Thursday, February 28, 2013
VUE Appoints Media Logic As Exclusive Distributor For Germany, Austria & Switzerland
Berlin-based company distributes premier pro audio brands
VUE Audiotechnik has announced the appointment of Media Logic as the company’s new distribution partner for Austria, Germany and Switzerland.
Established in 2006, Berlin-based Media Logic also distributes premier pro audio brands Fairlight, Eurotaker, Ardis and SCOTTY.
“For nearly two years we have been searching for a top sound reinforcement line to add to our portfolio,” remarks Media Logic CEO Holger de Buhr. “With a team of established audio professionals behind it, VUE has already made a strong impression with a rapidly growing line of well-engineered products.
“Excitement surrounding VUE is extremely high, and we’re looking forward to working with such a dynamic young company.”
The announcement marks the latest milestone in the aggressive expansion of VUE Audiotechnik’s worldwide sales and distribution organization. The company, which launched at the ProLight + Sound Expo in Frankfurt last March, has already introduced more than 25 new products in its first 10 months of operation.
Jim Sides, VUE executive vice president, adds, “Media Logic is the perfect fit for VUE because they share our core belief that strong, long-term customer relationships are essential for a successful business. Media Logic has a tremendous commitment to their customers and an excellent reputation for service and support. I’m delighted they’ve agreed to join the VUE team.”
Fostex Shipping New PMO.3 Studio Monitors
Fostex is now shipping it's new 2-Way Powered Studio Monitors in 3 Color Styles
Japanese pro-audio manufacturer Fostex has announced that the company’s first 2-way powered studio monitor system, the PMO.3, is now shipping.
Offered in three color styles that include gray, white and classic black, the PMO.3 is a professional quality speaker system that combines a dedicated 3-inch LF fiberglass cone woofer and ¾-inch HF silk dome tweeter drivers together with a built-in class D amplifier for unparalleled performance in its class.
Compact in stature, yet efficient and robust in accuracy, the PMO.3 system fits well into any home studio environment. The versatile color options and elements such as a smooth, matte finished front baffle give the system a nice aesthetic profile that suits any taste, and a hard wood enclosure promotes dynamic sound quality.
“The new PM0.3 represents a new era for Fostex studio monitors,” commented Dave Hetrick, the National Sales Manager for Fostex’s US distributor, American Music and Sound. “Utilizing the latest in amplifier and cross-over technology, Fostex has developed not just a compact computer speaker but a professional monitoring tool that occupies the same physical space.”
The PMO.3 system is priced at $129.99 per pair and is available now at authorized US Fostex retailers.
Posted by Julie Clark on 02/28 at 02:43 PM
QSC Releases EASE.GLL Data Files For Entire Loudspeaker Line
New files can work across all three popular EASE platforms
QSC Audio has announced the release of complete EASE .GLL data files for its entire loudspeaker line, including AcousticDesign, Acoustic Performance, K Series, KW Series, KLA Series, WideLine and others.
“Unlike legacy file formats such as .efo, .spk, and .dll, these new .GLL files can work across all three popular platforms; EASE Address, EASE Focus 2, and the full 3D modeling version of EASE 4.3,” states David Fuller, QSC director of technical marketing. “This has been a great effort from our engineering team, not only in the file creation, but also in the hours spent collecting & validating the mechanical and acoustical data.”
QSC director of marketing communications Ray van Straten adds, “By offering consultants and contractors this enhanced ability to integrate our loudspeaker products into their EASE-based design proposals, we are now able to deliver these channel partners a more compelling business solution that fully compliments our products’ sonic performance.”
The QSC EASE library can be found at http://qsc.com/support/resources/ease_downloads.htm#Ease_Data.
Wednesday, February 27, 2013
Meyer Sound Constellation Adds Sonic Dimensions To ROCKY - Das Musical
A daring stage adaptation of Sylvester Stallone’s iconic film, ROCKY - Das Musical opened in Germany to glowing reviews. This scope of dramatic moods is supported sonically through variable acoustics generated by a permanently installed Constellation acoustic system from Meyer Sound.
A daring stage adaptation of Sylvester Stallone’s iconic film, ROCKY - Das Musical opened in Germany to glowing reviews. This scope of dramatic moods is supported sonically through variable acoustics generated by a permanently installed Constellation acoustic system from Meyer Sound.
With an investment of nearly $20 million by Germany’s Stage Entertainment GmbH, Rocky is staged in Hamburg’s 1,400-seat Operettenhaus, a Stage Entertainment-owned theatre with excellent technical amenities but very dry acoustics.
“I started discussing Constellation with technical management at Stage Entertainment early in the project,” says sound designer Peter Hylenski. “As the design for the show came together, it quickly became evident that Constellation would be playing not only a technical role in enhancing room sound, but also an artistic role by providing enhanced ‘environments’ to match specific locales in the story.”
According to Hylenski, the Constellation system realizes a dramatic, two-fold enhancement of his sound design. “As an acoustic enhancement system, even the slightest addition seems to ‘lift the sound off the page’ by giving more depth and excitement to the room,” he says. “But we also use it as an environmental simulator. Two examples are scene where Rocky is punching sides of beef in the meat locker—using short,
hard early reflections—and the final fight scene with long pre-delays to create the feeling of an arena. It’s exciting and theatrical, yet it sounds incredibly real.”
The permanent Constellation system at the Operettenhaus is built around the Meyer Sound D-Mitri digital audio platform with two core processors, a core matrix unit, and nine digital and analog I/O units. Four DVRAS modules generate early reflections and complex late reverberations in four zones. Forty-two cardioid and omni microphones pick up ambient room sound, with the variable acoustics created through 235 Meyer Sound self-powered loudspeakers comprising seven different full-range and subwoofer models. The options for room variation include five reverberation times, each with three density settings, plus nine bass ratio and nine strength presets.
