Thursday, October 28, 2010
Tech Tip Of The Day: Live Sound Systems And The Noise Floor
Is considering the noise floor necessary when purchasing PA components?
Q: I want to put together the best live sound system possible for my band, choosing mics, a mixer and amps with the widest possible dynamic range, highest signal-to-noise ratio, etc.
But how do these specs apply in a real live club setting?
How do they affect overall volume?
A: While component specs such as self-noise, dynamic range and signal-to-noise ratio are useful in selecting gear, in a live sound environment they are usually outweighed by a major outside factor - the noise floor of the room itself.
This is a slightly different definition of the term “noise floor.”
Every room, whether it’s a club, a church, a huge arena or anything else, generates noise. One of the most common sources is the whoosh of air conditioning units, often combined with whine from the fan motors.
In a club, the inescapable sound of conversational voices becomes part of the noise floor, as do the clinking of glasses, sounds from the bar, and the click of billiard balls. Ultimately you can end up facing a noise floor of 40-50dBA.
So it’s normally the environment that determines both the dynamic range and signal-to-noise ratio of a sound system, as far as the audience is concerned.
Since most electronic components in the system have a dynamic range on the order of 100dB or more, the sound system itself should never be the weak link when it comes to the end result.
Only in a studio should the equipment noise floor become a factor in determining either the dynamic range or signal-to-noise ratio at the listener’s position.
In your sound system, the noise floor would be established by the ambient pickup of the open microphones. In a room with a 40dBA noise floor, the signal-to-noise ratio of a typical vocal microphone will be limited to the mic’s ability to exceed that noise floor.
If a strong vocalist produces 120dBA into a hand-held mic (not unusual for close-miked vocalists), the signal-to-noise ratio would be a healthy 80dB, since the 120dBA - 40dBA = 80dB.
Remember that each open mic increases the noise; ten open microphones could increase the noise floor by another 10dB if their sensitivity and level setting were the same as the original mic (since the 10 log (# open mics) = 10dB).
Unfortunately, the signal-to-noise ratio of the system cannot exceed the worst-case condition at the open microphone since the sound system has no choice but to amplify the room noise along with the desired signal.
Distant miking (such as drum overheads) and failure to mute unneeded mics can cause audible problems with your signal. This is why good mic technique is essential for good system performance, as it ultimately establishes the signal-to-noise ratio of your system.
So what’s your goal - to just be able to compete with ambient noise or to annihilate it? Get an SPL meter and take a reading of room noise.
If your output signal exceeds the noise floor by at least 25dB you’ll establish yourself nicely without too much risk of being too loud. Of course, it’s rather easy to move well beyond this level.
As always, we welcome input from the PSW community and would love to know your thoughts on the issue at hand. Feel free to let us know in the comments below!
For more tech tips go to Sweetwater.com
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Soundcraft Vi6 Digital Consoles Amplifies Pope’s Public Mass In Scotland
SSE Audio Group provided multiple Soundcraft Vi6 consoles for FOH and monitors.
With a crowd of 75,000 followers, Pope Benedict XVI’s voice during mass at the Bellahouston Park in Glasgow, Scotland was entrusted with SSE Audio Group’s deployment of two Soundcraft Vi6 digital consoles.
With a 35-piece orchestra, 800-piece choir and pre and post-show concerts, the religious event required reliable and vast amounts of digital channels to ensure no note or prayer was unheard.
With years of personal use and high-quality results with Soundcraft consoles, SSE Project Manager, Dan Bennett is well-acquainted with the Vi6 digital consoles. “Soundcraft is personally becoming a favorite of mine for specs,” he said.
“The amount of channels available with the Vi6s were perfect for this type of event and the surface features ensured the utmost quality for the performance.”
Under tremendous pressure dealing with inclement weather, narrow deadlines and organizing a concert-style sound system for the Pope, SSE utilized one Vi6 for the monitors and another Vi6 plus a Vi1 console for the FOH management.
It was vital to the organizers that the seating area sounded perfect for the Pope’s service.
“We were able to keep the large stage and all of the entertainment under control with the Vi6s as if it were a large concert, while still offering a peaceful worship service environment for the Papal mass itself,” said Bennett.
“Everything was absolutely perfect and we couldn’t have asked for a better job all around from the team or the Vi6s.”
The Soundcraft Vi6 digital console uses a derivation of the Studer Vistonics touch screen user interface to allow engineers to operate the desk intuitively. With 96 mono inputs and 35 outputs, the console is one of the most comprehensive and detailed, yet simple-to-use boards in the industry.
SSE purchased the Vi6 consoles through Sound Technology, Soundcraft’s UK distributor.
Soundcraft Website
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TC Electronic LM2 Loudness Meter Makes Stateside Debut At AES 2010
With the LM 2 TC Electronic has taken on a important leadership role in the global loudness control concern.
TC Electronic will be bringing its LM2 stereo loudness and true-peak level meter to the U.S. market by debuting the advanced technology at this year’s AES 2010.
The LM2 meter enables unprecedented levels of loudness precision and quality in audio applications, eliminating level jumps and other aural inconsistencies for polished sound.
With this new product, TC Electronic is taking an important leadership role in the worldwide concern for loudness standardization.
The new LM2 is one of the first products on the market today that complies with the European R128 loudness standard, as well as U.S. standards.
The meter analyzes any audio, be it speech, music or other sources, assigning it an ATSC A/85- or EBU R128-compliant loudness number. Numbers may be used to normalize programs, commercials and music tracks, and to set metadata in AC3 transmission.
This eliminates level jumps and other inconsistencies sometimes caused by human error. For example, after mixing for many hours, a user’s ears will get tired, making it harder to determine loudness levels. With the LM2 meter, the user can rely on an exact number as a reference for the mix instead of his ears.
