Article

Thursday, May 10, 2012

In The Studio: What Engineers Should Know About Meters

This article is the second part in a series on decibels, excerpted from Bob Katz’s book Mastering Audio: The Art and The Science. The first part is available here, and be sure to check out all our videos featuring Bob Katz.

We Won’t Get Fooled Again. Recording engineers rely heavily on their favorite meter, and this is not intended to change people’s favorite.

But as practicing engineers, it is prudent to learn the defects and virtues of each meter we encounter.

The VU Meter. Relative newcomers to the industry may have never seen a VU meter, and some of them may be using the word “VU” incorrectly to describe peak-reading digital meters.

VU should only be applied to a true VU meter that meets a certain standard. The first thing we must learn is that the VU meter is a dreadful liar… It is an averaging meter, and so it cannot indicate true peaks, nor can it protect us from overload.

However the VU does do one thing better than a peak meter—it comes closer to our perception of loudness, but even so, it is a very inaccurate loudness meter because its frequency response gives low frequency information equal weight, and the ear responds less to low frequencies.

Another problem is that the VU meter’s scale is so non-linear that inexperienced operators think that the greater part of the musical action should live between -6 and +3 VU, but this is wrong.

A well-engineered music program has plenty of meaningful life down to about -20 VU, but since the needle hardly moves at that level, it scares the operator into thinking the level is too low.

Only highly-processed (dynamically compressed) music can swing in such a narrow range; in other words, the VU scale encourages over-compression.

Hence the VU meter should only be taken as a guide. A much better averaging meter would have a linear-decibel scale, where each decibel has equal division and weight down to -20 dB.

Digital Peak Meters
Digital Peak meters come in three varieties:
       1. Cheap and dirty
       2. Sample-accurate and sample-counting (but misleading)
       3. Reconstruction (oversampling)

Cheap and Dirty Peak Meters. Recorder manufacturers pack a lot in a little box, often compromising on meter design to cut production costs. A few machines even have meters which are driven from analog circuitry—a definite source of inaccuracy.

VU meter operators are often fooled into treating the top and bottom halves of the scale with equal weight, but the top half has only 6 dB of the total dynamic range.

Some manufacturers who drive their meters digitally (by the values of the sample numbers) cut costs by putting large gaps on the meter scale (avoiding expensive illuminated segments).

The result is that there may be a -3 and a 0 dB point, with a large unhelpful no man’s land in between. When recording with a meter that has a wide gap between -3 and 0, it is best practice to stay well below full scale.

Sample-Accurate and Sample-Counting Meters. Several manufacturers have produced sample-accurate meters with 1 dB (or smaller) steps, that convert the numeric value of the samples to a representation of the sample value, expressed in
dBFS.

The Paradox of the Digital OVER. When it comes to playback, a meter cannot tell the difference between a level of 0 dBFS (FS = Full Scale) and an OVER. That’s because once the digital signal has been recorded, the sample level cannot exceed full scale, as in this figure.

We need a means of knowing if the ADC is being overloaded during recording. So we can use an early-warning indicator—an analog level sensor prior to A/D conversion—which causes the OVER indicator to illuminate if the analog level is greater than the voltage equivalent to 0 dBFS.

If the analog record level is not reduced, then a maximum level of 0 dB will be recorded for the duration of the overload, producing a distorted square wave.

After the signal has been recorded, a standard sample-accurate meter cannot distinguish between full scale and any part of the signal that had gone over during recording, it shows the highest level as 0 dBFS. However, a sample-counting meter can analyze a recording to see if the ADC had been overdriven.

This meter counts contiguous samples and can actually distinguish between 0 dBFS and an OVER after the recording has been made! The sample-counting digital meter determines an OVER by counting the number of samples in a row at 0 dB.

If 3 contiguous samples equal 0 dBFS, the meter signals an OVER, because it’s fair to assume that the incoming analog audio level must have exceeded 0 dBFS somewhere between sample number one and three.

Three samples at 44.1 kHz is a very conservative standard; on that basis, the recorded distortion would last only 33 microseconds and would probably be inaudible.

While an original analog signal can exceed the amplitude of 0 dB, after conversion there will be no level above 0, yielding a distorted square wave. This diagram shows a positive-going signal, but the same is true on the negative-going end.

While this type of meter was sophisticated in its day, current thinking is that the sample-counting meter is only suitable for evaluating whether an ADC has overloaded.

Authorities now feel that meters which display the digital value of the samples and which count samples to determine an OVER are no longer sufficient for mastering purposes and should be used with caution during mixing. Their place is taken by…

The Reconstruction Meter: Even More Sophisticated As long as a signal remains in the digital domain, the sample level of the digital stream is sufficient to tell us if we have an OVER. However, signals which migrate between domains can exceed 0 dBFS and cause distortion.

This includes any signal that passes through a DAC, a sample rate converter, or is converted through a codec such as mp3 or AC3. During the conversion from PCM digital to analog or mp3, filtering within the converter yields occasional peaks between the samples that are higher than the digital stream’s measured level, which can be higher than full scale.

This next figure shows that contrary to what we might assume, filtering or dips in an equalizer which we’d imagine would produce a lower output can actually produce a higher output level than the source signal. B.J. Buchalter explains that:

“the third harmonic is out of phase with the fundamental at the peak values of the fundamental, so it serves to reduce the overall amplitude of the composite signal.”

“By introducing the filter, you have removed this canceling effect between the two harmonics, and as a result the signal amplitude increases. Another reason for the phenomenon is that all filters resonate, and generally speaking, the sharper the filter, the greater the resonance.”

Equipment designers have known for years that because of filtering, the analog output level of complex audio from a DAC can exceed the sinewave value of 0 dBFS but very few have taken this into account in the design.

TC Electronic has performed tests on typical consumer DACs, showing that many of them distort severely since their digital filters and analog output stages do not have the headroom to accommodate output levels which exceed 0 dBFS!

While typical 0 dBFS+ peaks do not exceed +0.3 dBFS, some very rare 0 dBFS+ peaks may exceed full scale by as much as 4 or 5 dB with certain types of signals— especially mastered material which has been highly processed, clipped (turned into a square wave on top and bottom), and/or brightly equalized.

By oversampling the signal, we can measure peaks that would occur after filtering. An oversampling meter (or reconstruction meter) calculates what these peaks would be, but these meters are still rare. Products from TC Electronic (System 6000) and Sony (Oxford) have an oversampling limiter and reconstruction peak meter. RME’s Digicheck software includes an oversampling meter.

Reconstruction meters tell us not only how our DAC will react, but what may happen to the signal after it is converted to mp3 or sent to broadcast, both of which employ many filters and post-processes. Many DSP-based consumer players cannot handle the high levels at all and exhibit severe distortion with 0 dBFS+ signals.

Armed with this knowledge, no mastering engineer should produce a master that may sound acceptable in the control room but which she knows will likely produce severe distortion when post-processed or auditioned in the real world.

If the reconstruction meter is not enough to convince the client, she should also demonstrate that this “loud” signal becomes distorted, ugly, and soft when it is converted to low bit rate mp3. All the harmonics which made the signal seem loud in the control room have been converted to additional distortion.

Practice Safe Levels
What this means is that if you are mixing with a standard digital meter, keep peaks below -3 dBFS, especially if you are using aggressive bus processing.

The more severely processed, equalized or compressed a master, the more problems it can cause when it leaves the mastering studio.

We didn’t start hearing about this problem, or at least the severity of it, before the loudness race and the invention of digital processing which could be egregiously abused. Maximizing engineers should try to use a reconstruction meter and/or an oversampled brickwall limiter. If these are not available, use a standard peak limiter whose ceiling is set to -0.3 dB (see Chapter 10) and exercise caution.

But even the oversampled brickwall limiter is not foolproof; I’ve discovered that such limiters do not protect from very severe processing and can still make a consumer DAC overload unpleasantly. The best solution is to be conservative on levels. Clipping of any type is to be avoided, as demonstrated in Appendix 1.

The Myth of the Magic Clip Removal
If the level is turned down by as little as 0.1 dB, then a recording which may be full of OVERs will no longer measure any overs.

But this does not get rid of the clipping or the distortion, it merely prevents it from triggering the meter.

Some mastering engineers deliberately clip the signal severely, and then drop the level slightly, so that the meters will not show any OVERs.

This practice, known as SHRED, produces very fatiguing (and potentially boringly similar) recordings.

Peak Level Practice for Good 24-bit Recording
Even though 24-bit recording is now the norm, some engineers retain the habit of trying to hit the top of the meters, which is totally unnecessary as illustrated at left.

Note that a 16-bit recording fits entirely in the bottom 91 dB of the 24-bit. You would have to lower the peak level of a 24-bit recording by 48 dB to yield an effective 16-bit recording! There is a lot of room at the bottom, so you won’t lose any dynamic range if you peak to -3 dBFS or even as low as -10 dBFS, and you’ll end up with a cleaner recording.

Since distortion accumulates, if a “hot” recording arrives for mastering, the mastering engineer doing analog processing may have to attenuate the level to prevent the processing DAC from overloading. A digital mix that peaks to -3 dBFS or lower makes it easier to equalize and otherwise process without needing an extra stage of attenuation in the mastering.

In black is a complex wave. When the high frequency information (light orange) is filtered out, the result is a signal (orange) that is higher in amplitude than the original!

