Audio

Wednesday, July 18, 2012

A “Test” To Evaluate Your Knowledge Of Power

Gauging your knowledge of power as it relates to sound systems

With the wealth of knowledge about power available here on ProSoundWeb it’s a good idea to constantly evaluate our knowledge of power, or actually, your knowledge about power.

Rather than submit you to the typical “right or wrong” questions with exact numerical answers, I’ve elected to provide a different means of self-evaluation.

The test is “open book”, based upon the information shared in the article series and other resources, and I can tell you up front that the answer to every question is “It depends!”

But what’s really being asked is “What does it depend on?”

At first glance, it may seem that the question is not even related to audio. Don’t be fooled. Principles are principles.

You already know this stuff - you just may not know that you know it! Each question also paves the way for a short review of the concept.

That’s it!

Now relax, take out a sharpened number 2 lead pencil and…sorry about that.

1) I want to paint my living room walls and need to buy paint. How much will I need if the ceiling height is eight feet?

Obviously, estimating of the amount of paint requires more information. What we need to know is the area to be covered, which can’t be determined by the ceiling height alone.

The total length of the walls is needed to get the area (length x height). The paint store would also need to know how many windows are in the walls (they can subtract this area from the total), and how absorbent the surface is (one or two coats?). Only then can the required amount of paint be determined.

It’s equally ridiculous to calculate an amplifier’s output power by using its peak voltage rating. As with the wall, the area of a waveform must be known to determine how much power is generated.

This requires amplitude information (like ceiling height) and knowledge of length (time). We also need to know how much to subtract for higher crest factors (less intense program - like windows in the wall).

And lastly, we need to know how much the load will soak up (porosity of the surface). Think of one coat as eight ohms and two coats as four ohms. And two ohms? Don’t even think about it!

2) Which stock will yield the greatest earnings?

We’ve all learned this one the hard way. Stock A has some high amplitude values, but doesn’t last long. Stock B has lower “highs” but is more consistent over time.

Like painting walls and electrical waveforms, it’s all about area. An amplifier can have a very high peak rating, but may fizzle when loaded for long spans of time (that all-day outdoor show). Make sure that you look at the long-term continuous output power when shopping for amplifiers.

Short-term peak ratings are large numbers, but they don’t tell the whole story.

3) Which song will make the loudspeaker hotter?

This should be obvious by now. Grungy, highly compressed rock and roll has a much lower crest factor (more area) than an “audiophile” recording of a sitar solo.

Both types of music may occasionally light the clip light, but the R&R is much more likely to toast the loudspeaker.

4) How much must I increase the power applied to a loudspeaker to make it a little louder?

A bunch - 3 dB represents a modest change in sound level, yet a 3 dB increase requires the amplifier to generate twice the power.

So every time you turn it up “a little”, you are doubling the power to the loudspeaker. No wonder so many loudspeakers succumb to the last song of the evening.

5) How much of an amplifier’s rated power will the amplifier likely have to generate in a music playback system?

Not much. Given a typical crest factor of 20 dB for live music, the amplifier’s output power could, on average, be about one watt per 100 watts of rated power.

That kilowatt monster that you bought with the home improvement loan will likely need to generate about 10 watts continuous. If you break out the compressor/limiter, you may get this up to 100 watts, but that’s about it.

6) So why buy a big amplifier? After all, they’re expensive!

Loudness and generated power are all about area. Clarity is all about headroom. If program peaks get clipped by a small amplifier, it sounds like trash.

Amplifiers must be oversized relative to their average output power by a factor of 10 to 100 to allow for signal peaks. This translates into 10 to 20 dB of headroom. If you have deep pockets and an understanding wife, go for 20 dB.

If not, spend some of the money saved on a hard limiter to make the program peaks “fit” though the amplifier. All of this makes sense only if you look at power using the decibel.

Click to enlarge

7) Which subwoofer is better? One that handles 100 watts or 500 watts?

You simply can’t tell from the power rating alone. It’s just a “waste disposal” number. How much sound can each produce? This is the efficiency rating.

A 15-inch bass horn sitting in a corner and consuming 100 watts continuous could easily be much louder than an 18-inch in a sealed box hanging in free space and consuming 500 watts continuous.

It’s not what’s fed in, it’s what comes out. See the stock market question (Question 2, above) for an object lesson on this.

8) Can a 30-pound amplifier really keep up with a 300-pound amplifier? My chiropractor wants to know.

Maybe. It depends on what it’s asked to do. Power (like hot water) can be generated “on demand” or it can be pulled from storage.