Similarly, using the LM2 as part of a monitoring signal path ensures a consistent loudness level in the mixing environment.
“Accurate loudness level is not only an extremely hot topic right now for both the worldwide professional audio and broadcast markets, but it’s also absolutely critical to any production,” said Thomas Lund, HD Development Manager, TC Electronic.
“Until now, few measurement products were capable of taking all the guesswork out of loudness metering. TC Electronic, however, has designed the LM2 meter to offer loudness reference points based on algorithms, not just the human ear.”
“This ensures a completely consistent level of loudness standardization across a production, offering audio professionals tremendous reliability and peace of mind.”
Users can view the loudness numbers generated by the LM2 on the meter’s front panel or Stats display. Connecting the LM2 to a PC or Mac via USB allows access to TC Electronic’s patented “radar meter” technology, which displays loudness over a given period of time. The radar can show loudness data from up to 24 hours back in time, even if there was no connection to a computer during that period.
The LM2 meter is intended for a variety of broadcast audio applications as well. During ingest, it can be employed to measure loudness and the “true-peak” level of incoming audio signals, revealing any signal overloads.
Built-in gain normalization enables it to correct gain to a pre-set loudness level, while a 48-bit precision limiter ensures that if the gain has been positively invoked, there will be no overloads.
Pre-transmission, the LM2 can log the outgoing loudness level of the broadcast station for a full week. Detailed log files may be imported into Excel, and LM2 needs no connection to a computer besides from when log files are dumped. Post-transmission, LM2 can be used to monitor and log what is sent out.
The LM2 meter is compliant with ITU-R BS.1770, ATSC A/85, EBU R128, NABJ, OP-59, BCAP and many other guidelines.
It offers a wide variety of 24-bit resolution audio inputs and outputs, including AES/EBU, TOS, SPDIF/AES3 id, ADAT and analog. Digital I/Os are fully synchronous while analog I/Os are scaled in the analog domain for maximum utilization of converter dynamic range. Analog inputs can be trimmed at 0.01dB precision.
“TC Electronic is not new to delivering outstanding loudness technology options,” adds Lund. “With the LM2, however, TC Electronic has brought itself into a leading position in regards of loudness technology, able to offer unprecedented precision and quality to the professional audio and broadcast markets.”
TC Electronic Website
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Kramer Introduces VA-2H HDMI EDID Reader-Emulator
The VA-2H is an advanced Extended Display Identification Data capture and emulation tool for accurate connection to sources.
Kramer Electronics has announced the introduction of the high-resolution VA-2H EDID Reader-Emulator.
The VA-2H is a diagnostic and debugging tool for installers working with HDMI devices.
The unit can switch three different EDIDs to the HDMI input and has a USB connection and a software editor, allowing the installer to manipulate various parameters of the EDID.
The VA-2H supports up to 2.25Gbps bandwidth per graphic channel, and supports high-resolution graphics up to 1080p in video and up to UXGA in computer graphics video.
The VA-2H can store and recall a default EDID setting in non-volatile memory allowing convenient and accurate connection to any HDMI source.
EDID (Extended Display Identification Data) is 256 bytes of data that a display provides to the graphics card of a connected computer source.
The EDID describes the display’s capabilities so that the source can output the best possible signal for that display. The source and the display exchange this information over the Display Data Channel 2 (DDC2) using pins 15 and 16 of a standard HDMI connector.
The VA-2H ensures that if a display is turned off, temporarily disconnected or is out of communication with the source, the source continues to output the best possible signal resolution. The EDID emulation makes the source think that the display is still directly connected even when the EDID information is unavailable directly from the monitor.
Many systems including matrix switchers, and distribution amplifiers, for example, are unable to pass EDID information from the monitor to the source.
In these instances the VA-2H captures the EDID information from the display and provides the EDID communication to the source.
Note that an HDMI cable must be used that passes all 19 pins between the source and VA-2H. This allows the source to output the best possible signal for that display device.
The VA-2H is HDMI 1.3, HDCP 1.1, and DVI 1.0 compliant. This unit also comes equipped with Windows- based application software that enables the user to edit and save EDID settings, and create custom EDID timings.
The VA-2H sells in the United States at a list price of $445.00 per unit. It comes housed in a compact DigiTools enclosure, with a 12V power supply. The VA-2H is currently in stock and available from Kramer Electronics sales companies around the world.
Kramer Electronics Website
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Wednesday, October 27, 2010
Power Lines: Getting The Proper Flow
Factors that affect power transfer between an amplifier and transducer.
There are a number of possibilities for rating a power source.
The most straightforward is a simple rating for continuous available power, a product of the maximum voltage (pressure) output times the maximum available current (flow) from the amplifier.
This “Volt-Ampere” rating represents the largest number that could be used to characterize the amplifier.
For the “ideal” amplifier, we could simply multiply the DC rail voltage times the maximum current available from the power supply into a purely resistive load.
Unfortunately, the only value of such a rating would be its affect on retail sales. This large number must be de-rated in light of the actual conditions under which amplifiers must operate.
So what factors serve to reduce the “ideal” output power?
First, no sound reinforcement amplifier is called upon to deliver DC voltage and current into a load. If such a signal were indeed applied to a loudspeaker, it would quickly (and silently) burn up!
Loudspeakers only vibrate with the application of alternating current, or AC. We could pulse the DC current, producing a square wave, but this still would bear little resemblance to a real-world audio waveform. Most amplifiers are rated based on their ability to pass a sinusoidal waveform.
Figure 1: A sine wave plotted as a function of time.
The sine wave, when applied to a loudspeaker, makes it move in and out like a piston. Sine waves are discrete in their spectral content, meaning that they contain only one frequency.