A number of 24-bit ADCs are advertised as having additional headroom, achieved by employing a built-in compressor at the top of the scale, claiming that the compressor can also protect the ADC from accidental overloads. But this is specious advertising.

Level accidents don’t occur in a mix studio; engineers have control over their levels and when tracking live musicians, it is better to turn off the ADC’s compressor, drop the level and leave plenty of headroom for peaks. The only possible use of this function of a compressor is if you like its sonic qualities and are trying to emulate the sound of tracking to analog tape.

But since tracking decisions are not reversible, I suggest postponing “analog simulation” to the mixing stage. It’s easier to add warmth later than try to take away some mushiness due to an overdriven compressor. As we have just seen, there is no audible improvement in SNR by maximizing a 24-bit recording and no SNR advantage to compressing levels with a good 24-bit ADC.

How Loud is It?
Contrary to popular belief, the levels on a digital peak meter have (almost) nothing to do with loudness.

Here is an illustration. Suppose you are doing a direct to two-track recording (some engineers do still work that way!) and you’ve found the perfect mix.

Leaving the faders alone, you let the musicians do a couple of great takes. During take one, the performance reached -4 dB on the meter; and in take two, it reached 0 dB for a brief moment during a snare drum hit.

Does that mean that take two is louder? No: because in general, the ear responds to average levels, not peak levels when judging loudness.

If you raise the master gain of take one by 4 dB so that it too reaches 0 dBFS peak, it will sound 4 dB louder than take two, even though they both now measure the same on the peak meter.

An analog tape and digital recording of the same source peaked to full scale sound very different in terms of loudness. If we make an analog tape recording and a digital recording of the same music, and then dub the analog recording to digital, peaking at the same peak level as the digital recording, the analog dub will have about 6 dB more intrinsic loudness than the all-digital recording.

Quite a difference! This is because the peak-to-average ratio of an analog recording can be as much as 12-14 dB, compared with as much as 20 dB for an uncompressed digital recording.

Analog tape’s built-in compressor is a means of getting recordings to sound louder (oops, did I just reveal a secret?). That’s why pop producers who record digitally may have to compress or limit to compete with the loudness of their analog counterparts.

The Myths of Normalization

The Esthetic Myth. Digital audio editing programs have a feature called Peak Normalization, a semi-automatic method of adjusting levels.

The engineer selects all the songs on the album, and the computer grinds away, searching for the highest peak level on the album and then automatically adjusts the level of all the material until the highest peak reaches 0 dBFS. If all the material is group-normalized at once, this is not a serious esthetic problem, as long as all the songs have been raised or lowered by the same amount.

But it is also possible to select each song and normalize it individually, but this is a big mistake; since the ear responds to average levels, and normalization measures peak levels, the result can totally distort musical values. A ballad with little crest factor will be disproportionately increased and so will end up louder than a rock piece with lots of percussion!

The Technical Myth. It’s also a myth that normalization improves the sound quality of a recording; it can only degrade it. Technically speaking, normalization adds one more degrading calculation and level of quantization distortion.

And since the material has already been mixed, it has already been quantized, which predetermines its signal-to-noise ratio—which cannot then be further improved by raising it.

Let me repeat: raising the level of the material will not alter its inherent signal-to-noise ratio but will add more quantization distortion. Of course material to be mastered does not need normalizing since the mastering engineer will be performing further processing anyway. Clients often ask: “do you normalize?” I reply that I never use the computer’s automatic method, but rather songs are leveled by ear.

A 24-bit recording would have to be lowered in level by 48 dB in order to reduce it to the SNR of 16-bit. The noise floors shown are with flat dither.

Average Normalization
This is another form of normalization, an attempt to create an intelligent loudness algorithm based on the average level of the music, as opposed to the peak.

But when making an album, neither peak nor average normalization nor any intelligent loudness algorithm can do the right job, because the computer does not know that the ballad is supposed to sound soft.

There’s no substitute for the human ear. However, average normalization or better, a true intelligent loudness algorithm can help in situations where every program needs the same loudness, even if that doesn’t sound natural, such as radio broadcast, ceiling loudspeakers in a store, a party or background listening.

Judging Loudness the Right Way
Since the ear is the only judge of loudness, is there any objective way to determine how loud your CD will sound? The first key is to use a single DAC to reproduce all your digital sources and maintain a fixed setting on your monitor gain.

That way you can compare your CD in the making against other CDs, in the digital domain. Judge DVDs, CDs, workstations, and digital processors through this single converter.

More Than One Way: Alternatives To High-Voltage Audio Power Distribution

In audio terms, high voltage means that the output power of the amplifier is converted to a high-voltage/low-current signal for transmission over long distances and/or small wire gauges.

The advantages of the method include low cost and rather “bulletproof” systems, and the downside is that the transformers required present yet another filter for the signal to pass through, often degrading the audio quality.

Since loudspeaker lines should always be kept as short as possible, the ultimate realization of this involves placing the amplifier right at the loudspeaker, connected to it by inches of cable.

This method is best understood by looking at the workings of a traditional power amplifier. There are many shapes and sizes, but they all have some commonalties.

First, all amplifiers take AC power (alternating current) and use it to amplify signals. This requires converting the 60 Hz sinusoidal signal from the power company into something that looks like the audio signal that we wish to amplify. Several steps are required to accomplish this.

To begin with, the voltage component of the power is transformed from 110 or 220 volts (common distribution voltages) into the voltage required by the amplifier circuitry, which is determined by the power rating of the amplifier.

Next, the new value of voltage and current is rectified into DC (direct current). In DC form, the power can be “modulated” by the audio signal voltage to form a higher-power facsimile of the input signal voltage to the amplifier. This step is accomplished by the output stages of the amplifier.

Figure 1 (below) shows the parts of a typical power amplifier. Conventional systems take the amplified output of the power amplifier and feed it to loudspeakers through a wire gauge of sufficient size to minimize the power loss to 0.5 dB.

When the required wire gauge becomes too large, the power is delivered to the loudspeakers by transforming it into a high voltage/low current signal, more suitable for traveling long distances.

Distributed amplification systems involve separating the parts of the amplifier and distributing them to remote physical locations that, for some applications, better optimize the wiring and interconnects that make a sound system work.

For instance, if the output stage of the amplifier were placed right at the loudspeaker, there would only be a need for a few inches of loudspeaker cable. In order to avoid having to run AC power to all loudspeaker locations, a central DC power supply can be used to drive many amplifier sections.

The central supply can be located near the AC power source, and the DC output coupled to the amplifier sections through appropriate cabling.

For large systems, several power supplies can be distributed to keep the distance between them and the loudspeaker/amplifier units at a minimum.

All that remains is to get the electrical signal voltage to all of the “distributed” amplifier/loudspeakers.

Since this is a line level signal, it can be run very long distances without significant degradation.

Finally, if the DC and signal are run through the same multi-conductor cable, installation of such a system is greatly facilitated.

For electronic systems, DC is an ideal way to power things, since almost every unit in a sound system must convert AC to DC in order to work.

In fact, when the AC power standards were established years ago, there were many people, including Thomas Edison, that wanted to use DC distribution. It makes a lot of sense for much of what we use power for.

Advantages:
Short Loudspeaker Lines
Higher Fidelity
Lower Operating Voltages
Conduit is not required in many locals

Disadvantages:
Increased cost over conventional systems
Upgrades are more complex (this is probably an advantage. Ask anyone who has ever had a customer hang a transformerless loudspeaker on their 70 volt line and load it down).
Very high power amplifiers not available

Author’s Note: The technology was actually developed by Richard Heyser of the Jet Propulsion Laboratories in the 1980s.

Pat & Brenda Brown lead SynAudCon, conducting audio seminars and workshops around the world. For more information go to synaudcon.com.

Church Sound: The Keys To Presenting Audio That Will Engage Your Congregation

Producing great sound in a worship service can seem as elusive as finding a soloist who always sings on key.

However, this doesn’t have to be so.

Many factors influence the quality of sound: room acoustics, sound-system design and performance, operator experience, and quality of musical performance.

Here are some practical tips on how to tie all of that together to get the best sound.

Understand the Basics
To get the most out of a sound system, you must first understand how it works.

Basically, acoustic energy, or the sound you make, is converted to electrical energy via a microphone, then combine with other mics or input devices, and are then “colored” or equalized via a mixer.

The mixer sends the sound through processing equipment (crossover, equalizer, signal delay, usually a single DSP box), then to amplifiers to boost the signal. Finally, the amplified signal goes to speakers, where it’s transferred back to acoustic energy.

The key components of sound-processors, amplifiers, and speakers-should be professionally designed and set in a church, then left alone. The mixing board is where you should make adjustments in tone and sound levels.

Build a Sound Team
A sound system won’t run by itself. It needs a sound crew to function to its true potential. Some ideas on recruiting and developing a good crew:

I like to recruit one-on-one, much like a hunter who goes to the woods looking for a specific target. The hunter may see ducks, squirrels, and turkeys, but he sits tight for a certain kind of deer. When he sees exactly what he’s looking for, he pursues it with vigor.

Be the same way when developing a sound team. Decide what kind of people you need, and then recruit them vigorously.

You could also try the fishing-pond approach. That means recruiting candidates from a select gathering of people.

For example, when Marty O’Connor was at Willow Creek Community Church in South Barrington, Illinois, he and his video crew offered a yearly seminar on how to make great home movies with your camcorder.

