PA System

Mackie DL1608 - 16 Channel Digital Live Sound Mixer with I-Pad control.

2013-08-17T09:48:00.002-07:00

Proven Hardware That Sounds Great

    16 Onyx mic preamps
    High-end Cirrus Logic® converters
    Ultra low-noise, high-headroom design
    6 aux sends for monitor mixes
    Master L/R output for mains

Tons of Built-In Processing

    Choice of powerful, touch-sensitive plug-ins
    4-band EQ, gate and compression on inputs
    31-band GEQ and comp/limiter on outputs
    Global reverb and delay

Wireless Mixing

    Seamless wired to wireless mixing
    Tune the room from anywhere
    Get on stage to ring out monitors
    Personal monitor mixing with access control ability
    Use up to 10 wireless devices simultaneously

Total Control from Your iPad, iPhone or iPod touch

    Intuitive Master Fader™ app for all iPad models
    My Fader™ app for quick control from iPhone or iPod touch
    “Grow and Glow” visual feedback
    Preset and snapshot recall
    Record the mix to the iPad for instant sharing
    Integrate music from any app into the mix

Install Friendly Features

    PadLock™ feature locks down iPad for permanent installs
    Industry standard Kensington lock secures mixer
    Compact footprint saves precious workspace
        15.5″ x 11.5″ x 3.9″
        7.9 lbs / 3.6 kg
    30-pin connector version available now and Lightning version coming soon

Frequency for Instruments

2012-10-05T21:35:00.003-07:00

Magic Frequencies
Excerpted from The Mixing Engineer's Handbook
by Bobby Owsinski

Editors' Note: The following excerpt from "The Mixing Engineer's Handbook" by engineer Bobby Owsinski discusses how EQ can affect different frequencies in an audio mix. A Tips and Tricks section at the the end features various professional engineers offering their perspective on EQ techniques.

Before we examine some methods of equalizing, it’s important to note the areas of the audio band and what effect they have on what we hear. The audio band can effectively be broken down into six distinct ranges, each one having enormous impact on the total sound.

• Sub-Bass — The very low bass between 16Hz and 60Hz that encompasses sounds that are often felt more than heard, such as thunder in the distance. These frequencies give the music a sense of power even if they occur infrequently. Too much emphasis on this range makes the music sound muddy.

• Bass — The bass between 60Hz and 250Hz contains the fundamental notes of the rhythm section, so EQing this range can change the musical balance, making it fat or thin. Too much boost in this range can make the music sound boomy.

• Low Mids — The midrange between 250Hz and 2000Hz contains the low order harmonics of most musical instruments and can introduce a telephone-like quality to the music if boosted too much. Boosting the 500Hz to 1000Hz octave makes the instruments sound horn-like, while boosting the 1kHz to 2kHz octave makes them sound tinny. Excess output in this range can cause listening fatigue.
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• High Mids — The upper midrange between 2kHz and 4kHz can mask the important speech recognition sounds if boosted, introducing a lisping quality into a voice and making sounds formed with the lips such as “m,” “b” and “v” indistinguishable. Too much boost in this range — especially at 3kHz — can also cause listening fatigue. Dipping the 3kHz range on instrument backgrounds and slightly peaking 3kHz on vocals can make the vocals audible without having to decrease the instrumental level in mixes where the voice would otherwise seem buried.

• Presence — The presence range between 4kHz and 6kHz is responsible for the clarity and definition of voices and instruments. Boosting this range can make the music seem closer to the listener. Reducing the 5kHz content of a mix makes the sound more distant and transparent.

• Brilliance — The 6kHz to 16kHz range controls the brilliance and clarity of sounds. Too much emphasis in this range, however, can produce sibilance on the vocals.

Leo di Gar Kulka — “Equalization - The Highest, Most Sustained Expression of the Recordist’s Heart,” Recording Engineer/Producer, Vol. 3, Number 6, November/December, 1972

For those of you who have an easier time visualizing the audio spectrum in one-octave increments (like those found on a graphic equalizer), here’s an octave look at the same chart.		Easy-To-Remember Golden Rules Of EQ 1. If it sounds muddy, cut some at 250Hz. 2. If it sounds honky, cut some at 500Hz. 3. Cut if you’re trying to make things sound better. 4. Boost if you’re trying to make things sound different. 5. You can’t boost something that’s not there in the first place.
31Hz	Rumble, "chest"
31Hz	Rumble, "chest"
63 Hz	Bottom
125Hz	Boom, thump, warmth
250Hz	Fullness or mud
500Hz	Honk
1KHz	Whack
2KHz	Crunch
4KHz	Edge
8KHz	Sibilance, definition, "ouch!"
16 KHz	Air
16 KHz	Air

Figure 5 -- Frequency for Instruments--

Tricks and Tips
General Tips
Use a narrow Q (bandwidth) when cutting; use wide Q’s when boosting
If you want something to stick out, roll off the bottom; if you want it to blend in, roll off the top

For Snare —

To find the “point” on the snare, boost the upper midrange starting at about +5 or 6dB at 2kHz or so. Open up the bandwidth (if that parameter is available) until you get the snare to jump out, then tighten the bandwidth until you get only the part of the snare sound that you want most. Then fine-tune the frequency until you need the least amount of boost in order to make it jump out of the mix.