The permanent Constellation system works hand-in-glove with the configuration that Hylenski created specifically for the production. This system has its own D-Mitri-based show control system interfacing directly with Constellation, and the predominantly Meyer Sound loudspeaker system is anchored by twin arrays of 13 per side M’elodie line array loudspeakers with two ground-stacked 700-HP subwoofers under each array. The balance of the system incorporates seven other Meyer Sound full-range loudspeakers and three additional subwoofer models, totaling 124 cabinets in all. Amptown System Company GmbH (ASC) of Hamburg supplied all Meyer Sound components for both the permanent Constellation and current production systems.
The permanent installation of Constellation in the Operettenhaus was made possible in part because of the unique business model of Stage Entertainment, explains Andreas Hammerich, production sound engineer
and sound designer for the company.
“Unlike Broadway and West End, at Stage Entertainment we both produce the shows and own the theatres,” Hammerich notes. “That means we take a long-term look at what technical investments will realize a return. And with Rocky, Constellation is already making a difference. The sound is so special that audiences walking out of the theatre cannot believe what they just experienced. It is that extraordinary.”
ROCKY - Das Musical was produced by Stage Entertainment GmbH, the German subsidiary of Amsterdam-based Stage Entertainment, together with Sylvester Stallone and Ukrainian boxing champions, Vitali &
Wladmir Klitschko. Book was by Thomas Meehan, with music by composer Stephen Flaherty and lyricist Lynn Ahrens.
Adamson Perfect Fit For Carlos Vives Outdoor Concert
Adamson E15 lines arrays were at the heart of the sound reinforcement system utilized during the Fiestas del 20 de Enero festival in Sincelejo, Columbia.
With a career spanning more than three decades, Carlos Vives, is a Latin American entertainment icon. The five time Grammy award winner recently took to the stage in Sincelejo, Colombia for ‘Fiestas del 20 de Enero’.
The outdoor event was held on Friday January 18th, 2013 at Estadio Mochila, and put on by the City Of Sincelejo. 12,000 wildly excited fans filled the grounds to catch a thrilling “Vallenato” style performance by Vives and other Latin American hit makers: KVRASS, Maluma and Alfredo Gutierrez.
C. Vilar Amplificacion Professional provided the complete sound, staging and lighting for the event featuring an Adamson E15 line array system.
Mauricio Vilar, FOH Engineer for Carlos Vives for the past 4 years, wanted to use the highly acclaimed Adamson E15’s for the performance.
“Initially we had a production company proposing another brand of P.A. I had requested the E15’s on the rider and insisted that they were used for the event,” explains Mauricio Vilar, FOH Engineer for Carlos Vives for the past 4 years.
In the sound design process Vilar used the latest version of Adamson Shooter software (2.8.5) to nail down the setup, and glowingly stated the sound was: “Extremely similar to the computer prediction.”
Other tools of the trade Mauricio finds essential, are first and foremost his ears, Smaart 7.4, a wireless measurement system and a live recording with Protools HD.
“To tune the system, this to me is the most accurate way to do it,” he adds.
24 Adamson E15’s were flown above the stage in addition to 8 SpekTrix boxes as front fills. They used 16 stacked T21 subs to fill out the low end, with 8 boxes placed left and right of the stage. Adamson M15’s were used as stage monitors. The boxes were powered and controlled by Lab Gruppen amplifiers and Dolby Lake Processing.
When selecting a system for a large event like this, Cesar Vilar, owner of C.Vilar who also took on double duties as systems tech for the event.
“I look for even coverage, great polar response, and it has to be easy to set up so it doesn’t take hours and hours,” Cesar Vilar explains.
With time being a key factor, both Vilars were pleased when the entire system was up in the air in less than 30 minutes.
“The E15’s were chosen this time because it is easiest for me to get the best and most accurate mix,” Mauricio Vilar says. “Most sound systems I listen to, sound decent for the first 30 meters, after that sound simply gets fussy, not to mention if you walk horizontally.”
“The show is loud and the music is very dynamic. It requires a powerful and coherent sound system in order to generate the feelings that the artist wants the audience feel,” Mauricio Vilar explains. “If the band is playing a slow ballad we set it back, if it’s a Latin hit, I will demand more energy from the system.
“I always try to know in advance what kind of audience we’re catering to.”
Vilar was extremely happy with the final result: “The E15’s gave us enough headroom to get a great mix and I was glad to have such a powerful system.”
In his outboard/effects rack Mauricio uses some tube processors for the main vocal, and a high end multi-channel fx. For this Carlos Vives performance he used a Digidesign Venue Profile, but he’s got a short list of consoles he likes: “I think there are actually 3 very good consoles for FOH mix; Digico, Venue and Midas. Yamaha’s very good for monitors.”
He also did a live recording of the show: “I am half a live sound engineer, and half studio engineer. I spend a lot of time in the studio listening, in detail, to every instrument and every song. This is how I realize what microphone techniques to use, how to improve the show.”
Mauricio Vilar currently also mixes the top music TV shows in Colombia.
Church Sound: Getting Guitar Amps On Stage Under Control
Working in tandem with the musician and pursuing the right techniques
Guitar amps, on the stage, can be a burden for the sound tech. They can be too loud and even blast out the people who are in front of the guitar’s amplifier. You can take control of these stage amps and still meet the needs of the musician.
Getting a handle on stage amps isn’t too hard as long as you keep a few things in mind:
1) Musicians want a specific tone coming from their instruments.
2) Musicians use their own amps for self-monitoring.
3) You can gain enough control over stage amps that get the musicians their tone, meets their self-monitoring needs, and gives you the right amount of volume control.
Respect The Tone
The tone of the instrument is picked to meet the needs of the song or the preferred sound. An electric guitar patched through the sound system won’t have the same sound as if it was run through their amp.
Guitar amplifiers have tone controls and even the make and model of amp has its own tonal characteristics. This isn’t to say guitarists should always use amps on the stage. They can get a lot of tonal controls through pedals and pedal-boards. My point is you need to respect their decision on the importance of tone.
Guitarists who know what they’re doing will have some sort of amp stand or other means of pointing their amp up at their head. This is what you want. If they don’t, you’ve got a bit of work to do because, in those cases, the amp is usually pointed at their knees.