After the seminar, the crew would bring out their studio cameras and invite seminar attendees to try operating one of the “big boys.”

All the while they’d look for people in that “pond” with special aptitude for working on a video crew. Then they’d recruit them. Members of a sound crew might be found through a similar approach.

Grow A Team
The acronym TEAM - meaning “Together Everybody Achieves More” - particularly applies to a sound crew. To be truly effective, team members must grow together on the job in knowledge and experience as well as in spirit and emotion.

Make sure that you provide spiritual, emotional, and technical food for sound-team members.

When I was on staff at a large West Michigan Church, every week, I spent about 30 minutes in prayer and devotions with my sound crew before our hour-plus sessions in sound training. That time helped unite us and focus our work.

It’s also important to keep the team informed of what’s happening in the sound industry, such as regular visits to ProSoundWeb and reading other industry publications and sites.

Finally, to encourage ownership and 100-percent participation, every sound-crew member should be encouraged to make suggestions about the sound system. I took seriously crew member suggestions on equipment purchases.

Thank the team. Saying thanks is powerful, but showing thanks is even better. My favorite way of showing gratitude to crew members was to send thank-you notes to them and their spouses.

Aim For Consistency
“We are what we repeatedly do,” Aristotle once wrote. “Therefore, excellence is a habit not an act.”

Doing everything right with sound in a performance is hard enough, but repeating it can seem impossible, especially when different volunteers are involved.

To raise the percentage of success, standardize the layout of your mixing console, label it, and then get everyone to conform to it.

Example: I always lay out my mixing console with drums on the left, followed by bass, electric and acoustic guitar, then keyboards, and finally vocals.

The lead vocal is always in the farthest right channel next to the subgroups and masters. I’ve been doing that for the past 25 years. My technical team follows this layout consistently.

How you lay out the board doesn’t matter as long as it’s logical and everyone follows it. The advantage of such a layout is that when something goes wrong or there’s feedback, you know instinctively what to grab to fix it.

Aim for consistency also with equipment storage. Organize cables, stands, and mics so that even with last-minute changes, such as having to work with five singers instead of the four you had planned on, you can secure the proper equipment to keep a rehearsal moving.

Preparation, Preparation
When I was on staff as a technical director, I was blessed with a worship leader who provided worship-service outlines weeks in advance. I used to kid him that the Spirit moved in him two weeks before it hit the congregation.

One lesson I learned from him is that someone who is well prepared is able to respond much better to last-minute complications than someone who wings it.

I have served as a consultant to churches that supposedly had sound system problems, only to discover that the real problem was poor preparation.

Example: A sound team shows up at 8 a.m. to set up for a 9:30 a.m. service in a temporary facility. By 9 a.m. the sound system is set up, and a CD is playing. Musicians begin arriving for a last-minute rehearsal.

The service starts seven minutes late. That’s bad enough, but what’s worse is that there has been no time for sound checks and input testing. The service proceeds, accompanied by hums, cracks, pops, and a lousy sound mix.

Ninety minutes later, the sound crew is exhausted, the musicians disgusted, and the pastor fed up. He decides to call in a sound expert.

He needn’t have spent the money. Preparation would have alleviated most of the problems.

Preparation means sending information to your team well in advance of a service. Email (or post on the church’s website) the order of worship for the Sunday service to crew members early in the week so they can get a jump-start on what they’ll need to do.

Preparation also means performing sound checks with musicians prior to the service and testing all microphones. Even if the same person leads worship every week, he or she may have a cold or feel insecure about a piece of music and need their monitor turned up.

The key is to show up early, anticipate the unexpected, and be prepared. You can’t be too prepared.

Provide Technical Training
Offer ample opportunities for your team to grow in technical knowledge.

Find a sound expert you respect and hire that person to come in two to four times a year to train your crew.

Team up with other churches to sponsor a regional conference on sound, like the HOW TO seminars.

Send for brochures and guides or reprint articles on sound for your crew.

Many manufacturers, such as Shure and Crown, provide free guides, and often, these are posted online for convenient download.

Encourage your crew to participate in focused online discussions about sound with online communities such as the Church Sound Forum here on PSW.

Lead your team by example. If you want your crew to be on time, be on time yourself. If you want others to keep the sound booth and related areas organized and clean, keep your areas organized and clean.

Encourage Relationships
To do its work well, a sound crew must work in harmony with musicians and pastors.

All too often there’s friction between sound technicians and performing artists. Some of that could be eased organizationally by including sound technicians in the church’s fine arts or music ministry.

The lead person of the technical team would report directly to the worship leader or minister of music—no one else. They would work things out, striving for communication and harmony.

Example: I saw how that could work at a recent sound seminar. A local worship leader and his worship team participated in a session I led titled “Mixing a Worship Team: A Live Demonstration.”

We purposefully had no rehearsal or sound check before the seminar. We merely tested the inputs to make sure they were working.

During the seminar, a conflict arose between the piano and keyboard players. The keyboard player wanted more of him in the monitor, and the piano player wanted less. The problem: they were sharing a monitor mix.

The worship leader let me know about the problem, and I told the players that since there were no more monitor mixes available, they should work out a solution together.

He led the players through a quick trial on the monitor until the players reached agreement. They reached harmony in less than three minutes.

Tip: The key was the worship leader’s willingness to tell me about the problem, and the opportunity I had to explain the setup limitations to the players.

I’ve discovered that when technical people are given the opportunity to explain a problem, performers are very cooperative.

Of course, technicians must never abuse that trust by blaming their mistakes or ignorance on equipment, or by refusing to listen to a musician who needs adjustments in a monitor.

Trust can also be destroyed by performers or technicians whose egos get in the way of working with others. In the sound booth or in front of a mic, the motto should be: “Check your ego at the door.”

I also know how important a good relationship can be between a technician and artist. I spent four years working with the same worship leader.

We had such rapport that we could communicate from sound booth to platform via hand signals.

When the worship leader put two hands on the mic, I knew I had to put more piano level in the monitor. Two hands with a raised index finger meant he wanted more voice. A step back from the monitor meant it was too loud.

The signals worked well because I kept my eyes on the platform, and the worship leader always made eye contact before signaling.

Serve Others First
If we serve others first, we have far less friction between sound technicians and performers. Here are some ways sound people can serve others to enhance their ministry to the church:

Show up early to set the sound equipment with enough time left to pray with speakers and singers before a service.

Provide little extras for platform participants, such as a glass of fresh, cold water near the lectern.

Take the pastor and/or worship leader out to lunch in appreciation for their support. Tell them how much you value their contribution.

Explain to singers or speakers what you’re doing to adjust their sound and why.

For example, tell them you’re moving a monitor two feet to the left so that the sound from the monitor is in the non-pickup area of the microphone and will thus give them a purer sound with less risk of feedback.

The Ultimate Goal
The sound ministry is like custodial service. When it’s done well, few will notice. When done poorly, everyone will notice.

Work as a respectful team, and you’ll find that your sound is consistently excellent, and you’ll have a great time to boot!

Gary Zandstra is a professional AV systems integrator with Parkway Electric and has been involved with sound at his church for more than 25 years.

Simple Tool: The Importance Of Recording Live Shows For Later Evaluation

Several years ago, I had the pleasure of mixing a few times for a very talented rock band.

The first show was a CD release party at a local all-ages club, and I got a call about an hour before the band was to take the stage.

On the other end of the phone was the band’s bass player (an old friend), who asked me to come down and mix. They were the second band that night and had no confidence in the mix engineer after hearing the first band. We sealed the deal with food, a T-shirt and a CD.

When I arrived 20 minutes later, the first band was still on stage, and it was immediately apparent why they called me. In addition to several “issues” on the console side of things, there were improperly set gates on kick and snare that were doing nothing more than cutting off the attacks.

Yet my first reaction was compassion for the engineer, because I’d been in similar situations in the past where I had absolutely no comprehension that whatever I was trying to do was actually doing more harm than good. We can often get fixated on one idea and completely fail to hear the damage we’re leaving in our wake.

When it was time for my guys to go on, there was only time for a quick setup. I pulled down the gains, opened up the faders, ditched the gates, and then jumped on stage to set the microphones.

After a two-minute check, the band hit the ground running. It was a great show, but certainly not on my account – all I did was make sure the system didn’t get in the way, and the musicians did the rest.

Some of us have been fortunate enough to apprentice under a great engineer and receive plenty of input on how to do things better. But for every person who benefits from such a relationship, there are many more who don’t.

Fortunately, there’s a simple tool that can help. I’m talking about recording the live mix for later evaluation. Too often, those who try it quickly abandon the idea because they don’t like what they hear, and it’s easier to blame the recording by saying it doesn’t sound like it does in the room rather than accept that what they’re hearing live is tainted by what they think.

I record my live mix at every opportunity so that I can listen to it later for insight about myself and what I thought I was hearing during the gig. So far, I’ve come away from this process with two revelations.

First, it’s critical to not over-think things. I believe this to be the bane of most technically oriented people. There comes a point where the specifications become more of a distraction rather than helping.

Second, being able to evaluate a mix and determining what needs to be cut in level is a greater expertise than boosting levels for certain mix elements that can’t be heard.

Should we expect to hear a ready-to-release recorded mix? Of course not. But we should be able to hear the foundation, pad, rhythm, lead and fill, all more or less in their correct relationships.