They are also the building blocks of more complex waveforms. Since the typical sound system must pass many thousands of frequencies, it would take a lot of sine wave testing to fully characterize an amplifier!
One possibility is to rate the amplifier using a 1 kHz sine wave, and then add an additional descriptor for how much this will vary over the bandwidth of the amplifier.
A power bandwidth of -3 dB would mean that the guaranteed power output of the amplifier is one-half of the 1 kHz rating. That seems like a big difference, but it’s not when you consider the logarithmic characteristics of human hearing. One-half power is just noticeably lower in sound level.
So let’s go with the sine wave for rating the amplifier. A 500-watt continuous amplifier will deliver 500 watts to my loudspeaker, right?
Well, only if you play sine waves through it! In reality, the power flow will be much lower. This is due to the complex nature of the audio waveforms generated by real-world program sources.
The waveforms produced by a drum kit or lead singer bear little resemblance to sine waves. They are inherently more complex. Before we take this thought further, let’s look at some characteristics that describe time-varying voltages.
It’s About Time
Figure 1 shows a sine wave plotted as a function of time. This statement means that the amplitude (vertical displacement) of the voltage is time-dependent.
Like the stock market and the air temperature, the question of “how much?” depends on “when?” So what value do we assign to the voltage and, ultimately, the power generated by the sine wave?
One possibility is to use the maximum displacement from zero. This is the peak voltage of the waveform, and it represents the largest value that we could give it. Unfortunately, the peak voltage has little to do with loudness or power.
Earlier we showed that power generation is a time-dependent parameter. A peak can have high amplitude but not last long enough to produce much power flow. Power is ultimately tied to the root-mean-square (RMS) value of the waveform.
The RMS voltage can be thought of as the “area under the curve” described by the waveform. It is numerically equivalent to the DC voltage that would generate the same heat for the specified time interval. This “heating value” is what must be considered to assess power flow and loudness.
RMS voltage is determined by squaring all amplitude values (this makes them positive), and then taking the square root of the mean (average) value.
Figure 2: A lower RMS voltage means a higher crest factor, which in turn means less power delivered to the load.
Power is then calculated by squaring the RMS voltage and dividing by the resistance of the load, the result being termed “continuous average power” or just “continuous” for short. For a sine wave (and a sine wave only), the RMS voltage is 0.707 times the peak voltage.
This means that a sine wave will generate one-half of the power that the DC rail voltage would for the same time span.
Using the “hypothetical” DC rating as a reference, we would say that the sine wave has a “crest factor” of 3 dB. So, an amplifier with a sine wave rating of 500 watts continuous could be rated at 1000 watts “peak” output. So why not rate it at 1000 watts?
There are several reasons, including the fact that most amplifiers cannot sustain such an output power for any appreciable time period. It is also unlikely that any real-world audio waveform would stay at the peak voltage for any appreciable time span.
So, when you bring “time” into it, peak values become less meaningful. And since power is the “rate of doing work”, it is impossible to consider power independent of time.
Also, just like passing your finger quickly through the flame of a candle, loudspeakers and amplifiers can be given very large heat generation and dissipation ratings if the time element approaches zero.
Room At The Top
Real-world audio waveforms have much lower RMS values (less area under the curve) than sine waves do, even though their peak values may be the same (see Figure 2).
A lower RMS voltage means a higher crest factor, which in turn means less power delivered to the load.
One way to determine how much power a complex waveform will generate is to use the peak power output based on the DC rail voltage as a reference, and then subtract the crest factor of the complex waveform from it. This will yield the power generated by the complex waveform.
For example, an amplifier rated at 500 watts continuous average power with a sine wave will have a theoretical peak output power 3 dB higher (the crest factor of the sine wave) - 1,000 watts in this case.
Using this as a reference (that’s all it’s good for), we can subtract the crest factor of the real-world waveform. Refer to the following chart for some common decibel relationships:
0 dB - reference value
-3 dB - one-half power
-6 dB - one-fourth power
-10 dB - one-tenth power
-20 dB - one-one-hundredth power
These numbers will be useful for determining how much power a waveform generates based on its crest factor.
Our 1,000-watt “theoretical” amplifier would only generate 500 watts for a sine wave signal (-3 dB), 250 watts for a 6 dB crest factor signal (highly compressed music or speech), 100 watts for a 10 dB crest factor signal (slightly compressed music or speech), and only one watt for a 20 dB crest factor signal (“raw” music or speech).
So, in the real world of live performances, a 500-watt “sine wave rated” amplifier is likely to deliver only a fraction of this power to the loudspeaker.
The implications? First, if a loudspeaker can safely dissipate 100 watts continuous average power (based on destructive testing - sounds like fun, doesn’t it?), then the required amplifier size to deliver 100 watts will be considerably higher.
This is because the amplifier is rated using a sine wave, which yields a much higher power output than a real-world audio waveform.
Assuming a crest factor of 10 dB, the amplifier would have to have a peak rating of 1,000 watts (sine rating of 500 watts) to actually deliver 100 watts into a resistive load.
This “extra” room is called “headroom”. This is why it is common practice to oversize the amplifier relative to the loudspeaker’s power rating.
But if you connect a 500-watt amplifier to this loudspeaker, and someone feeds it a low crest factor signal (like a sine wave), it is likely that the loudspeaker would burn up since its power rating has been exceeded.
So, we want a large amplifier to provide sufficient headroom for high crest factor signals, but we need to make sure that a low crest factor signal is not applied that will burn up the loudspeaker.
Catch 22. No free lunch. It depends. There is no escaping these realities in the real world of live performance.