Last I checked, my knees didn’t have the ability to hear sound. Pointing the amp at their head also means less volume is required.
Talk with the musician about angling the amp towards their head. If they question this idea, ask them to play while you point the amplifier at their head. They should be on your side in no time.
The proper amplifier angle can be achieved using a commercial amplifier stand or, if you need to get a little creative, one can find free blueprints on the internet for building a cheap custom amp stand. Once the amp stand is in place and pointed properly at the guitarist, have the musician set the amp tone so it’s how they want it.
Note the tone of the amp sounds differently when you are in front of it versus to the side. In the case of musicians who have been using the wrong setup, they will need to change their tone settings because now, for the first time, they are on-axis with the amp’s speaker(s).
Using the proper self-monitoring setup as described above, the musician doesn’t need the same level of volume. This means you get volume control. It’s time to grab a microphone.
Follow these instructions for setting up a microphone:
1) Using an instrument microphone, attach it to a small microphone stand and put the head of the microphone about 2 inches from one of the amp’s speaker cones. If you’ve got a Shure SM57, that’s a good one for this type of application.
Note that some amps have more than one speaker, so pick one.
2) Next, set the microphone so it’s pointing at the outer edge of the speaker cone. Listen to how it sounds in the house speakers. Now move the microphone to the near center of the amp speaker and listen in the house speakers. Note the difference between the two sounds. The farther away from the center of the speaker, the more treble you will hear in the amp.
Be aware that the center of the speaker doesn’t produce sound waves, so don’t point the microphone there. Experiment with several locations on the amplifier’s speaker until you find the spot you like.
3) Once the amplifier is set up and the microphone is in place, you need to find a volume level on the amplifier that’s loud enough as a monitor but doesn’t blow away everyone on the stage. This might take some trial and error and a little assertive pushing on your part if the musician is used to using their amp as the main volume source.
Let them know their only concern is the volume for self-monitoring and that you’ll take care of the house volume.
The Take Away
Proper on-stage amp usage is about directionality, proper microphone placement, and on-stage volume control. Respect the musician’s views on amplifier tone and their needs for self-monitoring.
A common question I get via e-mail concerns controlling guitar amps on the stage. You can do it. It just takes a bit of time and the willingness to talk with the guitarist.
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, and can even tell you the signs the sound guy is having a mental breakdown. To view the original article and to make comments, go here.
Tuesday, February 26, 2013
Solving A Conundrum: Deployment Of A New Subwoofer Configuration
Any reservations were quickly forgotten once we heard the rig...
I’ve spent the past several years experimenting with various types of subwoofer arrays and have arrived at the same conclusion each time: with each method, something is gained while something else is lost.
Left-right placement enhances coverage on the sides at the expense of drastic power alleys and cancellation zones.
Horizontal arrays produce even coverage in front of the array at the expense of a major drop-off on the sides.
With delayed arcing, side coverage is improved, at the expense of tightness.
With cardioid “front-back-front” setups, tightness is enhance at the expense of output SPL.
I could go on, but you get the idea.
Sub deployment is especially tricky in arenas, where the large open area and hard (usually concrete) surfaces makes for the addition of low-end “mud” in the diffused field, along with the need to cover a wide area in the horizontal plain – usually 180 degrees or more.
One of the more common practices I see in the field is flown left-right sub arrays, where hangs of usually eight or more boxes are flown either just outside of or behind the main PA hangs. While the throw is fantastic, comb filtering from left to right is a mathematical certainty.
My choice has been a horizontal array, where boxes are placed along the front of the deck and spaced evenly at a quarter-wavelength of the target frequency. The coverage on the floor is extremely solid and even, but it’s virtually non-existent elsewhere.
I wrestled with this problem last year with a system provided for an Easter Sunday worship service presented by Austin Stone Community Church at the Frank Erwin Center on the campus of the University of Texas in Austin.
While the primary function of the Erwin Center is hosting college basketball games, it’s also booked solid most of the year with A-list tours that draw 15,000-17,000 in attendance. The floor is elongated, stretching 110 feet from the deck to front of house, while the upper levels begin to widen, until the arena becomes almost perfectly round at the top of the mezzanine.
The scene on Easter Sunday at the Erwin Center—click to enlarge. (Credit: Scott Wade)
My typical horizontal array was less than ideal for this venue. While the floor sounded (and felt) fantastic, you only had to climb a few rows off to the sides before the low-end began to drop off quickly. While some of this can be corrected by delayed arcing (delaying the outside boxes to create a virtual arc), the added difference in arrival times causes the entire array to lose much of its tightness due to smearing in the time domain.
I left that gig determined to find a solution. There had to be a better way to get even coverage, both in front and on the sides, without power alleys or comb filtering. In an ideal situation, the low-end should come from a single source, equidistant from the main left-right hangs. But where would it be located?
This past Easter Sunday, when the church returned to the arena, I had the unique opportunity to try a different approach. Working with local audio providers Big House Sound, we designed and deployed a large-scale, flown, end-fire subwoofer array. This had yet to be tested, but once the rig was in the air, jaws began to drop.
Part of the idea came from fellow engineer/tech Sarah Butt, who had worked under Kenny Chesney system engineer John Mills for one of the “Goin’ Coastal” tour dates at Cowboys Stadium in Arlington. Mills’ design for a large Electro-Voice X-Line rig incorporated two monstrous hangs of subs (20 per hang) flown dead center of down stage, between the mains. The theory made sense – two hangs acting as a single source, dead center of the rig, with no cancellations or drop-offs anywhere in the venue. This got my brain spinning.
At left, a screen shot of the system modeling; at right, polar data of the “EF6”6 end-fire array, based on a ground stack two cabinets high—click to enlarge. (Credit: Adamson)
There were four primary issues to address in order for my approach to be a success. First, it had to be cost-effective. This is already an expensive gig, and I couldn’t afford to add a lot more boxes. Second, we needed to take advantage of the 90-degree off-axis null directly beneath a vertically flown sub array to reduce the back-pressure on stage. Third, the arrays needed to be short, since longer arrays reduce vertical coverage by making the hang more directional. They needed to behave as a point source.