For Drums—

Dave Pensado: A lot of the music I do has samples in it and that gives the producer the luxury of pretty much getting the sound he wanted from the start. In the old days you always pulled out a little 400 on the kick drum. You always added a little 3 and 6 to the toms. That just doesn’t happen as much any more because when I get the tape, even with live bands, the producer’s already triggered the sound he wanted off the live performance and the drums are closer.

For Bass —

The ratio between the low bass (80–120Hz) and the mid-bass (130Hz–200Hz) is important. Try using two fairly narrow peaking bands, one at 100Hz and another at 140Hz and boost one and cut the other. If the bass is too warm, sometimes reducing the upper band can make it more distinct without removing the deeper fundamentals that live in the 100Hz band. Also, try boosting some of the 1kHz area since this is where a lot of the sound of the Fender bass lives.

For Fatter Guitars —

Boost midrange a lot (9dB or so) and sweep the frequencies until you hear the range where the guitar sounds thick but yet still bright enough to cut through. Now, back the boost down to about +4 or so until the guitar cuts through the mix without being too bright.

Don Smith: I use EQ different from some people. I don’t just use it to brighten or fatten something up; I use it to make an instrument feel better. Like on a guitar, making sure that all the strings on a guitar can be heard. Instead of just brightening up the high strings and adding mud to the low strings, I may look for a certain chord to hear more of the A string. If the D string is missing in a chord, I like to EQ and boost it way up to +8 or +10 and then just dial through the different frequencies until I hear what they’re doing to the guitar. So I’m trying to make things more balanced in the way they lay with other instruments.

For Vocals—

Boost a little at 125Hz to 250Hz to accentuate the voice fundamental and make it more “chesty”-sounding. The 2kHz to 4kHz range accentuates the consonants and makes the vocal seem closer to the listener.

Ed Seay: On a vocal sometimes I think, “Does this vocal need a diet plan? Does he need to lose some flab down there?” Or sometimes, “We need some weight on this guy so let’s add some 300 cycles and make him sound a little more important.”

David Sussman: If I’m recording vocals, I like to roll off quite a bit on the bottom end so the compressor doesn’t start kicking in and bringing up any low end rumble or noise. If I’m EQing a piano or something that’s already been recorded, I sometimes roll off a lot of the bottom so I leave a lot of room for the bass and the kick drum to occupy. A lot of times I don’t need anything under probably 100Hz. I’ll do some rolling off with the filters and then I may take a bell curve and zone in on a couple of other woofy areas on certain instruments.

Dave Pensado: I think of EQ as an effect much the same way you would add chorus or reverb to a particular instrument or vocal. Like, I might have a vocal where I think it’s really EQed nicely and then I’ll add a little more 3k just to get it to bite a little more. Then it just makes me feel like the singer was trying harder and it brings out a little bit of passion in his or her voice. So I tend to be most effective when I do the standard equalizing, then take it to the next level, thinking of it as an effect.
Source: ArtistPro

EON 210P - Portable PA Self-Powered 10” Two-Way system with detachable powered mixer

2012-10-03T08:51:00.001-07:00

The EON210P consists of two 10-inch, two-way, powered loudspeakers, one with a detachable powered mixer and the other with a detachable storage compartment, one pair of unshielded speaker cables and a power cord. Capable of reproducing full bandwidth sound at high levels the 210P is comprised of a 250 mm (10 in) woofer, a 37.5 mm (1.5 in) neodymium high frequency compression driver coupled to a 100° H by 60° V waveguide driven by a 300 watt Crown® Class-D power amplifier and integrated mixer.

Features :

300 watt High-Performance Powered Speaker System

Ergonomic soft touch top grip for easy handling

10” low-frequency driver with neodymium magnet for low-distortion and light weight

1” throat diameter next generation JBL neodymium compression driver

Efficient Crown® Class-D amplifier technology

100° H x 60° V asymmetrical wave guide for uniform audience coverage

8 channel powered mixer with intuitive interface

On board digital multi-effects

Integrated 36 mm pole mount socket with stabilizing securing screw

Highly designed composite enclosures for durability, lightweight, and acoustic performance

Specifications :

System Power Rating

300 Watts (2 X 150 Stereo)

Maximum SPL Output

124 dB peak system
output (pink noise)

Net Weight of System

33 lbs. (19 lbs. for unit
with powered mixer,
14 lbs. for unit with
storage pod)

System Type

Powered mixer with 2 two-way bass-reflex enclosures

Frequency Range (-10 dB)

60 Hz - 20 kHz

Frequency Response (±3 dB)

75 Hz - 19 kHz

AC input

120 - 240 V 50/60 Hz, voltage selector switch sets operational range

Amplifier Design

Crown® Class D

Output Connector

2 X 1/4” TS (unbalanced) amplifier outputs

Input Connectors

8 inputs (4 Mono Mic/Line, 2 X Stereo)

Channels

1-4 are XLR / 1/4” jack combo connectors, XLR is a mic level input, 1/4” is a line level input. 5-6, one pair of 1/4” balanced TRS jacks (stereo), and a pair of RCA jacks (stereo). 7-8 is a 3.5 mm stereo jack