Let’s sum it up with some guidelines. The sensitivity rating of a loudspeaker specifies a sound pressure level that will exist at a one meter distance when one watt of electrical power is applied to the loudspeaker’s terminals (2.83 VRMS across eight ohms).
In other words, if you put one watt in, it will be this loud at this distance. The amplifier size that is required to deliver one watt is dependent on the crest factor of the program material.
Using 10 dB as a rule-of-thumb, the required peak power rating would be 10 watts. So, it takes 10 watts peak to get one watt continuous!
If we scale these numbers up to a loudspeaker rated to dissipate 100 watts continuous, the peak amplifier rating would be 1000 watts.
Given that amplifiers are typically rated with sine waves, the continuous power rating of the amplifier would be one-half the peak rating, or 500 watts.
Figure 3: A meter that can display the peak and RMS voltage of the waveform at the same time.
Therefore, it takes a 500-watt sine wave rated amplifier to deliver 100 watts to a loudspeaker if the program material has a 10 dB crest factor. How’s that for confusing?
Crest Factor Awareness
So, how do I determine the crest factor of my program material? After all, this is what ultimately determines the required power rating of the amplifier and the power flow into the loudspeaker. There are two ways to approach this.
For recordings, the crest factor can be calculated quickly and accurately using a wave editor program. Open the wave file, select a time span, and look for a menu selection called “statistics” or something similar. This should display the crest factor for the time span selected.
For live music, it’s a bit harder. You need a meter that can display the peak and RMS voltage of the waveform simultaneously (see Figure 3). On such a meter, the crest factor can be monitored in real-time - very cool.
If the meter is calibrated so that its highest peak indication is the clipping point of the system (this is the right way to do it), then the lower (RMS) indication on the meter will correlate with how much power is being generated. This allows the operator to turn the system down if the RMS gets too high (which is also the right way to do it).
The third way is to use some rules-of-thumb for typical crest factors. Once you factor in all of the variables (I won’t do it here), the crest factors in live performance are often in the 6 to 10 dB range for mid and high frequencies.
They tend to be lower at low frequencies where synth and bass guitar signals look more like sine waves. This means that at mid and high frequencies you will be using one-fourth to one-tenth of the peak rating of the amplifier. At low frequencies you will be using up to one-half of the peak rating of the amplifier.
If you are a disciplined sound operator, you can oversize your amplifiers by this much (relative to the loudspeaker’s rating) and probably not get into trouble.
The safest overall approach is to use large amplifiers, but then monitor the program’s RMS and peak levels to stay within the loudspeaker’s limitations. Also remember that compressors and limiters reduce the crest factor of the signal, which means that more power is delivered to the load. Again, real-time monitoring will tell the story.
The Conclusion?
The bottom line is that there are many variables that determine the power flow from amplifier to loudspeaker. An understanding of these basics can allow us to stay within the operating limits of our hardware. Power ratings are meaningless when there is smoke coming out of the loudspeakers.
This series of articles hasn’t attempted to present a black-and-white recipe for amplifier/loudspeaker selection. What it has done is put the variables on the table that the system designer must consider when selecting components. There’s a lot to think about!
Pat Brown teaches the Syn-Aud-Con seminars and workshops. Synergetic Audio Concepts (Syn-Aud-Con) has been a leader in audio education since 1973. With nearly 15,000 “graduates” worldwide, Syn-Aud-Con is dedicated to teaching the basics of audio and acoustics. For more information visit their website.
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Lectrosonics Introduces Belt-Mounted Mute Switch
The new mute switch provides pop-free audio muting for company’s belt-pack transmitters.
Lectrosonics has announced the introduction of the Mute switch.
Designed for use with the company’s belt-pack transmitters, the Mute switch is a belt-mount unit that provides instant, pop-free audio muting.
The Mute switch is an active device that is powered by any 5-pin Lectrosonics transmitter, including the new Servo Input transmitters such as the SM Series, LMa, and UM400a.
It works with most 2- and 3-wire lavaliere microphones wired according to Lectrosonics guidelines.
The new Lectrosonics Mute uses an optically coupled switch to silence the audio signal without any clicks and pops, even when located in a strong RF field.
The unit’s toggle switch is weather resistant and the included cable has weather resistant vinyl boots at each end.
A wiring kit is included that provides a 5-pin connector, strain relief parts, and an additional water resistant vinyl boot to protect the lavaliere microphone connector from moisture.
With its weather resistant package, the Mute switch makes an ideal choice for outdoor sporting events. The housing is machined aluminum, powder coated, and laser engraved for ruggedness and legibility.
The toggle switch is conveniently sized and can easily be controlled under garments. The reversible, stainless steel belt clip enables it to be adjusted to the user’s preference.
“The new Mute switch is an indispensible accessory for our 5-pin transmitters,” commented Karl Winkler, Lectrosonics’ Director of Business Development. “This is the perfect solution for muting the audio feed of referees, ministers, or others who require the ability to either temporarily silence or temporarily engage their microphones.”
“The unit’s robust design protects it from moisture and the housing is specifically built to withstand the rigors of outdoor use. I’m confident many sound professionals will find this an invaluable accessory.”
Lectrosonics Website
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Crown Audio Calculators: Inverse Square Law
A useful calculator for designing audio systems with Crown amplifiers.
This calculation will give you the amount of attenuation, in decibels, you can expect with a change in receiver distance, in a free field (outdoors).
For example if you were standing 20 feet from a loudspeaker, and were to move to 40 feet away from that loudspeaker, you would expect to see a drop in level of 6 dB. Sound that is radiated from a point source drops in level at 6 dB per doubling of distance.