Finally, we needed it to be cardioid. I’ve found that the boomy low-end in arenas typically results in me pushing the subs harder to get that tight “feel” I’m looking for. On the flip side, I tend to use less low-end in my mixes with cardioid subs, attaining a clean, tight feel at lower levels when I’m not dealing with a lot of late reflections from all of that backfiring energy.
PLAYING IT OUT
I decided on two arrays, each comprised of six Adamson Systems T21 subwoofers, one behind the other in an end-fire formation.
With significantly more output power than most dual-18-in subs on the market, it required less of them to get the desired SPL, and also kept the hangs short.
Four additional T21s were spread across the floor to help fill in some of the gap before the flown subs took over.
The main hangs consisted of 12 Adamson Y18s per side, with four Y10 for down fill and 12 Y10s for out hangs.
The rig was powered by 30 Lab.Gruppen PLM 10000Q amplifiers, with all system DSP done inside the amps.
Determining the exact end-fire spacing to optimize gain, spacing and delay time was a joint effort involving Adamson and it’s distributor in France, DV2. The result was a textbook cardioid pattern from 40 Hz to 80 Hz, where the subs were crossed over. The “EF66” preset, as Adamson called it, spaced the two hangs 66 inches apart around a center frequency of 51 Hz.
To prevent the rig from swinging (and hence changing the precise spacing), a piece of steel pipe was linked between the rear pick point of the downstage hang and the front point of the upstage hang. Cheeseborough clamps allowed us to adjust spacing once the rig was floating.
A look at the centrally located end-fire array, as well as some of the mains—click to enlarge. (Credit: Cody Hester)
Finally, a cable bridge constructed of minibeam spanned from the sub hang downstage center to the stage left cable pick, in order to reduce added tow on the hang.
Any reservations were quickly forgotten once we heard the rig. I was afraid that the flown subs might lose some of their “punch,” but that certainly wasn’t the case here. The low-end was extremely tight, even and clean. There was no more than 7 dB of drop-off from the first row to the top of the mezzanine, and more importantly, the ratio of subs-to-mains remained constant, leaving the mix to sound full and clear, no matter where you were located.
Behind the rig, the cardioid pattern performed beautifully. The sections behind the stage were so quiet it almost felt eerie. In the past four years working with Big House, I’ve heard this rig deployed dozens of times, and became used to the large amounts of back pressure these subs typically produce – but not this time. It was practically silent, except for the late 1.5-second reflections coming back from behind front of house.
When all was said and done, we couldn’t have been more pleased with the outcome. It was hands-down the smoothest, tightest and most even low-end I’ve ever gotten out of an arena system. The comments about the superb low-end coverage haven’t stopped coming in, and one production manager told us that it sounded better than the rig a major concert tour brought in just a few weeks prior. Whether or not that statement is true, I don’t know, but I’ll take the compliment nonetheless!
The Easter Sunday sound crew at the Erwin Center, left to right: Mark May (system tech), Sarah Butt (A2), Todd Hartmann (front of house, designer), Fiona Cheung, (patch), Mateo Rodriguez (monitor engineer), and Cody Hester (A3)—click to enlarge. (Credit: Scott Wade)
I fully believe in giving credit where credit is due, especially in this industry. We’re fortunate to benefit from some pretty great minds, using their ideas to take things to a new level. I have tremendous respect for people like Dave Rat, who consistently put new concepts out there simply because they’re passionate about live audio, and want to make us all better as a whole.
We are, in a sense, standing on the shoulders of giants. I truly look forward to what future discoveries will be made and how they will continue to improve one of the coolest fields a person could find themselves employed in.
Todd Hartmann is the audio engineering coordinator for the Austin Stone Community Church as well as A1 systems engineer for Big House Sound in Austin, TX.
An Examination Of Bandwidth, Dynamic Range And Normal Operating Levels
The nature of peak and average levels of music and speech, methods of dealing with signal peaks, and more
Audio signals are, of course, speech and music, and in this article we will examine the nature of those signals in terms of their requirements in bandwidth, dynamic range and normal operating levels.
The nature of peak and average levels of music and speech will be discussed.
In addition, we’ll look at the standard methods of dealing with signal peaks and required shifts in signal operating levels.
The data of Figure 1 shows the approximate limits of bandwidth and dynamic range of music and speech signals as normally perceived in concert halls and in face-to-face communication.
The outer limit indicates the maximum envelope of audible sound for young listeners with normal hearing.
Music occupies a more limited range, especially at higher frequencies.
And amplified speech occupies a still smaller range.
Figure 1: Normal limits of hearing, music and speech.
If we were to analyze cumulative speech signals using an octave-band analyzer we would find that a normal adult male speech spectrum would look like that shown in Figure 2. The speech power spectrum has a maximum value in the 250-octave band and falls off both above and below that band. In the range above 1 kHz the falloff is approximately 6 dB per octave.
Figure 2: Long-term octave-wide power spectrum for male speech.
The long-term octave-wide power spectra of classical and rock music are shown in Figure 3. Note that the spectrum of classical music is similar to that of speech at middle and higher frequencies.
Figure 3: Long-term octave-wide power spectra for classical and rock music.
OCTAVE BANDWIDTH, SPEECH INTELLIGIBILITY
Quite separate from the normal power spectrum of speech is the octave-band contribution to speech intelligibility, as shown in Figure 4. Speech does not have to sound natural in order to be intelligible, as we all know from using the telephone, where bandwidth is limited more or less to 300 Hz to 3 kHz.
Figure 4: Octave band contributions to speech intelligibility.
As we can see in the figure, the two octaves between 1 kHz and 4 kHz are dominant, and this is why, in very noisy listening environments, sound reinforcement systems are often band-limited to this range. Ideally, we would like for reproduced or reinforced speech to sound both natural and intelligible, and this is certainly possible in reasonably quiet environments.