Input Impedance

Ch 1-4 Combo: XLR 3 K Ohms Balanced
Ch 1-4 Combo: TRS 20 K Ohms Balanced
Ch 5-6: TRS 25 K Ohms Balanced
Ch 5-6: RCA 12 K Ohms Unbalanced
Ch 7-8: 3.5 mm 30 K Ohms Unbalanced

Phantom Power

30 V

Output Connectors

Monitor outputs: one pair of 1/4” balanced TRS jacks (stereo), and a pair of RCA jacks (stereo)
Headphone output: one 3.5 mm stereo jack

Signal Indicators
Main Output LED Ladder

Limit: Red LED indicates limiter active condition
Signal: Green LED indicates signal present

Signal Indicators: Ch 1-4

Peak LED: Green = signal present, Red = input overload

Individual channel Bass and Treble controls, center detent, +/-6 dB cut and boost

System Limiter

On Board DSP Limiting and Tuning

LF Driver

1 x JBL 328 H 250 mm (10 in) woofer

HF Driver

1 x JBL 2414H-1 37.5 mm (1.5” ) annular polymer diaphragm, neodymium
compression driver

Coverage Pattern

100º x 60º nominal

Crossover Frequency

2 kHz

Crossover Type

Passive network, 2nd order filters (-12 dB per octave) for high pass and low pass filters

Enclosure

Polypropylene

Suspension/ Mounting

36 mm pole socket with stabilizing screw

Handles

One on top

Grille

Powder coated perforated steel

Shipping Package (HxWxD)

21.5” x 26” x 13.75”
(546 mm x 660 mm x 349 mm)

Applications :

Live sound reinforcement, speech and vocals, music playback in entertainment, A/V, and institutional venues – especially when ease of use and portability are important factors.

Amplification, mixing, and monitoring for electronic musical instruments.

Everywhere you need to be heard.

- Source (http://www.jblpro.com/)

STAGEPAS 300 - Portable PA System

2012-10-03T08:41:00.003-07:00

Live Sound Has Never Been Easier

This extraordinarily portable PA system has 8 channel (including 4 microphone) inputs and delivers a very respectable 300 watts of high-quality power to the supplied pair of compact speakers.

No matter where your music leads, Yamaha's new STAGEPAS™ 300 Portable PA system has the power and portability to go with you. It has all the power and high-quality sound you need to fill a small performance space, yet it is small and light enough to take wherever it is needed. Combining a built-in powered mixer and PA speakers, it's incredibly easy to use - you can be set up and playing in a matter of minutes! STAGEPAS™ 300 is the perfect road companion, providing powerful, reliable, convenient sound reinforcement for a wide variety of applications. Great sound has never been so portable - and so easy to use. High quality, convenience, and power to go!

Features

STAGEPAS300: On Stage Anywhere, Anytime

STAGEPAS300 is a portable PA system that can be set up to provide high quality live sound anywhere, anytime. Designed primarily for musicians on the move, STAGEPAS™ 300 makes it possible to deliver high-quality live sound in just about any venue or environment with maximum handling ease and minimum hassle.

Speakers, Powered Mixer, and Cables

This all-in-one system provides a pair of passive speakers, a detachable powered mixer, and a pair of speaker cables - everything you need except for your sources. The whole system weighs just 18 kilograms!

Detachable 8-channel Powered mixer with 150W + 150W Output

The powered mixer is normally stored in the rear of one of the speakers. The STAGEPAS™ powered mixer can be detached (the only tool you need is a coin) for convenient operation. The power amplifier is a class-D* type that delivers a solid 150 watts + 150 watts from a remarkably compact unit. The mixer has a total of eight input channels: four mono microphone/line inputs and two stereo line inputs. In addition to speaker outputs for the supplied speakers, the STAGEPAS™ mixer has line outputs that can be used to connect additional powered speakers for monitoring, or to send the mixer's output to a recording device. * Class-D power amplifiers provide high-efficiency amplification with low power consumption and minimal heat generation, allowing the amplifier to be built into a smaller, lighter, and cooler package.

Two-band EQ and Reverb Built In

Each mixer channel has an easy-to-use two-band equalizer that can help to shape the sound of each channel to achieve the best possible mix. Boost the highs to give a vocal channel more air and clarity, or boost the lows on a guitar channel for more punch. The four mono input channels have reverb on/off switches, and a reverb level control lets you add just the right amount of reverb to bring your sound to life.

For Singers & Instrumentalists

The built-in high-quality reverb is indispensable for great vocal sound, and it can add a lush dimension to the sound of electric-acoustic guitars as well.

For Speech

The STAGEPAS™ mixer has a "speech" mode that optimizes the system settings for maximum clarity with speech. Of course, background music from a CD player or other source can be mixed in with the spoken performance.

For Bands

Multiple microphone inputs make the STAGEPAS™ 300 system a good choice for band rehearsals or performances. MSR100 powered monitors (sold separately) can be added for monitoring.

Optional Mic Stand Adaptor

The optional BMS-10A Mic Stand Adaptor allows the STAGEPAS™ mixer unit to be mounted on a standard straight microphone stand. Mic stand mounting is an ideal way to position the mixer for easy control access while performing.