This is why “Anywhere Gas and Electric Company” dams up rivers to create huge reservoirs for turning hydraulic turbines.

The water flow can remain constant even through the dry season so that the lights don’t dim when the creek gets dry.

Amplifiers with large, heavy power supplies can typically maintain a more constant current flow under severe conditions - like reproducing low frequency synth tones through subwoofers that can peel paint (see question 1) at 100 feet.

Percussive sounds at mid/high frequencies aren’t nearly as “meaty”. You can save your back and your wallet with smaller, lighter amplifiers.

9) Can the loudspeaker’s power rating be trusted? The “Killbox 5000” is rated at 5 kW and the “Lighttones 100” is only rated at 100 watts.

Most reputable manufacturers use standardized power testing to rate their loudspeakers. These methods define the type of waveform, time duration, crest factor and a few other metrics.

These tests are designed by experts to simulate real-world demands on loudspeakers.

They tell you how long the loudspeaker survived under the described conditions. Since they can’t possibly know how the end user will use (or abuse) the product, they can’t guarantee that you won’t blow it up.

Power ratings are very useful for comparing the relative differences between products from the same manufacturer.

When using them to compare one brand to another, make sure that the ratings are based on the same standard (i.e. AES, EIA-426B, etc.).

And even when the standards are used, the “honor system” governs the writing of spec sheets. Since overrated loudspeakers generally don’t kill or maim people, the government doesn’t require validation of power ratings. Only in “Auditopia” are there “data police” that check all the ratings.

You can pinch yourself to wake up now. Also remember that large power ratings can be achieved with resistors (which don’t produce much sound!). That “Killbox 5000” might have a couple of water-heater elements as part of its crossover network.

Click to enlarge

10) What happens if my power amplifier is too large?

It’s better to have an oversized amplifier than an undersized one, provided that one stays within the thermal limits of the loudspeaker. A bigger amplifier is less likely to clip the signal. But there are limits here, too. A piston can only travel so far before it becomes non-linear.

If you hook a zillion watts up to a loudspeaker to provide 40 dB of headroom, and then someone drops a mic, you may see impressions of all of the cones in the metal grills. Over-excursion kills fewer loudspeakers than heat, but it must still be considered.

11) Should I use a rubber band or a chain to pull my boat trailer?

I had to get one in regarding reactance. There’s a big difference between the impedance of a real loudspeaker and that of the non-inductive load resistors used to test many amplifiers.

Reactive loads reflect power, and the amplifier has to deal with this. Amplifiers with extended bandwidths are often unstable into reactive loads. If presented the choice between a “20 Hz to 40 kHz” bandwidth spec and a “DC to gamma rays” spec, I would pick the former.

The Point
How did you do? Hopefully this series has provoked some thought. Power ratings are useful for getting a general idea of the performance of a device, but as I have shown, there are many variables and caveats when you assign numerical ratings to amplifiers and loudspeakers.

Don’t give these numbers any more or less attention than they deserve. They are but one piece of the puzzle, and probably not the most important piece.

Also, don’t be afraid to do your own power testing. The A/B comparison of two products is still the best way to tell the difference between them, regardless of what the numbers say. Music at full volume might be a more relevant power test for your system than pink noise.

Just be sure going in which party is paying for the toasted voice coils.

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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Posted by admin on 07/18 at 09:25 AM
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Clear-Com Improves Communication At Hope Community Church

Clear-Com brings flexibility to the communications backbone at Hope Community Church in Raleigh, North Carolina. The worship facility has installed Clear-Com’s Tempest2400 digital wireless intercom to manage its weekly services and children’s productions, enhancing its overall communications workflow.

Clear-Com brings flexibility to the communications backbone at Hope Community Church in Raleigh, North Carolina. The worship facility has installed Clear-Com’s Tempest2400 digital wireless intercom to manage its weekly services and children’s productions, enhancing its overall communications workflow.

Tempest2400 allows the Hope Community Church’s audio, lighting and video departments, located throughout the facility’s 1,500-seat sanctuary, to coordinate critical production cues.

Previously, the church depended solely on a wired partyline intercom system for this. While reliable, the system inhibited the movement of the cameramen, technical director and stagehands.

This prompted the church to choose a wireless intercom to augment its communications. The system would also need to address the RF interference issue caused by the church’s 250 laptops, wireless microphones and consumer wireless devices.

Since the Tempest2400 wireless system offered freedom and flexibility as well as interference-free communications even in Wi-Fi crowded environments, it was the logical choice for the church.