INTELLIGIBILTY, AMBIENT NOISE LEVEL
Ideally, the local noise floor should be about 25 dB below average speech levels for the most natural reinforcement of speech.
If the ambient noise level in a space is only 15 dB below the speech level, most listeners will have no trouble understanding the message, but many of them will complain about the noise level.
As the speech-to-noise ratio is further reduced there will be a pronounced loss in intelligibility for all listeners, prompting sound system operators to increase the level of the reinforced speech signal. There is a limit to this procedure however.
When is speech level too loud? Normal face-to-face speech communication is in the range of 60 to 65 dB SPL; however, most speech reinforcement systems operate in the range of 70 to 75 dB SPL.
If the level of amplified speech is increased beyond the range of about 85 or 90 dB SPL, there will be little increase in overall intelligibility, and most listeners will complain of excessive levels.
At even higher levels there will be a diminishing of intelligibility as most listeners will literally feel oppressed by the too-high levels. The trend here is shown in Figure 5.
Figure 5: Effect of speech level on intelligibility.
There is an optimum operating range for a speech reinforcement system. For those systems in very quiet surroundings a normal level of 65 to 75 dB SPL is ideal. In progressively noisier environments the system operating level should be raised so that the signal-to-noise ratio is at least 15 dB.
Typical here would be a transportation terminal at peak travel times, where noise levels in the 60 to 65 dB(A) range would call for system operation at peak levels of 80 dB SPL for greatest intelligibility.
Sports venues often present high crowd noise levels in the range of 85 to 95 dB SPL, and under these conditions it is virtually impossible for a speech reinforcement system to work at all. It is better to wait until crowd noise subsides before making announcements.
MATCHING SPEECH LEVELS
We have seen that amplified speech levels must be contained within a fairly narrow range of about 15 or 20 dB for most effective operation, and systems should be designed with this requirement in mind.
First, we will show the waveforems for sine and square waves of an amplifier capable of delivering 100 watts into an eight-ohm load. Note that full utilization of the amplifier’s voltage drive limits, the sine wave output is 100 watts, while the output of a square wave will be 200 watts.
Why then do we rate this amplifier at only 100 watts? All amplifiers are rated according to their maximum sine wave output capability into a stated load impedance. The sine wave has a 3-dB crest factor (peak-to-RMS ratio), while the square wave has a crest factor equal to unity, as shown in Figure 6.
Figure 6: Examples of sine and square waves at the output of a 100-watt amplifier.
Since music and speech signals are composed primarily of sine-like waves, the amplifier’s power nominal rating is stated as 0.707, the actual peak output voltage rating of the amplifier, or 3 dB lower.
If we actually record a typical speech signal over a period of about 20 seconds, the signal envelope will look much like that shown in Figure 7. You can see that average signal hovers largely around the baseline, with occasional higher values and only rarely reaching the full scale of the figure.
Figure 7: Speech envelope over a 20-second interval.
Now, let’s feed this speech signal to an amplifier with an output capability of 100 watts into an 8 ohm load, as shown in Figure 8. We have labeled the left axis with the actual output voltage produced by the amplifier, and we have indicated the approximate average signal voltage at the right axis.
Figure 8: Speech envelope at the output of a nominal 100-watt amplifier.
It is clear in this figure is that the average signal output is about ±10 volts, while the full voltage output capability of the amplifier is ±40 volts. The difference here is 12 dB, which corresponds to a power difference of 16 to 1.
Stated differently, in order to provide peak output capability of 100 watts for speech signals, the amplifier in question can only deliver an average output of 6.3 watts for normal speech signals.
In order to handle the occasional speech peaks, the amplifier is operating at an average power output of 6.3 watts.
This may not be enough power output for effective system operation, and we can solve the problem two ways:
1) Use a larger output power amplifier. For example, a 200-watt amplifier would provide a new average operating level of about 12.5 watts (-12 dB relative to 200 watts).
While this might get the job done, it is still an inefficient mode of operation.
2) Peak-limit the input signal so that the normal peak-to-average signal ratio is less than 12 dB.
If we do this, a higher average output from the amplifier can be attained.
SIGNAL PEAK LIMITING, CONDITIONING
Figure 9 shows the result of limiting the input signal by about 3.5 dB, while retaining the 100-watt amplifier. When this is done, the new peak signals may now be raised so that they correspond to full output of the amplifier.
Figure 9: New speech envelope with 3.5-dB of signal compression.
Values of + 15 volts now correspond to normal signal levels, resulting in a new average power output of 14 watts for normal program.
We can extend the process a little further by adding another 2.5 dB of limiting for a maximum of 6 dB signal limiting overall, as shown in Figure 10. Here, we have raised the power available for normal signal levels to 25 watts.
Figure 10: New speech envelope with 6 dB of signal compression.
It you study Figures 8, 9 and 10 you will notice that, at each step, the amount of useful “signal space” has effectively doubled. The dark area under the curve is roughly proportional to signal power, and thus relates to perceived loudness.
At the same time, peak levels have remained the same, and this invariably raises the questions: Is the signal limiting we are applying deleterious to the signal? Can you hear it in operation? The answer is mixed; an experienced listener may be able to identify the signal limiting as such, but it will not sound unnatural if it is properly done. The limited signal is louder and as such permits an improvement in intelligibility.
In normal speech applications 12 dB would be about the maximum amount of signal limiting that would be employed. However, for music applications it is customary to provide for a higher degree of signal limiting, plus some degree of compression. Compression and limiting are related operations, and a combination of both enables level manipulations to be made over a fairly wide dynamic range.
An example of the need for both limiting and compression would be a speech reinforcement system in a house of worship where both clergy and lay persons may be called upon to talk. Both experienced and inexperienced talkers will present a wide range of levels at the microphone which can be safely processed by a limiter and compressor in tandem.
