The precedent effect states a few things that are important to us in the audio reinforcement realm.

http://en.wikipedia.org/wiki/Precedence_effect

“The precedence effect or law of the first wavefront is a binaural psychoacoustic effect. When a sound is followed by another sound separated by a sufficiently short time delay (below the listener’s echo threshold), listeners perceive a single fused auditory image; its perceived spatial location is dominated by the location of the first-arriving sound (the first wave front). The lagging sound also affects the perceived location. However, its effect is suppressed by the first-arriving sound.”

In a nutshell, when two signals arrive together they act like a single signal. When one arrives first it’s perceived as the origin of the signal even if the secondary signal is louder.

This has many important implications for live sound.

IF the PA is in front of the band & IF the PA signal arrives before the direct signal coming from the stage (guitar amps, drums, bass amp, etc…) the origin of the total sound will appear to come from the PA.

IF we delay the PA (using DSP) back to the physical backline or further, the sound origin appears to come from the stage instruments.

I have seen suggestions that we should delay the PA to the drum kit, guitar amps, etc…

Since a drum kit is a 3 dimensional instrument (has height, width & depth) * a guitar amp will typically have a single dimension, I would be inclined to delay to the middle of the drum kit.

How?

The best way is to place a speaker at the center of the stage on the line at which you want to delay too. Use that as your (0ms) measurement & then delay the PA until it matches. It’s amazing what a difference doing this simple delay procedure makes in focusing the audiences attention to the band & not the PA.

Here is the basic procedure:

Place reference speaker at center stage
Measure pink noise through reference speaker with mic placed out in the house & note delay time
mute reference speaker
Send pink noise thru PA & measure (same mic position) and note delay time
Compare reference speaker delay time with main PA delay time
Delay main PA until it matches delay time of reference speaker

If you can’t measure, don’t have a reference speaker or simply don’t have time, you can guess how far back the reference line is and delay accordingly. If when you look up & listen to the band @ sound check, you feel the main PA is either too late or still too early, adjust & listen again.

Keep in mind that in this day and age of digital consoles & digital signal processing, there is an inherent amount of latency by default. Typically your reference speaker wouldn’t be fed a signal from the main PA DSP so there is latency involved with the main PA that might not be involved with the reference speaker signal. For example, if you fed the reference speaker off an aux or matrix that bypasses the main PA DSP device. In that case, one system has more inherent delay than the other. Something to keep in mind.

Due to digital signal latency, you may find that the main PA is already arriving late enough that there is no need to add further delay to the signal to align it with the reference target point. If so, fine.

If a signal passes through enough digital processors in route to the speakers, you could easily find that your PA is significantly behind the band’s backline gear already. If so, there isn’t much you can do but physically move the band or the PA.

DI (direct input / direct injection) boxes are not created equal. Most cheap DI boxes are passive (no electronics) and use relatively poor quality transformers.

In some cases using a cheap passive DI may be fine but in some cases you will need something better.

Let’s split DI boxes up into a few categories.

Inexpensive passive
Expensive passive
Inexpensive active
Expensive active
Inexpensive tube based
Expensive tube based

Let’s draw the line between expensive & inexpensive at $100. That should separate the mice from the men.

Ignoring tube based DI units,

Here are my favorite DI options for live work.

Radial Engineering -JDI
http://www.radialeng.com/jdi.php

Radial Engineering – J48
http://www.radialeng.com/j48.php

Radial Engineering –

If you want to get one guitar amp to sound like another, one of the measurements you can make is to send pink noise to each amp & measure the speaker output via your measurement mic.

The procedure would be as follows:

Play your guitar through amp A & adjust until you like the sound
Send pink noise to the input of amp A & capture the frequency & phase response

Send pink noise to the input of amp B & capture the frequency & phase response

Now adjust the EQ & gain settings until the measurement matches that of amp A.

It’s possible that the EQ adjustment of amp B won’t allow for a complete match.
It’s possible that the two amps will be different enough in tonal flexibility that they won’t match.
It’s possible that the two amps will have different distortion characteristics in which case they won’t match.

Regardless you should be able to get as close as possible by measuring the two amps.

While you’re setup to measure, it would be wise to take measurements with all the EQ parameters at their minimum & maximum settings & make captures of each one by itself. Then take the same measurements & make the same captures using different minimum & maximum settings between EQ bands. This way you will know exactly what knob is adjusting what range of frequencies & how they react to each other.

Engineers rely on studio monitors for accurately representing their mixes for the general public.

Things to consider when placing studio monitors?

1. secondary reflections off the walls, ceiling & floor.
2. secondary reflections caused by your furniture (console / desk / etc…)
3. locating monitors up against a wall or in a corner will add low end
4. locating monitors in non symmetrical locations (one in a corner / one along a wall) will create anamolies in the stereo image.

When it comes to accurate reproduction of an audio signal thru the air, reflections are bad. When it comes to making mixing decisions, the better the phase & frequency response of the monitoring system, the better one knows they can trust their mixing decisions will translate universally to the average playback system (car stereo, home stereo, Ipod, Iphone, etc…)

One thing is assured. Any speaker placed in a acoustic space will interact with the space. At worst, this interaction will cause frequency imbalances & reflections that smear the time domain information. At best, the engineer will still have to pay close attention to placement, height & reflections. In live audio it is now common practice to have a multichannel DSP for system processing & frequency response correction. It only seems logical that the same should be true for studio reference monitors but I am unaware of this as a generally accepted practice. Regardless, I would suggest that studio reference monitors should be corrected with DSP since the laws of physics don’t change just because your purpose is different. Maybe some day a DAW (digital audio workstation) will include master buss processing to correct reference monitors for frequency response. Better to correct the speakers once & maintain that correction than to try to fix the frequency imbalance per channel.

Likely the most popular studio reference monitor of all time was the Yamaha NS-10.