“Part of my role is to oversee all the production elements taking place during the service. For me to not be tied to one place by a piece of cable is definitely a win,” says Bob Blair, Technical Director for Hope Community Church. “Also, we have two camera operators moving on stage, so a wireless intercom made sense to minimize the amount of cables that those guys were tethered to there.

“Tempest gives us the reliability we need for cues to be heard while providing us with the option to easily and effortlessly change the setup when needed.”

At Hope Community Church, a two-channel Tempest2400 master BaseStation is installed at the front-of-house position, with one channel feeding a submaster BaseStation in the video control room. The system has five beltpacks, including two for the cameramen working on stage, one for the video director, one for the technical director, and one for the stage manager. 

The Tempest2400 has been integrated with the existing four-channel partyline system to aid communications with the rest of the technical crew, including the staff manning the stage, those recording the services backstage, additional cameraman in the back of the auditorium and staff in the green room.

In addition to the four weekly services—two on Saturday and two on Sunday—Hope Community Church also hosts kids-oriented sitcom productions every month. The Tempest system comes in handy here as well, particularly for the stagehands, because it enables communication and mobility.

“Our kid’s productions demand one or two stagehands in addition to the one or two camera operators on stage,” Blair explains. “Previously, the stagehands had wired intercoms and would take off their wired beltpacks to move set pieces.

“That would leave us to run the rest of the production elements without knowing what was going on while we were waiting for them to run back to their wired beltpacks and headsets. Now, the stagehands run on stage with their Tempest BeltStations to do what they have to do and run back, and we can stay informed the whole time.”

Tempest2400, which operates in the 80 MHz of spectrum in the 2.4 GHz ISM frequency band, does not interfere with traditional wireless microphone or in-ear monitor systems operating in the UHF band.

Because of its Frequency Hopping Spread Spectrum (FHSS) technology, the Tempest system does not compete with signals from other 2.4 GHz wireless devices, minimizing frequency coordination and enabling flawless performance.

Further, with state-of-the-art Redundant Data Transmit (2xTX), which sends each packet of audio data twice on different frequencies, the system ensures uninterrupted audio communications. 

Clear-Com

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Posted by Keith Clark on 07/18 at 08:28 AM
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Tuesday, July 17, 2012

Harrison Introduces Mixbus v2.1 With Cloud Integration

Harrison Console's Mixbus is a virtual audio mixing console with an embedded DAW, making it an excellent tool for learning console mixing techniques as well as complex audio editing and processing. Mixbus presents controls in a what-you-see-is-what-you-get format with knobs for EQ, Filter, Compression, Limiting, Saturation, and associated metering available directly on the screen without opening plugin windows.

Harrison Consoles  has introduced a new version of Mixbus with special features and pricing that make it the preferred choice for educational facilities.

Mixbus is a virtual audio mixing console with an embedded DAW,  making it an excellent tool for learning console mixing techniques as well as complex audio editing and processing.  Mixbus presents controls in a what-you-see-is-what-you-get format with knobs for EQ, Filter, Compression, Limiting, Saturation, and associated metering available directly on the screen without opening plugin windows.

Mixbus also provides full workstation features including industry-standard plugins, automation, nondestructive editing, crossfades, import/export, and bouncing.  Windows runs on Windows, OSX, and even Linux.

Mixbus 2.1 includes three new features that realize the benefits of “cloud computing”:

- Importing directly from Freesound.org allows users to search a massive online database of royalty-free sounds, searchable by tags such as “drums”, or “car engine”, and import them directly into a session from the Mixbus import window.

- Exporting final mixes to Soundcloud.org directly from the Mixbus export dialog allows users to securely share their mixes worldwide via the power of cloud computing.

- New “Metadata” dialog allows users to store critical metadata inside the audio session, and also store critical global user metadata to speed up operations such as BWAV region-stamping, session copyright dates, and login details for included services such as Soundcloud.

Metadata stays with the session and can be searched, sorted, stored and scripted using tools that recognize the industry-standard XML format.

The reason Mixbux is ideal for education is that it includes a real mixer console window; it acts as a bridge between analog signal flow and DAW technology. 

Students who are taught basic signal flow on analog mixer will find an easy transition to DAWs using Mixbus or Mixbus can be used in computer-only curriculums to introduce students to analog console operation.

Mixbus incorporates many best-in-class DAW concepts such as Playlists, Ripple Edit, Transient Detection and “Smart” tools.  It also includes pre-configured keyboard shortcuts that match some popular DAWs.
Mixbus was specially designed to incorporate the best practices that have evolved in other DAWs.

The new “Freesound” import allows students to search a massive online database for royalty-free sounds which can be used in their sessions.“Metadata” window stores information about the user, session, and the course/instructor.  This information is available in the XML session format and may be searched, sorted, and stored for future reference.