METERING IN TRANSMISSION SYSTEMS
Today there are basically two kinds of metering, average and peak. The common VU meter is an example of an averaging meter and as such has nominal rise time and fall-back times of about 0.3 second.
The meter’s rise time is the time taken for a steady-state input signal to the meter to reach 63 percent of its final deflection; the fall-back time is the time taken for the steady-state signal to return from full deflection to 37 percent deflection. Rise and fall-back times are known collectively as the ballistics of the meter.
The original VU meters were passive devices and as such, had ballistic characteristics of a spring-loaded coil with inertia immersed in a magnetic field. Since it is basically an average-reading device, the VU meter has met with continuing success in broadcast work, inasmuch as its readings correspond to the perceived loudness of speech signals.
From their inception, peak program meters (PPM) have been electronic devices an as such can be made to respond very quickly. Typically, a PPM has a rise time of about 10 milliseconds and a fall-back time of about 4 seconds. The rapid rise time permits accurate reading of signals of very short duration, while the slow fall-back time gives the operating engineer adequate time to observe the signal’s value.
Figure 11 shows views of the VU meter (A) and the PPM meter (B). Rise time ballistics of the two types of meters are shown at C.
Figure 11: The VU meter (A); the peak program meter (PPM) (B); rise-time ballistics for VU meter and PPM (C).
Relative calibration points on the meter faces for four kinds of meters are shown in Figure 12. If both VU and PPM meters are calibrated as shown in Figure 15-12, normal speech program will read maximum values of about +2 or +3 VU, while on the PPM the corresponding readings would be between markers 4 and 6 on the face of the meter, due to the more rapid rise time of the PPM relative to the VU meter.
Figure 12: Comparison of European PPM and American VU meter calibration standards.
As we have seen, normal speech has a peak factor of about 12 dB. Music on the other hand can have peak factors that are in the range of 16 to 20 dB, depending on the nature of the material.
Highly compressed music signals, such as are common in modern pop and rock music may have peak factors no greater than about 4 dB; however, classical music may present numerous operating levels, each requiring recalibration as the program progresses.
Many times during outdoor classical music events at summer festivals the sound reinforcement system is carefully adjusted manually, usually by an operating engineer working with an assistant producer with score in hand.
Figure 13 shows a typical example of how this is done. The engineer must be aware of how loud the orchestra will play and how these loudness peaks will translate through the music reinforcement system.
Figure 13: Example of shifting of operating levels in a musical program. A long-term classical music program progressing from an extended slow, soft section to a louder section (A); having raised the gain for the softer section, the mixing engineer must slowly reduce the gain by 8 dB as indicated (B).
The aim is to contain the peaks within an agreed upon level at selected positions in the large audience area. Such levels as these are often established so as not to produce any disturbance at monitoring points in nearby residential areas.
At the same time, both engineer and producer know that low-level music passages may get lost in the ever-present noise level of large audiences, traffic, overflights and the like. Operating level shifts of the order of 12 dB are very common, and when smoothly executed may be barely noticeable as such.
RECOMMENDED GAIN STRUCTURE
System headroom and operating levels are normally defined at the line output stage of the operating console, while system noise floor is defined at the microphone input stage.
The total dynamic range of the system is thus established and cannot be improved upon later in the audio chain.
However, through careless down-stream gain structure it can be degraded.
As an absolutely safe procedure we recommend that a music or speech reinforcement system be setup to provide a nominal 20 dB of operating headroom over the normal “zero level” calibration.
This should apply across the board, so to speak, to all electronic elements in the chain.
Basically, once the headroom value in dB has been determined, the precise relationship between headroom and operating level should be maintained through all following line level electronics.
At the end of the chain the power amplifier-loudspeaker combination must be considered as a separate entity, and adjustments made so that a given signal level (e. g., 0 dBu) is assigned a given sound pressure level in the house.
This process is shown in Figure14 for a relatively simple reinforcement system.
Figure 14: Setting gain structure in a speech reinforcement system.
Our recommendation is that a VU meter reading of “zero” at the output of the operating console be assigned a nominal level mid-way in the seating space of about 72 dB SPL. You may wish to change this value slightly, depending on local requirements.
This standard approach simplifies normal system operation; all the operator has to do is raise or lower the input fader of the console to attain a nominal zero dB reading in order to ensure consistent speech levels in the listening space.
This article is excerpted from JBL Audio Engineering For Sound Reinforcement, used by permission of JBL Professional.
XLR Joins L-Acoustics Certified Provider Network In Belguim
L-Acoustics has named the Brussels-based sales and distribution team XLR as a new L-Acoustics Certified Provider network for Belgium as a distributor.
L-Acoustics is pleased to announce that Brussels-based sales and distribution team XLR has joined the new L-Acoustics Certified Provider network for Belgium as a distributor.
Jan de Brucker and Louis Lukusa head the team dealing with Dutch-speaking Belgium and with the French-speaking side of Belgium, respectively, backed up by the technical expertise of Sébastien Desaever.
As a wholesaler, XLR offers exclusive pro audio equipment and service to a well-established network of partners in Belgium and Luxembourg in the rental and installation market.
Jan de Brucker and Louis Lukusa said, “L-Acoustics is the leading brand in our market and we are proud to be part of the Certified Provider network. The high level of quality offered by L-Acoustics fits perfectly with the needs of our customers. We look forward to working with L-Acoustics going forward.”
Tim McCall, Regional Sales Manager at L-Acoustics said, “We are very glad to work with Jan, Louis and Seb, who have extensive experience of the audio industry, to help promote L-Acoustics in the Belgian market.
“They are a valuable addition to our existing network of Certified Providers. The enthusiasm and knowledge of the XLR team alongside the recent additions to the L-Acoustics range of ARCS WIDE and FOCUS, 5XT and SB15m should be a winning recipe for our customers in the region.”
The Who Reign O’er Quadrophenia With DiGiCo
The flexibility of the DiGiCo SD7 console’s Snapshot programming enabled engineers to handle the show’s prerecorded music and extensive sound-effect cues
The Who’s 1973 rock opera Quadrophenia—which sets the tale of teen Jimmy Cooper amidst the global sociocultural upheaval and psychological angst of the times and the rivalry between Britain’s mods and rockers—has been reprised in a multimedia display on the band’s latest outing.