- Source (http://in.yamaha.com)

STAGEPAS 250M - Portable PA system

2012-10-03T08:35:00.005-07:00

STAGEPAS 150M and 250M can function either as an ultra-portable PA system, or as a powerful keyboard amplifier with sound quality and portability that far surpasses any other conventional keyboard amp.

Combining high-performance power and accurate sound reproduction with remarkable versatility in a portable package, Yamaha's STAGEPAS Series Portable PA Systems are fast becoming the standard for musicians and other PA users on the go.These combination mixer/amplifier/speaker systems are exceptionally lightweight, portable and full-featured. Since virtually everything you need is in one, easy-to-use package, you can be set up and playing within a matter of minutes. And when you're done, you can be packed and back on the road again with the same ease and speed. Yet despite the simplicity and compact size, these systems cut no corners in sound quality or versatility, and in fact give you some advanced features not normally found in systems of this class.The series now includes the new single-speaker STAGEPAS 150M and 250M, which can be expanded to a stereo system by simply adding a second speaker. They can also function either as an ultra-portable PA system, or as a powerful keyboard amplifier-with sound quality and portability that far surpasses any other conventional keyboard amp. No matter what system you choose, you have a high-power, highperformance sound system ready for a wide variety of venues and events, both indoors and out.

Ideal for Monitor Speaker Use

The STAGEPAS 250M and 150M are effective as a monitor speaker system (especially for keyboard players, drummers and solo performers), thanks to its fullrange reproduction, convenient mixing controls, and versatile speaker placement.

* For Keyboard Players
Keyboard players needing a simple, yet exceptionally high-quality keyboard amplifier will find ideal solutions in the STAGEPAS 150M and 250M. These compact, highly portable and easy-to-use systems can serve as your main system for small gigs, or as a sub-mixer for direct connection to the main mixer at large events.

* For Guitarists
The STAGEPAS 150M and 250M are excellent systems for guitarists playing in small venues, since they deliver outstanding sound for the house as well as great monitor sound for the player.

-Source (http://in.yamaha.com)

Walkabout - Portable PA System - Studiomaster

2012-10-03T08:06:00.002-07:00

The need for compact, portable PA systems has never been greater. Light weight, powerful, easy to set up but above all easy to use systems are compulsory. The Walkabout has perfectly addressed these requirements in a functional yet stylish package.

The Walkabout features a total of 300 Watts of clear audio power, 5 channel mixer (3x mic/line, 1x mic/stereo and 1x stereo), 16 program DSP and 5 band graphic equaliser. Sockets are also provided you record your performance.

It contained in its own carry case and include speaker stands, microphone and all cables. Once packed the Walkabout has a retractable handle and recessed casters making light work when moving.

Equalisation	HI (12kHz), LO (80Hz)
Graphic Equaliser	125Hz, 400Hz, 1kHz, 4kHz, 8kHz
Power	2 x 150W
Speakers	2 way
Connections Channels 1-3: Channel 4/5: Channel 6/7:	Combination XLR mic & TRS jack line XLR mic and stereo line (2) TS jacks Stereo line RCA phono
Record Output	RCA phono
Speaker Outputs	TS jacks
Speaker Input	Single TS jack
Power Requirements Voltage Selector	230V Range 50/60Hz 115V Range 50/60Hz

Source : (http://www.studiomaster.com)

PA monitors Stage Setup - 3D

2012-09-26T01:26:00.000-07:00

Stage Setup for Musicians

2012-09-26T01:05:00.000-07:00

Balanced Wiring & its Advantages

2012-09-12T09:03:00.003-07:00

Professional equipment solves this problem by using two closely-spaced conductors twisted together. Audio is balanced equally on these wires, flowing in a positive direction on one wire while in a negative direction on the other. Equipment looks at the voltage difference between those wires, and ignores everything else. A grounded shield is still used to prevent high-frequency noise, and it might form a hum-gathering antenna -- particularly if there are other ground connections. But since the ground isn't part of the audio path, nobody cares.

A balanced shielded cable.

Most modern circults do not use balanced wiring internally. The internal wires or printed-circuit traces are so short that noise pickup isn't a problem. But they balance the signal before it leaves, and unbalance any incoming ones. This is very easy to do with op amps or transformers. Balanced wires also reject noise that isn't coming from a ground loop. The two conductors are twisted closely together, so any interference radiated into the cable is picked up equally by both. But remember: the equipment is looking for a voltage difference between those wires. Noise is the same on both wires, so the equipment can't hear it.

Another advantage of balanced wiring

If a single-conductor shielded cable acts as an antenna, why doesn't two-conductor balanced wiring act as a double antenna? Answer: it does. Noises from nearby video or computer cables are picked up on each conductor. But remember, a balanced audio input cares only about the voltage difference between the two wires. Interference is radiated equally into each wire. Since the interference is equal on each, there's no voltage difference from it! The balanced input can't even see that the noise is there.
Or to put it into a chart:

Conductor           Audio Signal    Noise     Total on wire
    Black               +1 v          +1 v     +2 v
    White               -1 v          +1 v      0 v

Transmitted difference   2 v
                           Received difference  2 v

This noise-immunity of balanced wiring is why it's also used for high-speed computer networks. Category-5 cable contains four tightly-balanced pairs of wires. In fact, if your balanced input and output circuits are good enough, you can use Cat-5 cable for professional audio wiring! "Star Quad" is four-conductor shielded balanced cable. The four wires form a tighter, more consistent pack than two wires can and can resist even more noise. If you're using Star Quad, you must tie the two pairs of similarly-colored wires together at each end... reducing it effectively to two conductors. Don't try to use it as two balanced pairs for two different signals: this won't give you any noise-reduction benefits at all.