My recording process varies. If it’s a standing gig, I use whatever is at hand - cassette, DAT, CD-R, or straight into a Mac/PC audio input. For portability, I had an AIWA HD-S1 DAT recorder that
went with me to every gig, and when it died, I replaced it with an Edirol R-09 flash recorder (which fits beautifully in a cast-off Audix drum mic carry case).

When I have more time and want top quality, I bring a rack case with my Alesis HD24 multitrack recorder and Apogee PSX-100 converter for patching to a pair of the main out jacks on the console.

Are these mixes perfect? No. But we have to keep things in perspective. A good live mix will have all of the elements in place, and these should be there to be heard on the recording.

Great sound can be relatively easy to achieve. The solution is in removing the ideas we think up that get in the way.

Since his start more than 30 years ago on a Shure Vocalmaster system, James Cadwallader remains in love with live sound. Based in the western U.S., he’s held a wide range of professional audio positions, performing mixing, recording, and technician duties.

Designer Notebook: MLA Compact, A Lower Profile Approach From Martin Audio

In 2010, Martin Audio brought a radical new approach to touring sound with the Multi-cellular Loudspeaker Array (MLA), an integrated system that is self-powered with six channels of amplification and DSP, networked with both audio and control on the same cable.

Furthering the package is a combination of cellular drive and fast, automated optimization software. Cellular technology helps achieve phase-coherent summation across the respective coverage area, holding both frequency response and SPL within a very tight, user-specified window.

The new MLA Compact widens the scope of applications, designed for situations not requiring the full power and throw of the full-size MLA, or where a smaller, lighter system is called for. It is scalable, and can meet the needs medium-scale touring and fixed installations such as concert halls, theatres and churches.

The footprint of an individual cabinet is 31 (w) x 11 (h) x 19.7 (d) inches, and weight is 109 pounds. As a point of comparison, a full-size MLA box measures 44.7 (w) x 14.6 (h) x 26.5 (d) inches and weighs 193 pounds.

In The Box
Compact loudspeakers often use direct radiators for the lows and mids because more efficient acoustic technologies are hard to adapt to small cabinet volumes. The 3-way MLA Compact, however, applies new slot and horn-loading techniques to the low and mid sections.

Like it’s bigger brother, MLA Compact presents an integrated approach. (click to enlarge)

Further, it deliberately avoids the use of coaxial, co-entrant or cross-firing midrange/HF driver arrangements. which introduce acoustic discontinuities that can affect the on- and off-axis frequency response of both MF and HF sections. In MLA Compact, the MF and HF horns are completely separate in order to produce consistent, true 100-degree horizontal constant directivity coverage.

The LF section utilizes 2 x 10-inch (2.5-inch voice coil) neodymium drive units in a unique Hybrid configuration that fosters acoustic efficiency and enhances “punch.” Each driver is slot-loaded into a truncated horn with a low flare rate, to give a high sensitivity of 103 dB @ 1m/2.83V. The slot-loading allows the twin LF drivers to be optimally spaced within the enclosure.

A look at the slot- and horn-loading approaches inside MLA Compact. (click to enlarge)

This double-source arrangement significantly improves the directivity control of the LF section maintaining the 100-degree beamwidth down to 250 Hz and reducing mid-bass output at the sides and rear of the array. The LF drivers are very high excursion with vented poles to reduce power compression and eliminate turbulent air noise.

MLA Compact has separate MF and HF horns with horizontal constant directivity characteristics, so the horizontal off-axis response tracks the on-axis response exactly.

The MF horn utilizes 2 x 5-inch (1.5-inch voice coil) neodymium drivers to produce 109 dB @ 1m/2.83V, which represents a significant gain on the typical efficiency of 102 dB found in cross-firing direct radiator designs in comparably sized systems. It’s possible due to acoustic and thermal designs that utilize forced-air cooling and a thermally conductive aluminum housing.

The I/O capability on the back of each cabinet. (click to enlarge)

The HF section includes 4 x 0.7-inch-exit neodymium compression drivers that feed separate horns providing 100-degree horizontal constant directivity. The driver combination replaces the more traditionally used large-format compression driver and has less distortion, as well as having a more extended HF response.

Instead of adopting flat wavefronts as advocated by some of the early proponents of touring line arrays, sophisticated in-house BEM (Boundary Element Method) modeling techniques have shown that slightly curved wavefronts deliver much more consistent SPL to the audience where the array is curved – as in most practical, real-world applications.

Placing a kite-shaped “wedge” part-way down the horn enables a specific, desired curvature to be achieved – depending on the shape of this wedge.

In the case of MLA Compact, the HF wavefront is curved to provide a balance between optimal summation over distance and summation at the maximum inter-cabinet splay angle of 10 degrees.

Power & Control

Each MLA Compact has 5-channel Class D amplification, DSP and network electronics that enables the individual cells within each enclosure to be driven with the exact signal determined by DISPLAY 2.1 optimization software for independent control of each of the 24 LF cells, 48 MF cells and 48 HF cells in a 24-box array.

The amplification package provides a total of 2.1kW continuous and 4.2kW peak output. One channel powers both LF drivers in parallel and two channels drive each mid independently. For the HF section, two channels drive the four HF drivers in parallel pairs, making a total of five independently powered acoustic cells per enclosure.

Switched-mode power supplies auto-range to global mains voltages from 100 to 240 volts, 50/60 Hz, while Power Factor Correction smooths out the mains current draw over the whole of the AC waveform. Amplifier monitoring via U-NET includes input signal, output signal at the drive unit terminals, limiter status, heatsink temperatures and driver fault conditions.

Onboard DSP performs all crossover and EQ functions via a combination of IIR and advanced FIR filtering. Fast VanishingPoint FIR filters provide the ability physically separate the MF and HF horns so they do not compromise each other’s constant directivity dispersion pattern, yet achieve the spatial performance of a single device.

MLA Compact incorporates an automated optimization process. (click to enlarge)

Arrays can be remotely controlled over U-NET from a PC or wireless tablet running VU-NET control software with its intuitive graphical interface. VU-NET also enables the user to switch on enclosure identification LEDs with automatic identification of neighboring enclosures and connectivity confirmation.

Dialing It In

DISPLAY2.1 is the “brain” of MLA Compact, providing a virtual environment within which arrays can be configured and optimized, providing highly accurate prediction of the direct sound produced over the audience and also and over areas where sound is to be avoided. It also bolsters array design and deployment, generating accurate spot frequency responses and comprehensive rigging information, including mechanical load safety analysis.

DISPLAY2.1 interacts with MLA Compact’s onboard DSP to deliver consistent sound throughout a venue. It calculates the filter parameters for each enclosure, down to the resolution of individual drive units, and uploads them to the enclosure via the U-NET digital network. The link between DISPLAY2.1 and an individual MLA Compact enclosure is live and bidirectional.

DISPLAY2.1 reverses the sequence of array design software. Starting with a specified SPL and response over the audience floor, the software works backwards to configure an array that will give the required result.

BEM plots of a horn in an array (left) and of a single horn on its own. (click to enlarge)

Since it is a practical impossibility to measure every possible array configuration with different combinations of enclosure numbers, splay angles and drive signals, an accurate acoustic model is essential.

The previously noted BEM (Boundary Element Method) models enable hundreds of virtual array configurations to be investigated in fine detail in a virtual 3D environment. This level of research has transformed our understanding of how arrays really work and shown that the acoustic interactions between array elements are much more complex than originally thought.

An important factor and industry first is the inclusion in the model of the previously ignored effects of adjacent enclosures. If these are not incorporated into the model, prediction errors can be over 8 dB in the midrange. Including the effect of adjacent cabinets is key to the accuracy of the optimization process and increases the accuracy of the acoustic model of MLA Compact.

Completing The Package

The loudspeaker rigging system allows up to 24 enclosures to be suspended via its 2-point-lift flybar, and the same hardware can also be used for single point lifting of up to 12 cabinets, as well as ground stacking up to six high. Inter-cabinet connections utilize custom quick-release pins.

All loads are borne by the integral metalwork and release pins - not the enclosure. Accompanying software determines the safe limits and tilt angles of a specific array, with BGV C1 safety calculations done on the fly.

The enclosure is made of birch and poplar ply construction and finished with a thick polyurea coating. The enclosure sides are fitted with replaceable, steel-reinforced rubber moldings with integral interlocking skids, and an ergonomic bar-handle facilitates rigging and general handling. MLA Compact is supplied in flight-cased pairs, with the base doubling as the wheelboard for the pair.

The companion DSX subwoofer offers is a dual-18-inch (4-inch voice coil) ferrite drivers in reflex-loaded enclosure with four flared ports, designed to improve linear airflow. Each driver is housed in its own separate chamber to move any cabinet resonances out of band. The onboard Class D amplifier module can deliver up to 6 kW of peak power. DSX sub arrays can be designed with specific directional properties and DSP settings uploaded via the VU-NET network.

DSX Subs can be flown alongside or at the top of arrays, as well as groundstacked.(click to enlarge)

A flying version, the DSX-F Sub, can be flown alongside or at the top of MLA Compact arrays, as well as being ground stacked. A maximum of 15 DSX-F can be suspended from the MLA flying frame and symmetrical rigging allows flown DSX-F cabinets to face backwards—enabling directional flown as well as directional ground stacked arrays to be configured. The ground-stack DSX can be upgraded to the DSX–F by the addition of an easy-to-fit accessory kit.