Portions of Mixbus are open-source.  Advanced classes can view the source code of Mixbus, develop plugins, or otherwise take advantage of the open architecture.  Session files use an open text-based format that can be operated on with scripts or versioning systems: perfect for streamlining the educational workflow.

Mixbus is cross-platform and has the widest compatibility of any DAW available.  Because Mixbus runs on Windows, MacOS, and even Linux, your students can use Mixbus on any computer they have available.
Mixbus uses industry-standard CoreAudio, ASIO, and JACK I/O; and it loads VST, AudioUnit, and LV2 plugins (depending on platform).

Facility administrators appreciate Harrison’s sterling reputation at worldwide facilities including Full Sail, Sony, Universal, ARRI, Mosfilm, Soundfirm, and more. 

To accommodate the tight budgets of schools and students, Harrison is providing 3 new payment methods for Mixbus:

1) Anyone may purchase a Mixbus Subscription for $49 +  $9/month.
2) Facilities may purchase a recurring yearly site license for only $999 which provides 10 instructor/lab licenses.
3) Students may subscribe to Mixbus for only $9 per month, waiving the $49 initial payment!  This makes it VERY cost-effective for students to install Mixbus on their own systems.

All three new plans provide free updates to Mixbus ( including major and minor version updates ) during the period of the subscription, as well as email support from Harrison.  Subscribers may cancel their subscription after the initial 3 months, and continue to use Mixbus without the free updates or email support.  Mixbus is also available for the one-time payment of $219.00 (MSRP)

Instructors who are interested in evaluating Mixbus should contact Harrison Consoles at .(JavaScript must be enabled to view this email address).

Harrison Consoles

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Posted by Keith Clark on 07/17 at 04:47 PM
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Piton Engineering Introduces New Rack Handle For Rack-Mount Products

Handles mount outboard of the front panel space with supplied hardware -- no special panel prep is required

Piton Engineering has Introduced the Rack Handle, an assembly that provides handles and captive fasteners for rack-mount products. 

The Piton Rack Handle has a robust molded design with integrated screws that secure your enclosure to any equipment rack. The handles mount outboard of the front panel space with supplied hardware—no special panel prep is required.

The package provides customers with stainless steel fasteners, captive to their product, simplifying rack mounting. No more lost or mismatched screws. Captive fasteners are integral to the handle, eliminating the cost of press-fitting or swaging separate captive fastener assemblies.

Handles mount on the outer edges of the front panel, preserving enclosure volume and front panel space; a key benefit where rack and floor space are at a premium.

It is supplied as a kit with two handles, four captive screws and four mounting screws. The single part number kit reduces the cost of documentation, procurement, kitting and assembly. It can also be used with subracks where no fixed front panel exists.

Because the handle works with standard EIA-310 front panel mounting slots, it can be added to existing designs and products. Custom colors and M5 screws available.

The Piton Rack Handle is available with quantity price breaks at 25, 50, and 100 kits.

Handles mounts on the outer edges of the front panel, preserving enclosure volume and front panel space; a key benefit where rack and floor space are at a premium.

Features:

—High strength ABS/PC polymer
—Captive #10-32 screws have EIA-310 spacing to work with all equipment racks
—NEBS compliant flammability rating and oxygen index
—Mounts on equipment ears, saving valuable front panel space
—No special panel prep, press-fitting or swaging required
—Single part no. kit simplifies documentation, procurement and kitting
—Designed and manufactured in the USA

Piton Engineering

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Posted by Keith Clark on 07/17 at 12:03 PM
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Harman And The Recording Academy Host “From Mic to Mix”

Open to The Recording Academy members and industry professionals, “From Mic to Mix” featured a live acoustic recording session by the pop/soul group Elaine Faye & Company, engineered by 2-time GRAMMY winner Nathaniel Kunkel, and showcased the latest recording products and technologies from AKG, JBL Professional and Lexicon.

Harman Professional and the Producers & Engineers Wing of The Recording Academy hosted their first-ever “From Mic to Mix” event at Los Angeles’ Conway Recording Studios.

Open to The Recording Academy members and industry professionals, “From Mic to Mix” featured a live acoustic recording session by the pop/soul group Elaine Faye & Company, engineered by 2-time GRAMMY winner Nathaniel Kunkel, and showcased the latest recording products and technologies from AKG, JBL Professional and Lexicon.