- ( Source : http://www.dplay.com)

About ground loops

2012-09-12T08:58:00.006-07:00

People hum when they don't know the words. Audio circuits hum when they don't know what silence should sound like. The sensitive circuits that boost your camera's audio before it's recorded, or shuttle sound around your editing suite, need a reference they can be sure is zero volts. They compare the input signal to this reference, amplify or proces the difference, and generate an output voltage that's also compared to the reference. Designers designate one point within a piece of equipment (often connected to the chassis or grounding pin of the power plug) and call it "ground": all voltages inside the equipment are measured with respect to it.
That's fine for a single piece of equipment, but when you hook two devices together, both have to agree on the reference. Since the cable shield has to be grounded at least at one end, the usual scheme is to use it to connect the two devices' reference points together. It works in very simple systems.
But remember, that shield is picking up hum from the building wiring. And if the shield is carrying current -- something unavoidable if it's part of the audio path -- it has a slight voltage drop. Both these factors mean that the two devices are going to have slightly different references, and the difference is constantly varying. The input circuit can't tell that this variation isn't part of the signal, so it amplifies it. Again, in a simple NLE with short wires this interference may be tolerable. But in a complex room or studio shoot, it becomes hum and noise.

It's called "60 Hz hum", but it's not just 60 Hz

When power-line frequencies leak into an audio circuit, they generate harmonics. The 60 Hz base signal also hums at 120 Hz, 240 Hz, and up the band. That's why filters don't do a good job removing hum... you have to fix it at the source

Complex setups have other problems as well. If there are multiple ground paths, they combine to make a very efficient loop antenna for the 60 Hz noise. These "ground loops" are almost impossible to predict, since you don't know the internal details of your equipment, and can crop up even in very simple setups... particularly if both pieces of audio equipment also share a ground connection through their power plugs' grounding pins. In a practical video studio, the situation is apt to be far worse: the non-audio cables -- RS-232 and RS-422 control, video wires, and even cable TV -- all have their own grounds.
- (Source : http://www.dplay.com)

Why wires are noisy?

2012-09-12T08:38:00.006-07:00

Why wires are noisy

If your tracks are plagued by noise and low-frequency hum, it's probably because of a wiring problem. But the solution is simple, relatively inexpensive, and has been around for a hundred years. Balanced audio wiring was first used by the phone company, to send calls over hundreds of miles of low-quality wire without picking up too much noise. Today it's used by every audio professional for much the same reason. The problem is that wires are also antennas. When you plug a mic into a camera, or a DAT player into your NLE, you don't just get the desired signal. Any nearby electric fields are also picked up on the wire, adding a slight voltage which the equipment can't distinguish from the desired audio.
You can't avoid these fields. They're created by any other wires that carry a current. This includes video, timecode, and data cables, which can add all sorts of high-frequency whines and whistles to your track.
Manufacturers reduce this antenna effect in a signal wire by wrapping a shield around it, usually a copper braid or metal foil. The shield is connected to ground and shorts out the interference before it can reach the signal wire in the center of the cable. That's why phono, BNC, and cable TV plugs have a center pin and outer metal shell: the pin is signal, and the shell carries the shielding that in the wire.

A typical shielded cable.

Bah, Hum.

Of course the biggest currents in most places are in the wires that supply electricity to lights and and wall outlets. They radiate a lot. Cable shields aren't very effective with this 60 Hertz interference from the power-line frequency. Fortunately, it takes a much longer antenna to pick it up than the higher-frequency interference from video or timecode cables. In a small editing setup with short wires, the amount of 60 Hz pickup is very little compared to the audio voltages, so interference is minimal. But on shoot with long microphone cables, or in a complex post-production setup (my studio has about 3500' of analog wiring), the hum can be a major problem. Furthermore, the same shielding that protects against high-frequency noise can contribute to hum.

-Source (www.dplay.com)

3 Pin XLR Wiring Diagram

2012-09-11T20:12:00.004-07:00

Okay, this may be really basic, but for some of you it may just be what you were looking for. Here is the basic wiring diagram for a standard 3 pin XLR connector, used in audio for mics, playback machines, intercom, etc.

Cable designed for being cut into standard mic cables may have 2 pairs of wire and a shield around the outside, in that case pair the colors together and make sure they go to the same pin number on each end. The surrounding shield should be soldered to pin 1. Balanced audio cable designed for installs usually has a black, white, and ground wire. The uninsulated ground wire should go to pin 1, the red wire to pin 2, and the black wire to pin 3.

-Source (http://www.toffer.com)

Mixer Connections & Operation

2012-08-31T02:58:00.003-07:00

A detailed overview for beginners covering commonly asked questions on mixer operation.

Many people are confused by mixers because they are complex devices.
In this article I’ll attempt to answer some beginners’ questions on mixer operation:
• How do I hook up a mixer to the rest of the system? What jacks are best to use?
• How do I use graphic equalizers?
• What are compressors used for?
• How do I use groups?
• How do I set up monitor mixes?
• How do I set up the mixer to add effects?
• What’s a good resource for understanding mixers?
What jacks should I use to connect the mixer to my sound system?