What all of this adds up to is a compact, highly flexible package that provides startlingly consistent frequency response when listening off-axis and “walking the field.” In both mobile and permanent applications for audiences ranging up to 5,000, this is no small feat.

Jason Baird is research & development director at Martin Audio. After completing a degree in Electronic Engineering at Leeds University in England in 1991, he worked in the production industry until he was employed by Wharfedale Loudspeakers in ‘94. After starting his career in hi-fi, he spent 18 months at Fane Acoustics, designing pro audio transducers, subsequently joining Martin Audio in 1998.

Wednesday, May 09, 2012

In The Studio: Top Ways To Help Musicians Hear Themselves

This article is provided by BAMaudioschool.com.

 
I’m often heard saying that the recording engineer’s job is to create an environment conducive to musical creativity and then capture that creativity.

Headphones are usually the only way that a musician will be able to hear themselves and (more importantly) how what they are playing works with the rest of the band.

Every musician will (eventually) ask to hear themselves much louder than everyone else. This makes sense as it will allow them to play the nuances of their instrument.

However, if they are only listening to themselves (or the click track) and not the everything else then what they play may be wonderful by itself but terrible within the mix. They may even compromise the art of their own playing as a result of a poor headphone mix.

For Example:
•—Guitar players who hear themselves too loudly will not “bear down” with the pick as much as they may need to.
•
—Piano players who hear themselves too quietly may not play with the full dynamic range of the piano if they cannot hear themselves play softly.
•
—And finally, any musician that cannot hear the full rhythm will cause a combined pushing and pulling of all the instruments, and no one will be together or “in the pocket,” even if they are overdubbing alone.

Remember, you must make the musicians feel like they are playing together in a room without headphones (in fact I prefer to record bands that way). They have to be able to hear and feel each other clearly.

Sometimes you may have the luxury of multiple headphone feeds, which will allow you to tailor different mixes for the players that require them. Even given the advanced personal mixer technology available today, always be wary of letting musicians mix their own headphones completely by themselves, as they will tend to want to hear only themselves.

A Few Pointers:

1. No matter how loud the drums may be in the room, everyone needs some kick, snare, hat and other drum microphones. The timing and feel of the drum mics will sound different from the drum sound in the room.

2. Panning can be your friend. Sometimes moving some instruments just slightly off center will make it easier to the players to hear themselves without increasing volume or resorting to making the moniutor mix a solo mix for certain individuals.

3. You can always change the sound musicians hear in their headphones without compromising the sounds you record.

Once, I was recording a large horn section that was used to a compressed edgy sound. I wanted to go for something full, so I recorded them using a combination of ribbon and condenser mics going flat from Neve mic pre’s straight into the tape machine.

The section was not happy and complained that the sound was not what they were used to. I did not want to lose the fullness the mics were giving me, so I EQ’ed and compressed the monitor channels coming off the tape machine. Suddenly they were all happy and played well.

When I mixed, I was able to use all of the sounds with absolutels no EQ or compression (until those effects were called for) and was very pleased with the results. If I had changed the sound I was capturing to match what the musicians were used to hearing in their headphones, the final sound of the section would have suffered.

4. Make sure the musicians hear enough of the band and even the beat that they can perform to the song rather than just lay down their parts. Musicians will (and should) be concerned with their performances, but do not let them lose sight of the fact that they are playing within a song along with other musicians.

If they do not hear the others they will not be able to interact with them, even if it is only on a subconscious level.

5. Some drummers will ask for loud click tracks in their headphones. If you have only one headphone feed and the drummer needs to share the cue with other performers, it may be tricky for you to keep everyone happy. You may need to ride the click.

And, speaking of riding the click….

6. Be prepared to ride the click track down in softer sections of a song, especially at the end.

There is nothing worse than trying to mix the very end of a song and having to fade out too quickly to keep the click from the drummer’s headphones from being heard.

Bruce A. Miller is a recording engineer who operates an independent studio and the BAM Audio School website.

What Benefits Do Digital Consoles Hold For Clubs?

It’s common for live music clubs to economize with second-hand sound equipment, and some value can be found in older loudspeakers and analog desks which are no longer rider-friendly enough for their original owners.

Used analog consoles cost a fraction of their original price, so it’s not unusual to find club equipment older than the engineer running it.

However, concert sound production is accustomed to the benefits that digital consoles provide.

Let’s see what those benefits bring to live music clubs.

1) A multi-band night can be as simple as peeling off stacked backline, swapping drum kits and moving a few mics, which takes longer than recalling the correct console file.

“Charting” consoles to preserve their settings or using separate inputs for each act becomes unnecessary.

When sharing inputs, a 24-channel console is often plenty and the next act’s settings are recalled faster than the previous band’s gear can be carted off the stage.

2) Saving scenes for acts that return regularly, especially when they have to travel from out-of-town, makes it easy to quickly set-up and get sound checked when time is short, perhaps even after the opening act has finished.

Popular acts that return regularly can afford to skip soundcheck, making it possible for them to arrive later in the day, a great benefit when they must travel long distances. Off-line editors allow new acts to email their console settings ahead of time.

3) Custom input libraries for the house microphone inventory speeds and simplifies console programming.

Most clubs have a limited mic inventory, but writing and storing specific settings for the house mics allows sound check to get started in a few minutes by inserting the right files in the corresponding channels.

4) Application-specific settings for dynamics and effects also make it easier to program a digital console.

Meaningful, logical names for library settings – like “Tom Gate,” “Bass Comp” and “Vocal Verb” – can make it simple to find and load a file for a gate, compressor or reverb that is pre-tweaked for a particular application.

5) Output EQ for specific combinations of loudspeakers and vocal mics (and even music style) can make the chore of tuning the mains and monitors much easier. Different vocal mic and wedge combinations require different output EQ settings.

The same vocal mic with a double wedge would require more drastic cuts than a single wedge. Output EQ for the main loudspeakers often need to be adjusted for different types of music, especially for different levels of SPL.

Settings like “Double Wedge 58” or “Reggae Mains.” can save time before every show.

6) Having entire generic “festival” scenes for different types of bands can make it effortless to get a band’s sound check started, rather than starting from scratch with a zeroed console.

Just having the inputs named, initial input gains, EQ’s, dynamics and effects loaded saves time, and the time spent tweaking those settings can result in a better sounding mix.

A simple four-piece of drums, bass, guitar and keyboard setup with a vocal input for each musician could save an hour.

7) Templates for special functions in the room, such as DJ’s, singer-songwriter night, karaoke or wedding receptions can relieve your lead sound engineer from having to be at every function that occurs in the club.

The ability of every club staff member to call up a preset and easily get a couple mics and a playback source to work can help better utilize the venue outside of weekend evening “prime time.”

8) Security features that lock out certain functions can keep a fairly complicated console from being intimidating and dangerous in the hands of novice engineers.

Remote monitoring of console by an outside vendor can help with troubleshooting in the case of an emergency, perhaps saving the trouble and expense of an on-site service call.

Routine backup of the console’s files helps recovery from accidental or malicious catastrophes, such as the entire memory of the console being wiped out.

Mark Frink is editorial director of Live Sound International.

Church Sound: How Do I Budget For Our A/V/L Needs?

This article is provided by Church Audio Video.

 
The modern church is becoming increasingly more dependent upon technology.

Whether it’s just a simple analog mixer for a children’s area or a full-blown computer-controlled audio, video and lighting system, one thing is for sure: Knowing WHAT you need is not nearly as important as knowing WHY you need it.

Rick Warren’s Purpose Driven philosophy is a powerful reminder that if we’re not careful, we’ll allow the tyranny of the urgent to drive all of our purchases.

Ultimately, A/V/L gear is only a means to an end; it is a tool to help us accomplish our ministry’s vision. Here are some suggested guidelines that will help you determine how to accurately budget for current and future needs:

1) Have a clear and concise understanding of your ministry’s vision. I like to ask the question “if we don’t do anything else, what must we do?” Pastors and elders usually have a clear direction of what they want the church to accomplish in your particular culture and geographical location. Knowing that vision will be a tremendous help to seeing it realized through the tech ministry.

2) Match equipment choices with your specific needs. If your ministry doesn’t require all the features of a particular piece of equipment, weigh the options. Don’t get captured by “it will do this, and this, and this!” from a salesperson. Features are great, but it’s best to only get the features you need.

For example, line array loudspeakers are all the rage in sound systems right now. Line arrays are great for the right room, but if your room doesn’t need them, then the rage might come from your congregation – mismatching a line array speaker system to a room can cause echoes, phase cancellation, comb filtering, decreased speech intelligibility and other unpleasant effects.

Another popular trend is digital consoles. They, too, are great tools, but only if you need their flexibility and have the budget for one.

3) Consider a full-systems approach. As excited as techs can get about a particular piece of equipment, piece-mealing a system can cost way more time, money, and effort that it should. Make sure you know what tools are needed to get the job done, both now and in the future.

Assess where you are and where you’re going, because being a good steward of your church’s resources may require a complete system upgrade or overhaul.

4) Get some help if you need it. Don’t be afraid to consult with someone who is qualified to help you make good decisions. T

here are several ways to get the info you need. Talk with your friends who are techs or worship pastors; they usually have valuable experiences to share.

Secondly, do your research. Read trade magazines like Church Production and use online resources like ProSoundWeb, which offers excellent forums where you can get useful advice from your peers.

