingly, and with caution. Proper use of EQ can eliminate interference between instruments in a mix and give the overall sound better definition. Bad EQ-and most commonly bad boost-just sounds terrible.

Cut for a Cleaner Mix
For example: cymbals have a lot of energy in the mid and low frequency ranges that you don't really perceive as musical sound, but which can interfere with the clarity of other instruments in these ranges. You can basically turn the low EQ on cymbal channels all the way down without changing the way they sound in the mix. You'll hear the difference, however, in the way the mix sounds more "spacious," and instruments in the lower ranges will have better definition. Surprisingly enough, piano also has an incredibly powerful low end that can benefit from a bit of low-frequency roll-off to let other instruments-notably drums and bass-do their jobs more effectively. Naturally you won't want to do this if the piano is playing solo.

The reverse applies to kick drums and bass guitars: you can often roll off the high end to create more space in the mix without compromising the character of the instruments. You'll have to use your ears, though, because each instrument is different and sometimes you'll want the "snap" of a bass guitar, for example, to come through.

(Figure)
The fundamental and harmonic frequency ranges of some musical instruments.
Cymbal
Piano
Bass Drum
Snare Drum
Bass
Trombone
Guitar
Trumpet
Fundamental:	The frequency that determines the basic musical pitch.
Harmonics:	Multiples of the fundamental frequency that play a role in determining the timbre of the instrument.

Some Frequency Facts
The lowest and highest frequencies than can be heard by the human ear are generally considered to be around 20 Hz and 20,000 Hz, respectively. Average conversation occurs in the range from about 300 Hz to about 3,000 Hz. The frequency of a standard pitchfork used to tune guitars and other instruments is 440 Hz (this corresponds to the "A3" key on a piano tuned to concert pitch). Double this frequency to 880 Hz and you have a pitch one octave higher (i.e. "A4" on the piano keyboard). In the same way you can halve the frequency to 220 Hz to produce "A2" an octave lower.

Boost with Caution
If you're trying to create special or unusual effects, go ahead and boost away as much as you like. But if you're just trying to achieve a good-sounding mix, boost only in very small increments. A tiny boost in the midrange can give vocals more presence, or a touch of high boost can give certain instruments more "air." Listen, and if things don't sound clear and clean try using cut to remove frequencies that are cluttering up the mix rather than trying to boost the mix into clarity.
One of the biggest problems with too much boost is that it adds gain to the signal, increasing noise and potentially overloading the subsequent circuitry.

(Figure)
Signal Level (dB)
LOW Boost
LOW Flat
LOW Cut
MID Flat
MID Boost
MID Cut
HIGH Boost
HIGH Flat
HIGH Cut
Frequency (Hz)

Ambience
Your mixes can be further refined by adding ambience effects such as reverb or delay. On the EMX mixers these effects are built in. The internal DSP (Digital Signal Processor) can be used to add reverb or delay to individual channels in the same way as external effects processors, with the extra connections required by, or the loss in sound quality often caused by external processing. (Refer to page 22).
You need to be careful not to overdo effects, however, because going to far can undermine the clarity and quality of your mix. Use your ambience effects just enough to create the required feeling of depth, but no more than is necessary to keep your sound clean.

Reverb and Delay Time
A variety of reverb and delay effect programs are provided, and nearly all of then have a reverb/delay time parameter than can be adjusted via the panel PARAMETER control.
Small adjustments to the reverb/delay time can actually have a significant effect on the sound. The optimum reverb time for a piece of music will depend on the music's demo and density, but as a general rule longer reverb times are good for ballads, while shorter reverb times are more suited to up-tempo tunes. Delay times can be adjusted to create a wide variety of "grooves", and you need to select the time that best suits the music. When adding delay to a vocal, for example, try setting the delay time to dotted eighth notes corresponding to the tune's tempo.