Figure 1. Mixer connections.

• Connect each mic to the stage box (snake).
• Connect each snake XLR connector to each mic input XLR connector.
• Connect the mixer master or main output to your graphic equalizer input, and connect the graphic output to your house power-amp input. If you are not using a graphic equalizer for the house speakers, connect the mixer master outputs to the inputs of the power amp that drives the house speakers.
• If you are recording a board mix of the service or show, connect the mixer REC OUT or TAPE OUT connectors to the recorder line inputs.
Later in this article we’ll cover connections for compressors, the monitor system and effects devices.
Why would I put a graphic equalizer between the mixer and power amps? Isn’t that what the mixer EQ is for?
Mixer EQ affects the sound of each individual instrument and voice, while the graphic EQ affects the sound of the complete mix.
The graphic equalizer is used the flatten the frequency response of the house speakers and room so that the entire sound system is accurate or hi-fi.
One way to set a graphic EQ is to play some reference CDs alternately through high-quality headphones and through the house loudspeakers.
Adjust the graphic-EQ sliders to make the loudspeakers sound like the headphones in their bass-midrange-treble balance.
Here’s another way to set a graphic equalizer.
1. Obtain a measurement microphone, which is an omnidirectional condenser mic with a flat frequency reponse. Put the mic in the center of the audience area.
2. Plug the mic into a real-time analyzer (RTA) set to display 1/3 octave bands.
3. Play pink noise through one set of house loudspeakers (one combination of woofer, midrange and tweeter drivers).
4. On the graphic equalizer, pull down the frequencies that are the highest on the RTA display.
5. Try to get a flat spectrum (equal level in each frequency band) up to 1 kHz, then let the spectrum roll off gradually to about 10 dB down at 10 kHz. This is called a “house curve”.

It’s also common to use a graphic EQ between the mixer’s monitor send (aux out) jack and the power amp that drives the monitor speakers (see Fig. 1).
That EQ is used to reduce the levels of frequencies that feed back. You also can use the graphic EQ to reduce the bassy sound in the monitors caused by microphone proximity effect (the bass boost that occurs when directional mics are used up close).
The monitor signal from the board is pre-EQ, so turning down the bass (low frequencies) on the mic channel does not turn down the bass in the monitor speakers.
That’s where a graphic EQ can help: turn down frequencies a few dB below 200 Hz or so. Then the monitor speakers won’t sound too bassy and muddy.
What’s a compressor for? How do I connect it to a mixer?
A compressor is used to reduce the dynamic range of whatever signal you pass through it. For example, a lead vocalist might suddenly sing a very loud note, blasting the listeners.
The compressor is an automatic volume control - it turns down loud notes so they don’t get too loud. If this isn’t a problem in your venue, you don’t need a compressor.
You insert a compressor in-line with one of the mic channels (see Fig. 1). Find the mic channel on the back of the mixer, and connect its insert send to the compressor input. Connect the compressor output to the insert return on the same mixer channel.
If there is only one insert jack per channel, the tip of the jack is send and the ring of the jack is return, so use a stereo phone plug at the mixer going into two plugs (in and out) at the compressor.
Would I use grouping to combine several channels into one—say, for a monitor for just the vocalists?
The groups are for the house speakers, not the monitor speakers. You might assign all the vocal mics to Group 1 (also called Subgroup 1 or Submix 1). ‘
Then you can control the overall level of the vocals with just the Group 1 fader. Start with the group fader and master fader about 3/4 up (at unity gain, or 0 dB).
You don’t have to use groups, but some people find it convenient.
If you don’t use groups, just assign each mic channel to the stereo mix bus (the master stereo output of the console), and turn down all the group faders because they are not being used.
To confuse things, some consoles use Group 1 and Group 2 as the main stereo output channels. Other consoles have groups plus a separate stereo master output channel.
How do I set up monitor mixes?
The aux knobs in your mixer can be used either for monitor mixes or for controlling the amount of effects on each input channel. First decide which aux channel you want to use for a monitor mix.
You might use several aux channels (aux 1, aux 2, aux 3) to create separate monitor mixes for different performers. Each aux number is a separate monitor mix, feeding a separate monitor power-amp channel, feeding a separate monitor speaker.
Let’s start with just one monitor mix.
Suppose that you’ll create a monitor mix with all the aux 1 knobs. On the back of your mixer, connect the aux 1 send connector to the graphic equalizer (if any) used for the monitor speakers, and connect the graphic equalizer output to your monitor power-amp input (see Fig. 1).
If you are not using a graphic EQ with your monitor speakers, connect the aux 1 send to the monitor power-amp input.
Set all the monitor aux knobs to pre-fader so that the fader for each channel does not affect the monitor level.
What if you need several different monitor mixes? You might use all the aux 1 knobs to set up a monitor mix for the vocalists. Connect aux 1 out to the power-amp channel for the vocalists’ monitor speakers.
Then use all the aux 2 knobs to set up a monitor mix for the drummer. Connect aux 2 out to the power-amp channel for the drummer’s monitor speaker. Use aux 3 for the piano player, and so on.
For example, let’s say the vocalists need to hear only the piano and vocals in their monitor speakers. You would use all the aux 1 knobs across the console to set up a monitor mix for the vocalists. Turn up the piano channel’s aux 1 knob about halfway.
Turn up the vocal channels’ aux 1 knobs about halfway. Turn up the aux 1 master knob (if any) about halfway. Make sure the vocalists can hear the monitor mix, and adjust it according to what they want. Turn up the aux knobs slowly and stay below the feedback point.
Similarly, suppose the drummer needs to hear only the piano and bass. You might use all the aux 2 knobs across the console to set up a monitor mix for the drummer. Turn up the piano channel’s aux 2 knob about halfway.
Turn up the bass channel’s aux 2 knob about halfway. Turn up the aux 2 master knob (if any) about halfway. Make sure the drummer can hear the monitor mix, and adjust it according to what the drummer wants.
How do I set up the mixer to add effects?
As we said earlier, the aux knobs in your mixer can be used either for monitor mixes or for controlling the amount of effects on each input channel. First decide which aux channel you want to use for effects.
Suppose aux 4 is your effects channel On the back of your mixer, connect the aux 4 send connector to the input of your effects device. Connect the output of the effects device to the Bus In or Effects Return connector on your mixer (see Fig. 1).
Another option is to connect the effects outputs to the line inputs of two extra input channel strips on your mixer, and have those be the effects-return level controls.
Set the effects-send (aux 4) knobs to post-fader so that the fader level also controls the amount of effects. Set the dry/wet mix control on the effects unit all the way to wet (100% effect).
For each input channel (vocal or instrument), use the aux 4 knob to set the amount of effects you want to hear on that vocal or instrument.
Note that some mixers have effects built in so you don’t need to make any effects connections.
- Source ( prosoundweb.com)