Attend trade shows, like the hugely popular WFX Conference (coming to Atlanta this fall), which is a nuts-and-bolts event that provides a wealth of information through seminars, hands-on workshops and exhibits.

5) Be realistic with your expectations. Too often, a budget figure is just pulled out of the clear blue sky without any real basis.

There are three important components of a A/V/L system budget you need to consider:

—The actual “street price” of your equipment (this might be different when you buy a single piece of gear outside of a complete system);

—Shipping costs that can impact the bottom line, particularly on heavy items like loudspeakers, amplifiers, or a large mixing console;

—The technical assistance that is needed to install it. If you can do it yourself, then you’re golden. If your installation requires the help of an on-site integrator, you should find out how much it will cost ahead of time.

6) Use percentages and projections. If you’re in the planning stages of building a new facility, you can usually use a 10-15 percent figure to estimate what your complete audio, video and lighting system will cost.

For example, if you’re building a million-dollar building, then you can expect to pay between $100K and $150K to do it right. If you’re remodeling an existing facility and you have some gear that you can re-purpose, this percentage might be too high.

If you’re building more than a year down the road, then take into account that equipment prices might increase as much as 10 percent in that time based on global market fluctuations and the rising costs of raw materials.

7) Remember WHO you’re ultimately serving. We have a favorite saying about God’s provision: “If it’s God’s will, then it’s God’s bill.” That’s more than just a cliché; the truth is that if the Lord has directed you to do something, then He will make provisions to see it accomplished.

When you do a good job of consulting with others, researching your needs and understanding your goals, then you can trust Him to bring the funds that will see your vision realized.

Jeff McLeod is managing director and a certified church consultant for Church Audio Video.

Church Audio Video specializes in the design, installation and support of high-quality and affordable custom audio, video, lighting, broadcast and control systems for worship facilities. For more information, visit their website.

How It’s Made: Inside FaitalPRO, Driving Loudspeaker Driver Development

Although a relative newcomer to the pro audio world, Faital, headquartered in San Donato, a suburb of Italy’s business capital Milan, has more than half a century of loudspeaker driver manufacturing to its credit.

In 2006, the family-run concern launched FaitalPRO, a division of the company targeting the international pro audio market, which has grown by leaps and bounds since inception, as explained by FaitalPRO overseas sales manager Flavio Naggi, grandson of the company’s founder.

“Although my father is company CEO, my uncle is the president, and my brother is in charge of the sourcing and purchasing department, Faital has outgrown the typical family-size business format and is now an international group, with manufacturing facilities in Italy and fully-owned factories in Hungary and Spain, chosen to ensure fast coverage of the whole of Europe,” Naggi explains.

“We also have sales branches in the U.S., Mexico, France and Hong Kong,” he continues. “Our pro audio division has averaged a 70 to 85 percent annual growth rate, because our range of woofers, compression drivers and horns are appreciated as providing very high quality and long-lasting performance.”

The Faital 86,000 square-footmanufacturingplant in Chieve, Italy. (click to enlarge)

FaitalPRO did not intend shifting manufacturing or R&D to lower cost countries, so decided to focus on the higher end of the market. “We have big momentum in direct product distribution, but, although this gives more rapid gratification, OEM work is a key objective,” Naggi notes. “This takes considerable time to develop, as potential customers must decide to launch a new range of products or have a problem with a current supplier, after which you need to develop the product they need, as they don’t always buy catalog products, and this can take up to two years.

“However, we offer a guarantee of quality, continuity and R&D integrity comparable to top brands on the market, if not higher, because in many areas we have an infrastructure originally specific to our automotive background, guaranteeing the quality of processes, materials used, design and the entire development process.”

Area manager Gianluca Turra adds, “FaitalPRO began with market research to understand what the pro audio industry required for a number of key applications, then the study, concept and design of speakers that could be competitive with or better than those already available and could be produced with our highly automated production methods - able to turn out a woofer every 15 to 20 seconds, or up to 25 with 18-inch models with complex assemblies.”

Left to right, key Faital team members Gianluca Turra (area manager), Mario Passarelli (senior project leader) and Flavio Naggi (overseas sales manager) with some of the company’s drivers. (click to enlarge)

This led to the build-up of the current range of products, with senior project leader Mario Passarelli noting, “In 2008, we were the first to market an extremely high-power subwoofer with a 4-inch voice coil. Prior to our XL Series, subwoofers with 4-inch voice coils couldn’t go to more than 1,000 or 1,200 watts.

“Our extremely long excursion very high-power 18-inch subwoofer in neodymium reached 1,500 watts and beyond, which was quiet an achievement and was a trend followed by many other manufacturers.”

Avant-Garde Facility

Adjacent to the Faital headquarters is the R&D department, the starting point of all the new products and the patented technology adopted by the company.

The specialized staff of over 20 full-time technicians on the R&D team have at their disposal an impressive array of cutting edge systems and software used for the design, validating and testing of components and prototypes, as well as materials used by the company’s suppliers and many of the tools actually used on production lines.

The avant-garde facility also cooperates regularly on joint projects with universities and other bodies.

“We have a series of sophisticated instruments for checking all aspects of the components when they arrive – physical, magnetic, variations due to external influences, such as temperature,” says R&D manager Romolo Toppi. “We must also make certain that materials’ characteristics remain constant, particularly important as far as neodymium magnets are concerned, as there is considerable misconception among suppliers regarding standards.”

Loudspeaker performance is evaluated via acoustic measurements in two anechoic chambers (one fully floating), laser-based assessment, performance with large signals and analysis of geometry and behavior of moving parts.

An entire in-house validation infrastructure enables to carrying out a variety of tests on components, prototypes and end products include corrosion, thermal shock, UV rays and vibration and shock testing, to see how they’ll stand up to use (and misuse) in future applications.

Of particular importance is the capability of guaranteeing that all Faital products will be corrosion-proof, waterproof and capable of withstanding very broad thermal and vibration shocks,
making them environmentally impervious to anything mother nature (or users) will throw at them.

“A great deal of attention goes into developing components that are producible in the most economic manner and able to guarantee performance, but having implemented the strict regulations in other industry sectors enables FaitalPRO to maintain very high quality standards,” Naggi says.

A sophisticated product development process includes 3-D design, extensive prototyping, and evaluation in one of the company’s anechoic chambers.

From the incoming inspection of materials, there are stringent almost “military” level quality control and tests to ensure that products work in the conditions decided upon with clients at the beginning of the program. The company also tests, controls and even purchases the material – such as plastic – used by its suppliers.

Cones are tested on arrival before being mounted on actual loudspeakers, and there’s also a 3-D measurement system to compare components with the original models ordered. End products are also labeled to enable them to be back-tracked down the entire chain.

Highly Productive

Located in extensive tree-shaded grounds in the rural town of Chieve, Faital’s 86,000 square-foot (8,000-square-meter) manufacturing plant, just a half-hour drive from the Milan metropolis, features highly automated production lines designed for extreme flexibility.

Naggi explains: “The design and automation of the lines enables a number of different models to be produced with almost no down-time between job lots, apart from a few minutes required to reset the machines via touch panels, ensuring an extremely high productivity rate.”

Highly automated assembly lines provide precision manufacturing in addition to enabling different models to be produced with almost no down-time.

The facility’s warehouse system is equally streamlined and includes climatized zones for components more sensitive to temperature and a special dedicated adhesive store-room.

The actual production line begins with the assembly of the magnet assemblies, some of which are extremely complex, includes curing chambers that can be adapted according to the type of adhesives (also formulated to Faital specs). Along the line there are cleaning stations to make certain assemblies are absolutely free from unwanted particles (or “crap in the gap” as Turra memorably refers to it).

“Thorough cleaning inside and out before applying dust caps is fundamental, as the air gap is where you have the least space and the most movement, so very little tolerance,” he adds.

Test stations verify aspects such as correct magnetization and component bonding, and although component positioning on the line is almost all auto mated, certain aspects, such as ensuring that for example one part mounted inside another is fully inserted, require an experienced human touch.

Cone application for example is carried out manually, as soft materials are unsuited to robotic handling.

Naggi stresses, “Some manufacturers also apply adhesive manually, but dosage is of fundamental importance, since – as well as looking messy – surplus adhesive adds weight and moving mass plays an important role in performance. Applicators are thus fully automatic, have preset programs for the various speaker models and can apply two (or more) adhesives simultaneously.”

Before packaging, finished products undergo thorough test procedures, starting with a visual inspection and including tests with signals to check physical integrity visually, then computerized tests for reproduction parameters.

An extensive testing process on products covers a wide range of factors such as climate issues, shock/vibration, and much more, to see how they’ll stand up in actual use.

Bold New Directions

Never believers in resting on their laurels, the FaitalPRO team has decided to launch an additional new range of products, based entirely on ferrite magnet technology, but before going into detail on the company’s incredible commitment to this ambitious project, Naggi expresses in no uncertain terms their ideas on neodymium.

“To cut a long story short, we don’t fully agree with the ongoing panic of some of our market’s players regarding neodymium, much of which is caused by incorrect information,” he says. “My opinion - also that of the rest of our top management and sourcing department - is that the situation will not remain stressful for a long period, as there is the opportunity for other countries and other companies to start extracting rare earth minerals from several other sources not currently being exploited, including a very serious project under way at the moment for extraction from the seabed.”