Equation used to calculate the data:
Snew=Sref + (20 * Log (Dref / Dnew)
Where:
Dref = Reference Distance
Dnew = New Distance
Sref = Reference Sound Level
Snew = New Sound Level
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Tannoy Installed For Newly Constructed Opera House In Dallas
The Tannoy speakers provide the audio quality and ability to blend necessary for an opera system.
The newly constructed Winspear Opera House, located in downtown Dallas, Texas, is a reinterpretation of the traditional opera house.
The main performance space seats 2,200 and was engineered specifically for performances of opera and musical theater, with stages equipped for ballet performances as well as other forms of dance.
Jeffrey White of Clair Brothers Audio Systems (Dallas) along with Dan Heins of the Nashville Office of Clair Brothers, provided the engineering, shop drawings and installation of the audio systems for the space.
“As a multi-use facility it required a multi-use sound system,” noted White and Heins. “The space was acoustically tuned and designed for Opera complete with a system designed to provide necessary vocal or orchestral support.”
“Another sound system for live music productions was installed for non-opera use. There is very little overlap between the systems.”
The only components that bridge both systems are the 58 Tannoy CMS 501 ceiling speakers that are used for all under and over fill on four tiers of balcony, along with rear fill of the main deck, mezzanine and upper balcony.
Martin Van Dijk of Engineering Harmonics, based out of Toronto, consulted and designed the system and specifically selected Tannoy 501 ceiling speakers.
“Sonic quality was extremely important, especially for an opera system,” said Dijk. “The Tannoy speakers not only provide exceptional audio quality, but they blended seamlessly with both primary systems.”
Initial spacing of the speakers was determined using acoustic software, with critical listening employed in the upper deck to ensure quality coverage. The fourth balcony is situated close to a sloping ceiling which added to the acoustic challenge. Because the 501s are utilized in both systems, it was important that their placement be optimized for both.
These high power and high sensitivity ceiling monitors, containing a 130mm (5.00”) ICT transducer, are specifically designed for applications requiring the combination of premium sonic quality for music and speech reinforcement and exceptional reliability – making them the ideal choice for the Winspear Opera House.
Tannoy Website
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Electro-Voice Chosen By Pyxis For North Coast Church Installs
Pyxis trusted Electro-Voice manufacturer representative Quantum Sales to help them complete the North Coast Church projects on time.
Six weeks isn’t much time to go from an initial customer inquiry to having PA systems on-site and ready to rig.
However, when Pyxis Industries in Riverside, California was asked by North Coast Church to bid on the design and installation of sound systems at two different locations, owner Chad Costanzo didn’t hesitate.
“These installations had to come together extremely quickly,” Costanzo says.
“But our past experience with Electro-Voice gave us total confidence that everything would work out fine. And it did.”
The vendor that had originally been retained to design and install sound reinforcement systems for the sites had to pull out of the project only two months before opening day at both facilities.
Pyxis got the call from North Coast, and Costanzo went to work with systems engineer Alan DiCato, deciding how to handle the project within the church’s allocated budget.
While the previous provider had already specified designs based on products from a different supplier, Pyxis proposed using an Electro-Voice system instead. “EV has become our go-to manufacturer, because the installs we’ve done with their products have always sounded great right out of the box,” said Costanzo.
“And we also knew that EV and their local rep, Quantum Sales, would be able to assist us in making these tight timelines and staying within budget. Both factors were crucial, because, with that short of a time-frame, any issues we had would be a big problem for everyone.”
Pyxis had the two systems designed, priced, and approved within three weeks. “EV has such a broad line of different boxes that we were able to choose something that fit the application and the price point really well,” said DiCato.
The core elements of both systems are “exploded array” clusters drawn from Electro-Voice’s Xi-series. While the rooms are physically dissimilar — one is rectangular and the other more trapezoidal, and their ceilings are different heights — the clusters are identical except for the angles of the speakers.
“Line arrays would not have been a good choice for these rooms,” said DiCato says. “We didn’t have the ceiling height we would have needed, and in one of the rooms we had an odd shape to cover.”
“The exploded array design allowed us to angle the boxes in each room to get the coverage we needed, and also to use fewer boxes and fewer amps channels to get that coverage, which allowed us to stay within the customer’s budget.”
The clusters combine full-range Xi-1153A/64F three-way, 15-inch, medium-throw loudspeakers and Xi-122MHA/64F ultra-compact, high-output, two-way loudspeakers. “We use the 1153s for longer-throw coverage of the main seating area,” DiCato says, “and the 1122s for front fills and down fills.”
“The Xi boxes sound great without a lot of additional processing,” said Costanzo, “and they are very versatile, so they really cover the multiple uses that are planned for these rooms, from a Sunday church service, to a Friday night concert, to conferences and youth groups during the week.”
“And their waveform shaping gives them really good pattern control all the way down below 500 Hz. That allowed us to aim well and control our cutoffs precisely.”
“We have three down-fills hanging below the main cabinets, directly above the stage, so we needed very good control to get good gain-before-feedback. And we had no issues with that at all.”
The low end for both systems is augmented by three dual-18 Xsubs in concrete bunkers below the stage, while stage monitoring is handled by four TX1122 FM stage wedges at each venue.
“They are a great-sounding monitor at a great price point,” said Costanzo.
Power is provided by Electro-Voice CPS series amplifiers. “The amps have headroom galore,” DiCato says, “so nothing needs to work hard at all.”
Costanzo adds that the CPS amps are “known for their reliability, which is great for the HOW market. They will have the amps for a long time without any failures. They sound great – you really can’t go wrong with them.”
For system control, Pyxis chose an Electro-Voice NetMax N8000-1500 with three added DSP cards. “NetMax is an incredible tool,” DiCato said.