Setting Mic For Drums

2012-08-31T00:58:00.000-07:00

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. 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 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.

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 , and the 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 gives you a nice punchy sound.

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.

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 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 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 . 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.

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

- Source (dummies.com)

Large stereo tri-amped PA system.

2012-08-28T00:31:00.000-07:00

With the exception of the compressors, the additions incorporated into this system are simply a doubling of the system components covered in previous examples. At this point, you should have a pretty good grasp of how to hook it all together, so rather than listing a lengthy step by step, I have listed a brief description of each addition included in this example.

Two Monitor Mixes
Running more than one monitor mix can be very useful in that you can provide different monitor mixes for different parts of the stage. I have found that the drummer often wants to hear different things in the mix than the rest of the band. With two different monitor mixes, this is easily accomplished. Simply assign one monitor mix (channel A) to the drummer and another (channel B) to everyone else. This way you can adjust what the drummer hears independently of what everyone else on stage hears. In order to do this, your soundboard needs to be equipped with more than one monitor channel. These channels, usually designated as "Monitor A" and "Monitor B", will be controlled by separate knobs and will have separate outputs which must in turn be hooked into separate equalizers, amplifiers, and speakers.

Multiple Effects Loops
The same concept applies to the effects loops. To hook up two effects loops in a mixer that is equipped for it, all you have to do is run one loop through "Effects A" and another separate loop through "Effects B". Some boards come equipped with several different available effects loops that may be labeled "Effects 1", "Effects 2",etc. Sometimes the effects will be labeled as "auxiliary". You can run as many separate effects loops as your mixer is equipped to handle provided that you have enough separate effects units to pull it off. One possible use for this set up is that you could assign nothing but a long delay (echo) to Aux 2 and your general effects to Aux 1. Then when a song required a long echo on a certain part, all you would have to do is turn up the slider or knob for Aux 2 to get your desired echo without changing to rest of the effects at the same time. Then when the echo wasn't needed anymore, you could simply turn the Aux 2 all the way down effectively removing that effect from the mix.

Note:
The effects send, monitor out, and auxiliary out channels are essentially nothing more than specifically labeled line out channels. This means that as long as you pay close attention to where you plugged things in and you properly re-label your knobs, you can use them interchangeably. Usually, the only reason to do this would be to acquire another monitor mix in a board that is equipped with only one monitor channel but has an extra unused auxiliary channel. To avoid confusion, I would recommend doing this only as a last resort. Also, don't forget that in order for your effects to work, they must return to the soundboard to complete the effects loop.

Mains in Stereo
Running two separate channels for the mains is what is known as running a stereo PA system. To do this, you need a stereo mixing board. This board will consist of two output channels for the mains. These will be labeled either "left" and "right" or "A" and "B". The sliders for each input channel on the board will control both channels simultaneously, but there will be a "pan" knob above each that will allow you to pan the volume from left to right just like the "balance" knob on your car stereo. The output sliders on the board will operate the outputs for channel "A" and "B" independently of one another. Hooking this system up is essentially the same as what you did when hooking up two monitor mixes. Simply run the left channel "A" out to it's own equalizer, crossover, amplifiers, and speakers, and then do the same for the right channel "B". This is where those stereo (two channel) components come in handy. For instance, you can hook the "left out" from the board into the channel "A" input of the equalizer and hook the "right out" from the board into the channel "B" input of the same stereo equalizer. This principal can be followed all the way through the crossover, and the amplifiers.