Reverb Tone
Different reverb programs will have different "reverb tone" due to differences in the reverb time of the high or low frequencies, or differences in the overall frequency response of the reverb sound. Always be careful not apply too much reverb, particularly in the high frequencies. In addition to resulting in unnatural sound, excessive high-frequency reverb can interfere with the high frequencies in other parts of the mix. If you can hear more reverb than direct sound in the upper frequency range, try selecting a different effect program. It's always a good idea to choose a reverb program that gives you the depth you want without detracting from the clarity of the mix.

Reverb Level
It's amazing how quickly your ears can lose perspective and fool you into believing that a totally washed-out mix sounds perfectly fine. To avoid falling into this trap start with reverb level all the way down, then gradually bring the reverb into the mix until you can just hear the difference. Any more than this normally becomes a "special effect." You don't want reverb to dominate the mix unless you are trying to create the effect of a band in a cave-which is a perfectly legitimate creative goal if that's the sort of thing you're aiming for.

The Modulation Effects: Phasing, Chorus, and Flanging
All of these effects work on basically the same principle: a portion of the audio signal is "time-shifted" and then mixed back with the direct signal. The amount of time shift is controlled, or "modulated", by an LFO (Low-frequency Oscillator). When we say "time shift," however, we're not talking in terms of minutes or even seconds.
For phasing effects the shift is very small indeed - a difference measured in degrees of phase shift rather than time units. The phase difference between the modulated and direct signals causes cancellation at some frequencies and reinforces the signal at others - a "comb filter" effect - and this causes the shimmering sound we hear. Phasing is the subtlest of all these effects, producing a gentle shimmer that can add life to a wide range of sources without being too obtrusive.
For chorus and flanging the signal is actually delayed by several milliseconds (a millisecond is a thousandth of a second), with the delay time modulated by an LFO, and recombined with the direct signal. In addition to the comb-filter effect described above, the delay modulation in these effects causes a perceived pitch shift which, when mixed with the direct signal, results in a harmonically rich swirling or swishing sound. The difference between chorus and flanging effects is primarily in the amount of delay time and feedback used - flanging uses longer delay times than chorus, whereas chorus generally uses a more complex delay structure. Chorus is most often used to thicken the sound of an instrument, while flanging is usually used as an outright "special effect" to produce other-worldly sonic swoops.

Compression
Have you ever wondered why professionally produced recordings sound so different from your own? There are numerous reasons, of course, but one important factor is the judicious use of compression.
One form of compression known as "limiting" can, when properly used, produce a smooth, unified sound with no excessive peaks or distortion. Compression can also be used within a mix to make a voice or instrument seem to come forward, or simply to even out level differences. Compression can be used to make a mix seem bigger and louder by producing a more "saturated" sound. Professional compressors have numerous parameters that need to be carefully adjusted: attack, release, threshold, level, and sometimes more. A professional sound engineer might need to spend a considerable amount of time, based on a considerable amount of experience, to set each of these parameters to achieve the desired sound.
The EMX compressor makes achieving great sound much easier. All you need to do is set a single "compression" control and all of the pertinent parameters are automatically adjusted for you.
The engineers who designed this fine compressor paid careful attention to achieving the best sound quality possible so that you can quickly achieve pro-quality compression without having to worry about a confusing multitude of settings.
A common example of the use of compression is to "tame" a vocal that has a wide dynamic range in order to tighten up the mix. With the right amount of compression you'll be able to clearly hear whispered passages while passionate shouts are still well balanced in the mix. Compression can also be valuable on bass guitar, producing a smooth bass sound that stays solid through the tune. Compression can also be applied to guitar tracks to add extra sustain. Too much compression can be a cause of feedback, however, so use it sparingly.

(Figure)
OUTPUT
0
(Min)
10
(Max)
INPUT

Music First-Then Mix
In any case, the music comes first. Think about the music and let it guide the mix, rather than trying to do things the other way around. What is the music saying and what instrument or technique is being used to drive the message? That's where the focus of your mix should be. You're using a high-tech tool to do the mixing, but the mix itself is as much art as the music. Approach it that way and your mixes will become a vital part of the music.


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