“We’ve looked at other systems, and nothing comes close in terms of flexibility and ease of programming. It was easy to get everything set up because it has a very intuitive user interface.”
Click to enlarge.
“All the cabinet models are right there in the library, so you can you just drag them in from the menu.”
That ease of setup turned out to be critical because Pyxis didn’t gain access to the live venue until the Wednesday before the Sunday opening. “They were still painting the booth and hooking up power,” said Costanzo.
“But it all went together how it was supposed to, and worked the first time. We only made very minor adjustments, and I was able to do that all in real-time, running NetMax and Smaart together on my laptop out in the venue. I could make changes in NetMax and see it respond immediately in Smaart. It was great.”
The NetMax systems also included optional digital I/O cards that allow Pyxis to bring in 96 kHz/24-bit AES EBU digital audio directly from the digital console without any conversion. “We can even pull the NetMax interface up on a Midas console display,” said DiCato.
“So the EV gear works seamlessly with the other products we use, and we had no issue at all with interfacing or compatibility. It was just great to have everything work so well right out of the box.”
Electro-Voice Website
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Tips On How To Provide Flawless Wedding Audio
Weddings, more than other events, seem to be prone to audio issues. This unique experience must be flawless because you don't get a second chance.
Wedding horror stories abound.
They remain in the memory of the bride and groom (and their families…) and those memories often include anger, frustration, or anxiety.
When something goes wrong in a regular church service, you fix the problem and move on.
People might remember, but it’s usually no big deal.
The difference is that a wedding is a one time event.
I have been asked to run sound for weddings in two different ways. The first is when a friend says “I’m getting married, can you run sound at the wedding?”
The second is when I’m asked by someone in the wedding party or via the church secretary if I am available on a certain date to run sound for a wedding. No matter what the case, I know it’s time to get to work.
Running sound for a wedding can be much different from a church service. A videographer might want to tie into your mixer so he has a better audio track for his video.
Or, you might have to mic an instrumental quartet. Not only might you do things you normally don’t, you are dealing with new people, not to mention a new order of events that will keep you on your toes.
Where the Work Begins…
The most important detail you need, as soon as possible, is the name and phone number for the person in charge of the wedding service. This might be a hired wedding coordinator or the bride’s mother.
What’s important is that you know who they are, how to contact them, and that you are kept in the loop as to what is needed in the wedding and what is expected from you.
Meet with them well before the wedding rehearsal so you have all the information you need. Most importantly, make sure you are available if they have questions.
Once you know who is in charge of the wedding, meet with them to line up the event order and the requirements. This helps you get all your equipment in order, rent any equipment if necessary, and get copies of all pre-recorded audio.
If they have hired a videographer, contact that person and find out what requirements they might have that impacts your work. Ideally, after meeting with all of these people, you will know what you need, and have an order of events so you can plan your work.
An event schedule (order of events) is important because, just like your regular church service, you need to have everything cued up at the right time. I did a wedding once at a far away church I’d never seen.
The CD player was a portable boom box that was patched into the mixer.
While I made it work, cuing up accompaniment songs during the wedding was a pain. Bottom line, when you know what’s coming, you know how to prepare.
Before I forget, you might be handed a few iterations of the event schedule until a final version is secured the night of the rehearsal.
It happens. Each iteration is usually a slight modification of the previous.
The night of the rehearsal, you must be present. The rehearsal time not only is good for the people in the wedding but it’s a great time for you.
You are about to run sound for a one-of-a-kind wedding. A rehearsal gives you a chance to practice.
If you are handed any last minute audio such as a tape or CD, you should play it through completely so you know it works. You can also set your channel levels, do your eq’ing, and deal with the biggest problem I hear in weddings…“I do.”
The phrase “I do” might be the only phrase the bride and groom ever say but it’s the one that is said the most sincerely, the most heart-felt, and the darn quietest!
You have to decide how to mic the bride and groom. There are several options. First, wireless microphones can be a simple solution.
Just make sure you have them turned off when you hear the phrase “you may kiss the bride.” I’m just not a fan of amplified smooching. (of course the two wirelesss mics that close together would cause problems but where is the humor in that?)
Second, place a corded microphone between them. I don’t like this idea because they have to stand in the right place, it stands out on stage, and it can get in the way.
The last option is a handheld wireless microphone. The best man or the pastor can hand it to the couple or hold it in front of them as they speak. If individual wireless isn’t an option, use the last method.
The rehearsal time also gives you a chance to talk with the wedding coordinator if you have questions. Also, it gives you a chance to give a short instruction session on “how to hold a microphone” if required.
Before the Wedding
The day of the wedding, run through a sound check with your microphones and input devices such as a CD player.
Make sure the lapel microphones have been clipped on the people at the right location on their shirt; a fist below their lowered chin.
Also do a video-audio test with the videographer. When the wedding coordinator shows up, meet with them in case anything has been changed.
Show them your copy of the event schedule and have them verify it’s correct.
You don’t want to be running off an older version.
After the Wedding
After the wedding, I wait until all the visitors have left the sanctuary before I start putting up any stage equipment.
I also take down typical onstage equipment such as microphone stands and music stands because that’s where the wedding party will have photographs taken.
Once everything is put away, I hand the wedding coordinator an audio copy of the wedding.
If they have a videographer, I hand it to them instead.
Even if the videographer had an audio patch into the system, a separate audio copy might be helpful. They have their own horror stories.
Finally, I check in with the wedding coordinator. This usually is a simple “I’m done and I’ve put away all the equipment. Do you need anything else?” I’ve never been needed further but it’s best to ask.