FaitalPRO has adopted a two-fold approach to the situation. One part is to mitigate whatever damage has come from the way the market is behaving, purchasing on average from 200 to 220 metric tons per year of neodymium magnets.

“In the market we’re one of the largest purchasers, which is appreciated by our suppliers,” Naggi explains. “Therefore, we have the capability of minimizing the effect of cost increases for neodymium on the final price of the speaker, thus transferring to our clients as little increase as possible.”

The other part of the approach is a much stronger statement – the creation of a big alternative to neodymium products, on which the entire catalog with the exception of a few small-format models has been based so far – with the launch of an eye-popping 31 new ferrite-based products.

Just a few of the new ferrite driver models currently being rolled out by FaitalPRO. (click to enlarge)

“Although we made our neodymium products competitive with other manufacturers’ ferrite models, there was a slice of the market that wanted ferrite speakers no matter what, so before the end of the year we enter pre-series production on the new products, designed from scratch – new baskets, new magnet assemblies, new everything,” he states.

Ready To Go

The entire development process began in March, and after just nine months of intensive work, a full product range is ready to go. Company officials stressed that FaitalPRO is not discontinuing neodymium, but rather offering an alternative – in fact, it will continue offering all current neodymium models and even add new models next year, alongside ferrite, thus creating a new flow of development, not a substitute.

Turra adds, “This project is not a trade-in at the cost of quality either. In fact, some speakers actually improve with ferrite, since the magnetic field works in a completely different way, favoring some of the features that are particularly appreciated in subwoofers.”

Concluding, Naggi notes, “After lengthy simulation work and a considerable amount of in-depth practical work in the field by our R&D team, we have also devised a very innovative method for cooling the ferrite magnets mounted, so this has been a huge undertaking, a very exciting time for us, and one of the biggest achievements since the inception of the company.”

Based in Italy, Mike Clark is a long-time writer on professional audio topics.

Tuesday, May 08, 2012

Understanding Sound System, Loudspeaker & Room Interactions

If one could listen to only the direct sound of a loudspeaker, the world would be a very different place!

Unfortunately, free field listening, where you have no reflections, room modes or ambient noise, is hard to achieve in everyday life, so we listen to loudspeakers in real rooms.

The interaction of a loudspeaker system and a room can be very complex to understand, model or measure!

One way to measure this interaction is to measure the impulse response of the loudspeaker/room system.

The impulse response of a typical sound system in a room contains lots of interesting information, including:

1) The delay between the loudspeaker and measurement microphone

2) The direct sound-to-reverberent level ratio

3) The time arrival, frequency content and level of reflections of sound

4) The early and late decay rates of the sound

5) The frequency response of the direct sound.

This last point is particularly interesting. The question is “What do we want to measure and why?”

Figure 1: The impulse response of a 1250 seat multi-purpose hall. The x-axis is time (~0.75 sec) and the y-axis is magnitude in dB. Note the direct sound, reflections, the reverberant decay and the noise floor.

One question that goes to the heart of “system” measurement and optimization issues is “If the impulse response contains the frequency response of the direct sound, can we separate the loudspeaker response from the room response?”

Also “If we can, do we want to?”

Figure 1 shows an impulse response displayed in the time domain.

The “spike” that represents the direct sound actually contains the frequency and phase information about the loudspeaker.

To see this information we must transform this portion of the impulse response into the frequency domain.

Figure 2: The impulse response of a 1250 seat multi-purpose hall. The vertical lines suggest a time window that ignores most of the effects of the room at frequencies whose periods are longer than the time window (i.e. low frequencies).

To achieve this isolation of the direct sound from the room response, we must select a time window that includes the direct sound but excludes the reflections and decay of the room.

Figure 2 displays such a time window. This measurement was made using a full range loudspeaker system with the microphone approximately 60’ from the loudspeaker.

Pink noise was used as a reference signal and the impulse response was calculated using a 512K FFT (although only the first ~0.75 seconds are shown).

We can take the “time windowed” data and transform it into the frequency domain using FFT mathematics.

This transformation yields a result that shows how much energy is present at each frequency, as shown in Figure 3.

You can see the pronounced roll-off of low frequency energy. You can also notice the lack of LF resolution in this figure.

The lack of resolution at LF is offset by a excess of HF resolution.

This uneven resolution between LF and HF energy is the result of the FFT mathematics used to transform the data from the time domain to the frequency domain.

Standard FFTs yield data that is distributed linearly in frequency (one data point every X Hertz).

Unfortunately, humans perceive frequency logarithmically.

Figure 3: The frequency response of the direct sound portion of an impulse response of a 1250 seat multi-purpose hall. The response was calculated using a 512 point FFT (which equals a 512/48000 or ~11 msec). As you can see the frequency response shows a pronounced LF roll-off.

This lack of LF resolution in Figure 3 is a direct result of the use of a short time window in our transformation from the time domain to the frequency domain.

It is interesting to note that this plot does not correlate with what we hear.

Simply listening to the full range loudspeaker system we were measuring made it clear that the system was reproducing LF energy down to at least 100 Hz!

I would suggest that a primary goal of an effective measurement system should be to provide results that correlate well with what we hear.

So the lack of correlation between what we have heard and what we measured suggests a modification to our approach.

As an alternate approach to trying to find a measurement that correlates with what we hear, we can try using a longer time window to “see” the LF response with better resolution.

A longer time window of approximately 250 msec is shown in Figure 4.

Figure 4: The impulse response of a 1250 seat multipurpose hall. The vertical lines suggest a time window that INCLUDES most of the effects of the room. The time window shown is approximately 0.25 seconds.

To transform this longer “slice” of the impulse response into the frequency domain, we will use an 8k FFT which represents 8k/48000 seconds, or 0.171 seconds.

Notice again that this time window includes both the direct sound and the response of the room.

In Figure 5 the low frequency information is seen in adequate resolution, however the high frequency results look confusing. The plot shows data that has 5 Hz resolution (i.e. one data point every 5 Hz).

While this resolution provides excellent LF resolution (between 31 Hz and 62.5 Hz there are 15 data points.

However at HF we have excessive resolution - between 4 kHz and 8 kHz there are approximately 800 data points.

Simply stated, the longer time window provides good LF resolution, but excessive HF resolution.

The result of studying these plots might lead you to conclude that in order to make measurements that correlate well with our listening experience, we must use very short time windows that isolate the direct sound at high frequencies, and increasingly longer time windows as we look at lower frequencies.

At first glance this idea might seem to violate the often quoted phrase, “One can only affect the direct sound with processing.”

However this is not the case. At mid-low and low frequencies, the interaction of a sound system and a room can be affected and optimized by signal processing.

In other words, at low frequencies (long wavelengths) the direct sound and reflections from nearby surfaces combine to form a composite response. It is this composite response that a listener hears.

The ability to measure several time windows simultaneously provides a measurement that both correlates well with human hearing and provides insight into how the signal being sent to the loudspeaker can be tailored (via equalizers, or other processing) to optimize the loudspeaker/room interaction.

Figure 5: The frequency response of the direct sound portion of an impulse response of a 1,250-seat multi-purpose hall. The response was calculated using a 8192 point FFT (which equals a 8192/48000 or ~107 msec). As you can see the frequency response shows low frequency energy that is much more pronounced than seen with the shorter time window.

Our last figure shows a measurement of a loudspeaker system that includes multiple time windows and displays both the magnitude and phase response of the “system.”

The use of multiple time windows allows one to isolate the direct sound of a loudspeaker in a real-world situation at high frequencies.

However, at lower frequencies, longer time windows that include the loudspeaker/room interaction have been found to correlate well with our listening experience.

Multiple time windows in a single measurement is an extremely interesting way to measure and optimize the response of a sound system in a room.

Sam Berkow has completed a wide variety of acoustical design projects including: concert halls, recording studios, broadcast facilities, production facilities, house of worship facilities, large multi-purpose venues, amphitheaters and stadiums. His educational background includes a masters degree in Engineering from the Stevens Institute of Technology, where he specialized in acoustic measurement and design. He is also the original developer of Smaart acoustic measurement & system optimization software.

The Top 10 Things That Can Never Be Taught Often Enough In Audio

10. Musicians feel most comfortable and play best when they hear what they need to hear on the stage. Of course, the experienced monitor guys and recording guys already know this.

But it’s something for those less experienced to think about. No, it’s not about how much power you have or what kind of monitor wedges. It’s about psychology.

And I think it’s true that if you become good at monitors and understand how to please musicians, you are 90 percent there towards becoming a good mix engineer.

Sure, the last 10 percent might be the “magic” but you can’t make magic without the basics.

9. Sound travels at 1,130 feet per second, at sea level, at 68 degrees F and 4 percent relative humidity. This is important because if you understand how sound propagates, you’ll automatically know more about microphone placement, setting delay towers, and things like delaying the mains to the backline. And you should also know that the speed changes with temperature, humidity, and altitude. (If you don’t, it’s a good idea to look it up.)

8. The Inverse Square Law. You know, the thing about a doubling of the distance from the source means that the acoustic power is cut by 1/4, right? This applies all over the place, from mic technique to loudspeaker arrays. It relates to how much power you will need from the power amplifiers.

The Inverse Square Law, illustrated. (click to enlarge)

For instance, if you normally cover an audience at 20 to 60 feet from your stacks, but for the next gig, the audience will be 40 to 100 feet away, how much more power will you need to maintain about the same acoustic power? About four times as much! Maybe think about delay stacks (see #9).