The 37 date tour, which began in November and runs through the end of February, celebrates the four-decade anniversary of the album’s release and marks the band’s first major North American tour in four years.
The Quadrophenia tour reunites the band with production partners Eighth Day Sound, who have worked with the iconic rockers on their last three major tours.
This time out they’re carrying a pair of DiGiCo SD7 desks (each running the latest MACH III software) for FOH and band monitors, plus an SD-Rack at FOH and a d&b audiotechnik J-Series PA. The audio crew is comprised of longtime Who FOH engineer Robert Collins, Simon Higgs on monitors with support from Eighth Day’s Senior Audio Engineer Mark Brnich, and sound techs Drew Marbar and Carl Popek.
The tour also reunites long-departed drummer Keith Moon and bassist John Entwistle, who make cameo appearances, joining remaining original members Pete Townshend and Roger Daltrey. Entwistle’s virtuosity and famous bass solo on “5:15” are showcased in live footage shot at the Royal Albert Hall in 2000, which streams onscreen.
They also pay tribute to the late Keith Moon; their performance of “Bell Boy” incorporates video footage of a 1974 performance, with Moon’s vocals dubbed in from the LP (one of the only times in Who history his vocals were heard on an album).
Monday, February 25, 2013
JBL Professional Loudspeakers Fill Out the Sound System at Penn State’s Beaver Stadium
Systems contractor Clair Brothers recently upgraded Beaver Stadium’s sound system with the installation of more than 140 Harman’s JBL loudspeakers to improve its performance in a number of coverage areas.
Beaver Stadium, home to Penn State University’s Nittany Lions football team, is one of the largest in the country with a capacity of almost 107,000. In fact, it is the second-largest stadium in the Western Hemisphere.
The stadium has expanded six times in its history and recently, systems contractor Clair Brothers upgraded Beaver Stadium’s sound system with the installation of more than 140 Harman JBL loudspeakers to improve its performance in a number of coverage areas.
“We had already replaced the main PA system in 2007 with a comparatively small JBL VLA601 and VLA301 line array system, but work still needed to be done to address areas in the North and South end zones of the stadium where the sound from the main PA is blocked or where we determined that additional coverage was needed,” said Jim Devenney, Sr. Systems Designer and Project Manager at Clair Brothers.
“The balconies are large and present significant acoustical barriers. There were already speakers in place in these areas but they had been there for a while and needed to be upgraded,” Devenney continues. “In the South end zone there are three separate areas, the main PA does not cover since the PA shoots over the top of the entire area.
“The tight down angle from the top of the scoreboard makes it impossible for the main PA to adequately reach those areas.”
“The lower level is used mostly for student seating, we mounted 16 JBL AW295 all-weather speakers on the face of the balcony to cover these seats between the balcony and the field.
“In addition there are 36 AWC82 outdoor speakers positioned in two alternating rows of 18, with 18 pointed straight down and another 18 placed in between them and aimed back away from the field to cover the area under the balcony.”
“The Club Level is the next level up and here the balcony directly blocks the sound from the main PA. To remedy this, 14 AW295 speakers face toward the field and 52 AWC82 speakers are placed facing down. There’s a seating area under the scoreboard that also needed attention. We added 24 JBL Control 25AV indoor/outdoor speakers to deliver sound to the people sitting under the scoreboard.”
“The top of the lower level seating in the North end zone is also shaded from the main system by an overhanging balcony. We installed 10 AWC129 speakers here.”
When it comes to installation challenges Clair Brothers has seen it all, yet the Beaver Stadium upgrade threw the proverbial fly in the ointment for Devenney and crew.
“Putting up the speakers was the easy part,” Devenney said. “Our installers loved the speakers as they were well-designed and easy to mount. However, we had to install them during the Penn State football season so we had to put up and take down the scaffolding after every game.
“The schedule was also so tight that we had the speakers flown from the plant as soon as they were manufactured! We still were able to complete the installation in four weeks. The stadium’s operations staff was very helpful with logistics, ensuring the system installation would stay on schedule.”
Friday, February 22, 2013
Capital Sound Hire And Wigwam First In UK With NEXO STM Series
Both companies have been involved for two years in the development of STM
London-based Capital Sound Hire has become the first UK company to take delivery of a NEXO STM Series modular line array system, and Wigwam Acoustics has also confirmed its purchase of a STM system.
Both companies have been involved for two years in the development of STM.
The initial STM touring range incorporates three elements; the M46 Main cabinet, B112 Bass cabinet and S118 Sub-bass cabinet, which allow users to build systems for audiences from 200 to 100,000.
“With profit margins being squeezed, it’s crucial for the success of rental companies to get the maximum flexibility from their inventories,” says Paul Timmins of Capital Sound. “Prep costs money, and this system only requires configuration between projects, with no rebuilds necessary.”
The scalability of STM held great appeal for Capital, whose projects range from 10 people at small corporate events to stadium and outdoor shows for 60,000 plus.
“What has become evident is that most current loudspeaker systems require you to stock up to four sizes of line array boxes to cover the variation in room sizes,” Timmins continues. “Often, during the winter months, an entire inventory of large format loudspeakers can sit idle while there is a shortage of small and mid-sized cabinets.
“The concept of STM is amazing, with true multi-use options for all components including the amplifier racks. System designs can take into consideration not only room limitations but also musical content. It’s a fantastic addition to the tool box.”
Capital, which will ultimately own a 48-box system an inventory of 48x M46 Mains, 48x B112 Bass and 48x S118 Subs, is initially planning to introduce the components on small-scale events. This will ramp up during summer 2013 to larger projects that will allow engineers to get to grips with its full capabilities.
“NEXO has been a great company to work with,” concludes Timmins of the STM project. “They have a fantastic R&D team and they also know where all the best restaurants are!”