But Wait, There’s More
Weddings are ripe for problems of any nature. I don’t know why, they’re just like that. So, you need to plan for emergencies during the wedding. These can include:
• A spare wireless microphone behind the pulpit
• A spare XLR cable hidden on the stage
• A baseball bat to stop any ringing cell phones
• A plan if the power goes out
You know, just the usual…
The last one is very important. The wedding coordinator helps with the wedding. You help with the audio. If something happens with the electricity, who is in charge?
We’ve all been to a beautiful summer wedding but storms happen and tornadoes can pop up out of nowhere. Establish who will be in charge.
If the power goes out, does the service go on from the light of the exit lamps? If so, do you have a backup system that powers the sound system?
Usually the pastor for the wedding is the pastor of the church, so they take charge when emergencies occur. However, that’s not always the case, so by knowing who is in charge, emergencies can be managed properly.
In the End
Remember a wedding service is a unique experience between two people. You want the service to be flawless.
You don’t get a second chance.
What Bad Wedding Audio Experiences Have You Witnessed? What Wedding Audio Advice Can You Share? Let us know in the comments below!
Ready to learn and laugh? Chris Huff writes about the world of church audio at Behind The Mixer. He covers everything from audio fundamentals to dealing with musicians. He can even tell you the signs the sound guy is having a mental breakdown.
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Tech Tip Of The Day: Condenser Or Dynamic?
What's the right choice for live sound? Is there a right choice?
Q: Is a handheld condenser better for live vocals than a standard dynamic mic?
Why?
A: There’s not an easy answer to this question, as there are advantages and disadvantages to both types of microphones.
The most noticeable difference will usually be an extended high-frequency response in a condenser microphone when compared to a dynamic, which in and of itself most people would probably consider “better.”
They certainly do have the potential to be more accurate. However, there are other factors that need to be taken into consideration.
With that higher frequency response comes a greater susceptibility to feedback, and condensers may be too sensitive in some live situations.
Condensers typically aren’t as durable as dynamics, although that’s not necessarily the case with many of today’s handheld condensers that are built with the abuse handheld microphones tend to be subjected to in mind.
And of course, just as in the studio, there are certain vocalists that may just sound better with a dynamic microphone than a condenser. The extra bleed some condensers pick up can also be hard to deal with.
On the other hand, there certainly are advantages to condenser microphones in a live situation. Condensers do typically do a better job of reproducing the audible frequencies of the human voice, and many people find that it’s easier to mix a vocal picked up by a condenser than a dynamic.
The improved higher frequency reproduction helps the vocal cut through the mix better, and the engineer may not need to resort to equalization as much. And especially with the number of in-ear monitoring systems out there, vocalists may prefer to hear the detail in their voice that a condenser microphone provides.
As with most of the equipment we deal with, what it really comes down to is that what’s better can depend on a number of factors…the singer, the engineer, the system, the room…and really should be considered on a case-by-case basis.
As always, we welcome input from the PSW community and would love to know your thoughts microphone choice. Feel free to let us know in the comments below!
For more tech tips go to Sweetwater.com
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Shure Introduces Multiple New Beta Drum Microphones To The Line
New additions to Beta line offer exceptional sound quality and improved durability in the field.
Shure Incorporated has unveiled new and improved additions to its Beta microphone line.
The newest models include the Beta 91A boundary microphone, Beta 98A miniature instrument microphone, and Beta 98AMP miniature drum microphone.
With quality construction, low handling noise, and high gain before feedback, Shure Beta microphones redefine sensitivity and control for essential, expert sound reinforcement.
All of the new microphones offer high sound quality, while still delivering the same performance standards that users expect from Shure Beta mics.
“We had three goals in mind when we set out to design these new Beta mics,” said Chad Wiggins, Shure’s Category Director for Wired Products.
“To improve sound quality, to improve reliability and durability in the field, and to offer simple, secure mounting and placement.”
“We’ve accomplished all of those goals with these microphones. Engineers will love the functional advantage you get with these mics versus their predecessors.”
The Beta 91A is a half-cardioid condenser boundary microphone for kick-drum and low frequency applications. A new cartridge design provides a smoother, more natural response.
The Beta 91A’s low-profile design, with integrated preamplifier and XLR connector requires no external hardware to maximize setup efficiency while minimizing stage clutter. It also features a low-mid frequency EQ switch that offers additional tonal flexibility.
Rob Mailman, front-of-house engineer for Santana, has been using the Beta 91A for some of his recent performances and said, “I’m certain I’ll use the Beta 91A until I retire.”
The Beta 98A is a miniature cardioid condenser microphone for instrument sound reinforcement and recording applications. High SPL handling makes the Beta 98A ideal for a variety of acoustic or amplified instruments, including drums, piano, reed, wind, and strings.
The newly-designed cartridge features an extremely uniform cardioid polar pattern and provides a natural musical frequency response. It is available in two variations with either a gooseneck drum mount (Beta 98AD/C) or with a stand mount (Beta 98A/C).
“While trying the new Beta 98A prototypes on the last Pretenders U.S. tour,” said front of house engineer Roger Lindsay, “I was delighted to discover that Shure has managed to retain the essential character of the original SM98, which became a benchmark for all drum mics, while also using the latest advances in design and technology to further improve a much-loved model.”
The Beta 98AMP is a new variation of the Beta 98A that combines the new cartridge with a flexible gooseneck and integrated XLR preamplifier.
It ships with the new A75M Universal Microphone Mount for simple, accurate placement in any configuration of toms, snares, or percussion. These new features provide increased control and reduce complexity of setup.
“Because the Beta 98AMP is a new design, we didn’t have a way to clamp it onto a drum,” said Wiggins. “We developed this very small, versatile mount that can be attached or removed in seconds.”
“The A75M has almost infinite adjustability – it’s so universal that we’re also selling it as a standalone accessory. The A75M’s dual-jaw, quick release design offers easy mounting to drums, percussion, drum hardware, and stands.”