Note:
When using a stereo amplifier in this way, make sure the switch in the back is switched to "stereo" mode.

Compressors
Compressors are an effect that is usually run in-line in the main signal path. They make a subtle change in the sound of the entire system that amounts to "taking the edge off". There are a lot of technical descriptions for what they do, but my best description is to say that they do just what the name implies. They squash (or compress) the sound together to create a more compact and clean sound. Be careful with these, they can be used for either good or evil. Set to a moderate level, they can add to the quality of your overall sound, but set too high, they can take the life right out of your performance. Honestly, I am not the biggest fan of compressors, but I think I am in the minority, so I felt I should incorporate them into at least one example.

The Snake

At some point you may want to use a soundman to run sound from somewhere other than the stage. In order to get the board out in front of the stage and across the dance floor, you will need a PA snake.

The PA snake is used like an extension cord that connects all of the things on and behind the stage to all the things the sound man will be using in the sound booth on the other side of the room. The snake should have at least as many low impedance (low Z) channels as your soundboard, and at least 2 to 4 high impedance (high Z) channels depending on whether you are running a stereo setup or not. When setting your system up this way, the soundboard should be placed at least 30 feet or farther out in front of the main speakers so that the sound engineer will be far enough away to get a clear idea of what is coming out of the system. Also, the equalizers, the compressors, and the effects should be located along side the soundboard while the amplifiers and the crossovers should be located on or behind the stage. The on stage end of the snake will be a big box with individually numbered High Z inputs and Low Z outputs. The end of the snake toward the sound booth will have high Z outputs and Low Z inputs with numbers corresponding the inputs and outputs on the stage end. The ends toward the sound booth will be loose and look like the ends of microphone or instrument cords.

To properly hook up the snake, simply plug each of the loose low Z ends into the channel on the soundboard that corresponds with the number printed on that end, then plug all of your microphones on stage into their regularly assigned channels at the box end of the snake. For instance, the loose end marked "1" should be plugged into input channel number 1 of the soundboard. Then, on the stage you can plug microphone number 1 into the number 1 input on the box end of the snake thus assigning microphone number 1 to channel number 1 of the soundboard.

The high Z inputs and outputs are there to provide an extension between the compressors (if you're using them. If not, insert the word EQ for the word "compressor") at the sound booth and the crossovers on the stage. To do this, simply plug one of the high Z loose ends of the snake into the output of the compressor, and then plug a cable between the corresponding high Z channel on the box end of the snake and the input of the crossover located on the stage. This same principle applies to both the monitor channels and the main channels.
- Source ( thefxcode.com)

Crossovers

2012-08-27T23:53:00.002-07:00

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

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

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

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

Tri-Amping the Mains

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

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

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

Monitors and Effects

Connect these together as described in example.

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

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

Lows

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

Mids

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

Highs

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

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

Source (thefxcode.com)

Small Practical PA

2012-08-27T22:50:00.005-07:00

Small Practical PA

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

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

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

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

Mains

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

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

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

Effects loop

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

Multiple effects

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

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

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

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

- Source (thefxcode.com)

2012-08-27T22:31:00.004-07:00

Noise Gates

A Guide to Noise Gates in PA Systems

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

What it is

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

What it does

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

How it works

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

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

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

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

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

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

How do you use it?

If all else fails, read the manual!

Noise gates are usually connected on channel inserts.

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

To use a noise gate on a drum:

1. Set the channel gain.

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

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

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

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

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

Other than on drums, leave well alone.

Do you need one?

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

What sort do you need?

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

- Source (astralsound.com)

Dynamics Processors : Compressors / limiters

2012-08-27T22:18:00.001-07:00

2.1. Introduction

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

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

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

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

2.2. Controls

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

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

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

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

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

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

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

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

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

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

2.3. Meters

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

2.4. Side Chain

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

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

2.5. Setting a compressor depending on the application

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

Dynamics Processors. A tutorial

2012-08-27T22:06:00.002-07:00

1.1. Introduction

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

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

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

1.2. Defining dynamics processing

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

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

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

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

1.3. Types

The more commonly used dynamics processors are :

A compressor / limiter attenuates or limits signals exceeding a pre-defined signal level. There exists an special version of a compressor/limiter called "de-esser", which tames excessive levels of the portion of the frequency spectrum where sibilance occurs.
A limiter is only a form of compressor.

A noise gate mutes or attenuates signals below a pre-defined signal level. If it allows the selection of the attenuation level (as opposed to just providing total attenuation, i.e. muting), it is referred to as a "downward expander".

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

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

1.4. Controls

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

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

1.5. Meters and indicators

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

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

2012-08-27T22:01:00.001-07:00

Setting Up A PA System

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

The Bare Bones

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

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

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

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

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

The Essentials

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

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

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

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

- Source (thefxcode.com)

How to EQ a Room

2012-08-27T21:41:00.003-07:00

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

Source (Wikihow)

Graphic EQ

2012-08-27T21:38:00.004-07:00

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

What it is

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

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

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

What it does

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

How it works

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

How do you use it?

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

Do you need one?

Almost always.

What sort do you need?

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

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

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

- Source (astralsound.com)

2012-08-27T05:09:00.002-07:00

Basic Audio Setup for Office

2012-08-27T05:09:00.001-07:00