In STM terminology, Wigwam’s package is 48 sets: 48 x M46 main cabinets, 48 x B112 bass cabinets and 48 x S118 sub bass units. The system specification is identical to that ordered by Capital Sound.
The main criteria for the new system, according to Wigwam’s Chris Hill, was, “will it sound good, will it be flexible, and quick to rig? NEXO engineers really listened; STM is scalable, and we can use it for many different applications instead of it just being able to do certain types of work. We’re hopeful that this will give us maximum utilization of our stock.”
Capital Sound Hire
Turbosound Hits The Beat At Beirut’s Hippest New Club
A selection of wide-dispersion loudspeakers models from Turbosound were installed in a Beirut nightclub.
A selection of wide-dispersion loudspeaker models designed and built by British manufacturer Turbosound have been installed in one of Beirut’s most happening new nightclubs.
“Ora is first and foremost an after-work and night bar, with great music and an amazing vibe,” explains club co-owner Vivian Shallop. “A crucial part of our club’s reputation is secured on the ambience of the environment. The quality of the sound is a huge part of creating that.
“The other element is the design of the surroundings and it’s fabulous that the speaker design is so discreet – they certainly don’t interfere or detract from the smooth lines of our interior design, and we are delighted with the clarity of sound from the system.”
The club, located on the city’s Dbayeh Seaside Highway, has employed a cool, clean futuristic look – including curved white walls, rich colour washes of light, and plenty of windows looking out over the decking to the Beirut skyline beyond. The discreet design of Turbosound’s loudspeakers means that they blend seamlessly with the décor, allowing them to get on with the job of providing great sound.
“The Ora lounge bar invested in eight TCX-8 passive two-way loudspeakers, two TCX-15 passive two-ways, and a TXD-215 twin 15” subwoofer,” says Louay Agha of installer and long-term Turbosound partner, Thunder Electronics. “The Converging Elliptical Waveguides in the wall mounted TCX-8 and TCX-15 contracting series loudspeakers create a smooth wavefront and even dispersion, and provide masterful pattern control to optimize coverage of the room. The modern design and professional audio quality of the TCX loudspeakers are ideal for the changing atmosphere of the club, providing a flexible sound system tailored for the kind of environment that attracts Ora’s patrons.”
“I’ve been a supporter of Turbosound speakers for many years as the sound is simply amazing, really hi-fi, even when the system is turned up to high SPLs,” comments Agha. “We chose to install these models at the Ora bar because they fit into the environment well, but also, of course, because their wide dispersion means that we can achieve great sound in all of the bar’s areas with fewer speaker units.”
The opening night of the club saw an exciting mix of clientele, including Lebanese celebrities such as Joe Ashkar and his wife Grace Ayanian, as well as comic Mario Bassil. It was a busy and popular night, and Ora now looks set to become the Beirut hotspot.
VUE Audiotechnik Announces Distribution For Australia, New Zealand, South Korea & Indonesia
Three new distributors join expanding sales and support network for VUE
VUE Audiotechnik has announced the completion of three distribution agreements that will extend the company’s reach into key territories throughout the Asia-Pacific region.
Amber Technology is now VUE’s exclusive distribution partner in Australia and New Zealand, Chungbo Sound signs on as VUE’s distributor in South Korea, and PT Sarana Generasi Bintang assumes distribution duties in Indonesia.
“Since launching early last year there has been tremendous interest in VUE throughout the Asia-Pacific,” explains Jim Sides, VUE’s executive vice president. “In fact, VUE products have already been specified into several high-profile installations there.
“These teams possess the knowledge, resources and enthusiasm to build on this momentum, and I look forward to working with each group to best serve VUE customers in these important countries.”
All three distributors assume their responsibilities immediately, and in doing so, join a rapidly expanding sales and support network for VUE Audiotechnik.
For a complete list of VUE Audiotechnik’s reps and distributors around the world go to www.vueaudio.com.
Thursday, February 21, 2013
Kitchen & Bar Van Rijn Open With RCF
Inspired by the beauty and the masterpieces of Rembrandt van Rijn, guests can enjoy authentic Dutch cuisine in a cosmopolitan atmosphere with open kitchen and sound reinforcement from RCF.
The new Kitchen & Bar Van Rijn adjoins the famous Escape Club on Amsterdam’s fashionable Rembrandtplein, and is owned by entrepreneurs Ton and Dick Poppes. Recently opened, the restaurant is laid out similar to a grand café, with a terrace and upstairs area.
Inspired by the beauty and the masterpieces of Rembrandt van Rijn, guests can enjoy authentic Dutch cuisine in a cosmopolitan atmosphere with open kitchen and sound reinforcement from RCF.
The Poppes tapped 24/7 Audio, a locally based service company, to design and install the new system.
24/7 Audio owner Mike Ho specified RCF TTL11-A active steerable column arrays because of the high audio quality the system provides. When completed the entire venue only required two systems with two presets to service all the sound source requirements — including live music and a DJ on weekends.
“I heard a demo of this system at RCF’s headquarters in Italy, and was so blown away that I couldn’t wait to use it,” explains Ho. “Since the requirement at Kitchen & Bar was multifunctional, this presented the ideal opportunity.”
“The owners have been doing this for more than 30 years and one was a DJ back in the day — so we both know the importance of good audio,” Ho adds.
The system is composed of two modules, TTL11A-H for the mid-high and TTL11A-B for the bass frequencies. It is run with a separate a separate sub bass extension and is digitally steerable down to -10 degrees.
Since the system is used for live artists, DJ, various live playback devices and a pianist at a grand piano, Mike Ho has judiciously processed the sound with two different presets — one for daytime, the other for after dark.
“The TTL11-A provides a unique solution. I would have hung 20 or more conventional speakers to produce a result that delivers coverage to all four corners of this huge restaurant,” he explains. “People are telling me they don’t even notice the speakers because they are so discreet!”
“Everyone is delighted with the quality of the sound; it’s crisp and clear and genuinely multi-purpose.”