Like the other microphones in Shure’s Beta line, such as the new Beta 181 and the Beta 27, these new additions are engineered for superior sound reproduction, low handling noise, and high gain before feedback.
Beta Wired Microphones strengthen all performances, for fine detail in a wide variety of demanding applications and changing environments.
Shure Website
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Solid State Logic AWS 900 Chosen In Pinnacle College Upgrade
New Console brings the Sound and Power of SSL to Course Work.
Pinnacle College recently upgraded Studio A at its Los Angeles campus with a Solid State Logic AWS 900 console.
Formerly known as Sound Master Recording Engineer School, one of the first of its kind in the United States, Pinnacle is carrying on the traditions of knowledge and competence by offering students the very best learning experience.
“At Pinnacle we teach the students how things work, not how to work things,” said Francis Buckley, development manager for Pinnacle College.
“We love the SSL AWS 900+ SE because it is a full-on old school analogue console combined with the modern version of the digital audio workstation control surface.”
“It gives us great flexibility because we can use multiple DAW software packages and the AWS controls them all.”
Pinnacle College offers certificate programs in audio engineering and video game sound design. The coursework puts great emphasis on practical, hands-on training anchored on a comprehensive background of theory of sound and music.
Audio engineering courses include Recording Engineer, Post Production Engineer, Mastering Engineer, Audio Mixer, Sound Designer and Producer. Video game sound design courses include Sound Designer, Music Producer, Dialog Recorder, Foley Artist, Music Composer and Music and Sound Mixer.
Because of the varied missions, the choice of the AWS was critical for students going into the workplace.
“The main reason why Pinnacle chose the AWS is because first and foremost, it’s an SSL,” Buckley said. “It looks exactly like the full-scale SSL consoles our students will be interacting with in music studios, post production facilities, Foley/ADR rooms and even remote trucks.”
“The secondary, but just as important, reason was yeah it’s an SSL – a great sounding board with a great company reputation. Thirdly, the AWS provides us with a vehicle from which we can build the foundational knowledge our students need in every discipline we offer.”
“A dedicated controller for one software package can’t provide the range of learning experience the AWS provides.”
“I have been in the industry for over 30 years and I really appreciate how the AWS works AND feels,” Buckley said.
“The AWS makes my job easier as a teacher because it is a genuine analogue recording console. The AWS lets us teach the basics so our students can operate any console.”
“It is also great from the teacher’s perspective because we know what good audio sounds like and the AWS, as expected from any SSL product, sounds great. Any session we do on this console could be released to the public because the baseline sound is the best in the industry.”
Solid State Logic Website
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QSC Audio Founder Pat Quilter Featured Panelist At AES
Quilter to offer insights on Earth-friendly audio deployment methods and manufacturing challenges faced in today’s global economy.
QSC Audio Products, LLC is pleased to announce that company founder Pat Quilter will be a featured panelist at the upcoming 129th AES Convention held in San Francisco.
Quilter is scheduled to appear at “The Greening of Live Audio for Medium and Small Operators” live sound seminar held on Thursday, November 4 and the “Audio Manufacturing in a Global Economy” product design seminar held on Sunday, November 7.
“It’s an honor to be asked to share my insights, trials and conclusions with those who make up our audio professional community,” said Quilter, Chairman of the Board, QHI Holdings, Inc.
“The world of professional audio has changed tremendously over the past decades. We’re confronting a myriad of technological, logistical, economic and environmental issues today that 30 years ago could not have been fully anticipated.”
Quilter’s first panel discussion will focus on environmentally sustainable audio production practices with consideration given to reducing environmental impact through reduced power draw, decreased transportation, and labor costs.
His second panel discussion confronts the issue of how manufacturing in today’s global economy has affected the audio field from a number of perspectives including: economics, quality, and innovation.
The 129th AES Convention will be held at San Francisco’s Moscone Center from November 4 thru November 7, 2010.
QSC Audio Website
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Tuesday, October 26, 2010
ENTASYS Selected For Elementary School Multi-Purpose Room
The ENTASYS system helped to create a versatile performance space within an average multi-purpose space.
One of the busiest places in Ocean Road Elementary School is the multi-purpose room.
Hosting everything from lunch periods to presentations and performances, the “cafetorium” is a non-stop hub of activity from before classes begin until late into the evening.
It’s an open, acoustically live space that’s designed more for versatility and utility than for sonic performance. Crowds of kids and a loud HVAC system contribute to the room’s typically high ambient noise levels.
School officials were unhappy with the performance of their present sound system, and needed a system that would be effective and versatile enough to perform daily for announcements, after school programs, and a range of music and theatrical events.
“We had installed a Community Veris system Nellie Bennett Elementary, another school in the Point Pleasant Borough School District, and when the folks at Ocean Road heard it they asked us to come demo a system for them,” says Joe DiSabatino of Chews Landing, NJ-based JD Sound and Video.
“We set up an ENTASYS as well, and they were just amazed with the quality They wanted us to install it immediately.”
The system is configured as one ENTASYS full-range column and one ENTASYS low frequency column per side. A pair of VERIS 212 subwoofers are mounted on custom brackets above the ENTASYS.
“The ENTASYS systems deliver great spoken word intelligibility, and even with all the microphones open, the kids can walk right by the columns and get no feedback.” Crown XT-series amps power the system.
“Everyone is extremely impressed with the system,” DiSabatino continues. “The kids can stand on choir risers right in front of the columns and not be blasted out, while the parents all the way in the back can hear their sons and daughters loud and clear.”
“And the VERIS subs add a nice, smooth bottom end, which is great during movie days.”
Community Professional Website
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