7. The equal loudness contours of the human hearing system. Back in the 1930s, Harvey Fletcher and his team at Bell Labs did some tests resulting in this graph. “What it means is that the human ear is most sensitive in the upper mid-range frequencies, and least at very low and very high frequencies.

In other words, to hear a tone at 100 Hz equally as “loud” as one at 3.5 kHz, it must be 15 dB louder! (This is assuming the 3.5 kHz tone was at 85 dB SPL),

Fletcher and Munson were preHy cool for being complete audio geeks. (click to enlarge)

The implications are that to provide a really good, full mix, you need a combination of lots of power, a carefully designed subwoofer system, and a brain capable of realizing that it’s easy to overpower peoples’ ears in the mid range. Not to mention that distortion, especially in the mids, is really obnoxious. Which brings me to…

6. Distortion really sucks, unless part of the “sound”. I’m often amazed how seemingly intelligent people who otherwise know what they’re doing don’t seem to notice fairly high levels of unintentional distortion.

The first step in eliminating it is to learn about the causes, from gain structure to faulty connections to cold solder joints to aging tubes. Second, and perhaps even more important, is to learn how to hear and identify distortion. Is it harmonic distortion? Gross overload of one of the channels? Intermittent?

And then third, please do something about it!

5. Sound from the same source can combine acoustically out of phase to give a “bump” of +3 dB (doubling of power) all the way to complete cancellation, or minus infinity.

For instance, the nodes (bumps) and modes (cancellations) of bass signals—due to standing waves in the room—can actually completely cancel in some places. Zip, zero, nada.

And let’s think what happens if you place your RTA mic in that spot and take a reading. There will be a big notch in the lows at a particular frequency.

This is one reason why it is important to take a multitude of measurements before getting a real understanding of the bass in a particular room.

4. What wasn’t captured upstream at the mics or pickups can’t be recreated downstream.

Sure, plug-ins, effects, DSP and all that stuff can be cool and is often needed to create the sound you want, but the idea that an SM57 can be made into a U47 via “mic modeling” is absurd.

Distorted sounds can’t be fixed (see #6). Or as computer geeks stated in the 1950s: “garbage in, garbage out”. This is not to imply that an SM57 is garbage. To the contrary - it’s a great mic that has a ton of uses. But there are certain things it does not capture and no amount of processing can change that after the fact.

An audio chain is only as strong as the weakest link upstream.

3. The sound of the performance should match the music of the performance, simple as that. Glenn Miller music should not sound like rock. Rock should not sound like classical. Classical should not sound like there is a sound system present.

Audio signals combined out of results in comb filtering. (click to enlarge)

Most often, it’s a good idea to go to the original source to understand how a mix should sound. If the “original” is a record, then it’s a good idea to figure out what effects were used on that record, how it was EQ’d, and the overall “vibe.” If the original is a live performance of acoustic instruments and voices, well then check it out and learn how it is “supposed to sound.”

2. Grounding. Let’s not mince words here: this is a subject you need to understand. If you have more than one path to ground in your audio system, and the resistance to ground is different between them, you will have problems with hum and buzz.

Related to this is how you terminate your connections, especially if any parts of the system go back and forth from balanced to unbalanced terminations.

It’s also a good idea to learn the sonic signatures of different kinds of hum and buzz to therefore speed up your troubleshooting when the time comes. This is because some types of buzz are not related to grounding problems, but instead may be power supply issues, for instance.

1. Gain structure, baby. This is the main one, the real deal. The thing that, if you can’t learn, or don’t understand or have forgotten, will get you into more trouble than anything else. There will be more noise and/or more distortion in the system unless you get this right. And there will be less gain before feedback, too.

So here’s the deal: every input and every device has an optimum range of levels it wants to see or wants to work with. If you’re feeding something a signal that is too low, you have to make this up somewhere, and therefore you’ll be bringing up the noise more than it should be. And that noise will be in your signal from then on.

Oh, sure, there are noise reduction devices you can use, but why do that when proper gain structure will take care of it for you? And really, we should use the least processing possible to get the job done because things sound better that way.

Alternatively, if you an input or a mix bus is fed too much signal, headroom will run out, which means you’re adding distortion. And this, also, cannot be removed later. Artistically adding distortion via plug-ins, hot-rodded guitar amps or certain outboard gear can be cool. Adding it by slamming your inputs or your mix bus is not cool.

For instance, if a wireless microphone output can be set at line level, but you set it to mic level and connect it to a mic input on your mixer, you will have more noise than if you connect the line output to the line input. Why? Because essentially you’re padding down the output then boosting it back up again with a high-gain mic preamp.

Sure, sometimes you might want to put the signal through a transformer or other “good” distortion device—just be aware that from a gain structure point of view, this is not ideal.

OK, that’s the list. If you’ve already mastered these things, great! You’re probably doing better mixes, with more gain before feedback, better coverage and happier musicians than those who don’t. But please don’t rest on your laurels - get out there and learn as much as you can.

Those of us going to your shows will know it when we hear it!

Karl Winkler is director of business development at Lectrosonics and has worked in professional audio for more than 15 years.

Church Sound: Locating Your Loudspeakers & Related Issues

The decision on the location of your sanctuary main and monitor loudspeakers will have a decided impact on the success of your presentation.

A single source of sound is best for the spoken word, whenever possible.

In a perfect world, as it relates to audio systems for worship, it’s best practice to place the sanctuary main loudspeakers in a central cluster above the front edge of the chancel riser.

The loudspeaker (or loudspeakers) are selected to provide pattern coverage over the entire seating area without putting acoustic energy on the walls, floor or ceiling.

When we put sound on people, it is largely absorbed and only minimal reflections continue elsewhere in their journey about the room.

When the pattern coverage is poorly designed, putting acoustic energy on highly reflective surfaces such as walls, floors and ceilings, the reflected sound can pass the listener’s ears several times, creating a lack of enunciation and speech intelligibility.

A properly designed central cluster allows the sound to reach the listener only once, thereby creating the most concise possible listening situation.

In many sanctuaries, however, there are physical limitations such as low ceilings or tall crosses that require an alternate consideration.

What if we can’t use a central cluster?

When forced to consider an alternate placement, the choice is usually left side and right side. It’s important to remember that sound will arrive at two different time intervals to people seated along the sides, and so we must attempt to select loudspeakers with a narrower coverage pattern.

The goal is to put sound on people at the left with the left speaker, and on people at the right with the right speaker, with as little acoustic energy crossing over the middle as possible.

How high should the loudspeakers be hung/flown?

Generally speaking, loudspeakers should be flown as high as possible (however, not to exceed 18-22 feet) in order to increase their distance from the front pew.

If the room has extremely low ceilings, we can arrive at a condition where people seated at the front are complaining that it is too loud, while the people at the rear are commenting that the sound needs to be turned up.

In such an instance, it’s advisable to turn the system down to a comfortable level and hang a second and even third set of loudspeakers perhaps every 25-30 feet as we grow in distance from the chancel.

Because sound traveling through the air takes time, the second set of loudspeakers will need to utilize a time delay so that the sound traveling from the chancel coincides perfectly with the sound emanating from the second set of loudspeakers.

A third set of loudspeakers will have to be delayed at yet a different setting to coincide with the sound emanating from the first two sets of loudspeakers.

In this manner, all sound source material reaches the ears of the listener at the exact same moment in time, regardless of how far back they are seated in the room, thereby maintaining speech intelligibility.

Though a sanctuary may have adequate ceiling height, if the room is very deep it’s still advisable to use multiple loudspeaker placements on delay lines.

Even if the chancel mains could be turned up loud enough to be heard at the back of the room, the sense of distance is audible (due to wall and ceiling reflections) and intelligibility is again adversely affected.

How can we minimize the possibility of feedback?

Despite the general public’s degree of sophistication in regards to quality audio, it’s not commonly understood that microphones need to be out of the live sound field whenever possible in order to minimize the possibility of feedback and annoying lingering overtones.

In other words, keep loudspeaker enclosures in front of the mics, not behind them. Of course, almost all pastors wear wireless mics, and many like to move about the room while speaking. A good church sound operator will be able to provide equalization so this may be done.

Attempt to keep monitor sound confined to the chancel riser.

Monitor loudspeakers are a wonderful benefit for the performers using them, but they can have a deleterious effect on the sanctuary sound.

If the monitors are positioned so that the monitor mix bounces off the back of the chancel and reflects back out to the congregation, it’s now combining at a different time interval with the sanctuary main mix and we have now adversely affected the speech intelligibility we had been striving so hard to create out front.

How loud should the monitors be?

Monitors should be just loud enough to keep the performers comfortable. If the monitors are too loud in relationship to the sanctuary main loudspeakers, no amount of positioning will help maintain clarity in the general seating area.

Since many praise band players are now middle-aged veterans of once-youthful rock bands, gently remind them that the purpose of the monitor line is to lend support and enunciation so that they may execute the material more perfectly.

If the monitors are intended to provide a studio-perfect mix of all instruments and voices for the listening enjoyment of the players, then you will need to be blessed with highly experienced and adequately funded audio technicians. Many larger churches in metropolitan areas are able to create this benefit for the praise musicians.

Jon Baumgartner is a veteran system designer for Sound Solutions in Eastern Iowa, a pro audio engineering/contracting division of West Music Company.