Nathaniel Kunkel has earned GRAMMY Awards for his work with B.B. King and Robin Williams, received an Emmy for “A&E In Concert: Sting: Sacred Love” and has recorded a “who’s who” of top artists including Brian Wilson, Stevie Nicks, Ringo Starr, Zooey Deschanel, Cobra Starship, Lyle Lovett, Linda Ronstadt and many others. He also owns Studio Without Walls in LA.

The surround sound recording session took place in Conway’s large Studio C using a variety of AKG microphones and headphones, including a C12VR, P820Tube, C414XLII, C214, C451 and a pair of C414XLS.

The attendees were able to listen to Kunkel’s mix with 24 AKG K271 and K240MKII headphones placed around the studio. In the control room, Kunkel monitored using a JBL LSR6328P system in a 5.1 surround sound configuration. Ambience and reverb were provided by a Lexicon PCM96 Surround system.

In a separate control room, Lexicon demonstrated PCM96 and MX400 reverb processors as well as the I-O U42S USB audio interface. Nathaniel Kunkel gave attendees insight into his approach to micing, studio monitors and use of effect processing.

“The sounds came together in a very short amount of time,” Kunkel said. “For several years, I have relied on JBL studio monitors for their accuracy. I never have to second-guess my mix. While the AKG C414 and C12VR have been go-to mics for me, I was pleased to try AKG’s new Perception mics and headphones on this session.

“The Lexicon PCM96 Surround system was the only reverb I used and it added polish that put the mix over the top. The session went smoothly and I’m really pleased with the end result using the great HARMAN gear.”

“The P&E Wing evening at Conway was a great opportunity for us to meet with the engineers and producers on the front line shaping today’s recording landscape,” said Peter Chaikin, Senior Marketing Manager, Recording & Broadcast, JBL Professional. “Elaine Faye’s performance and Nathaniel’s stunning sounds coming across on our speakers made P&E Wing members smile.”

“We wanted to show attendees how well AKG microphones and headphones, JBL studio monitors and Lexicon reverb and effects processors perform in a real-world setting, and how they can help engineers make better recordings from the moment the sound hits the mic to when they’re putting the finishing touches on the final mix,” said Joe Wagoner, product marketing manager for wireless, tour and install, AKG.

Harman

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Posted by Keith Clark on 07/17 at 07:19 AM
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Armagh Theatre Installs New Allen & Heath GLD Console

The GLD mixer is comprised of a GLD-80 mixer plus an AR84 I/O rack in the control room, and an AR2412 rack down on the stage floor. The installation is part of a wider overhaul of the venue’s analogue audio infrastructure, which includes installing Cat5 around the building to allow connection of an expander I/O rack to the GLD-80 mixer.

Market Place Theatre & Arts Centre in Armagh, Northern Ireland, is one of the first venues in the world to install Allen & Heath’s new GLD digital mixer.

The GLD mixer is comprised of a GLD-80 mixer plus an AR84 I/O rack in the control room, and an AR2412 rack down on the stage floor.

Replacing an existing analogue desk, the installation is part of a wider overhaul of the venue’s analogue audio infrastructure, which includes installing Cat5 around the building to allow connection of an expander I/O rack to the GLD-80 mixer.

“We have a team of four technicians, and we all put forward names of desks we thought may be suitable, as well as consulting industry colleagues,” explains technician, Gary Bawden. “We demoed a few desks before making our choice but the GLD came out on top for price, ease of use and input capacity.

“Moreover, it sounded the best in our comparison listening tests.”

The venue hosts many different types of events, including theatre, music, dance, comedy and conferences, which the new GLD system will be managing.

“We’ve been really impressed with how easy it is to configure and move things around on the desk. Being able to colour code things is a neat feature. Also, the way this desk accesses auxiliaries is really good, almost a reverse logic of most other desks. The RTA is amazing and makes it so easy to squeak your monitors,” concludes Gary.

Allen & Heath

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Posted by Keith Clark on 07/17 at 06:43 AM
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Vintage King Audio Opens New Showroom And Demo Studios In LA

Vintage King Los Angeles (VKLA) is displaying a variety of classic and current gear in the state-of-art facility, including eight analog and digital consoles, twenty pairs of monitors, fifty microphones, and dozens of preamplifiers and outboard processors.

Vintage King Audio has opened VKLA, a private Los Angeles showroom and demo studio on Sunset Blvd., to better serve pro audio professionals in the heart of the world’s biggest recording market.

“We’re extremely proud of VKLA and eager to give our clients access to this extraordinary facility,” remarked industry vet Tom Menrath, recently named head of VK’s Strategic Development. “VKLA provides an unprecedented opportunity to compare the finest gear in the world in a proper recording studio environment.

“Visits to the facility will be arranged by appointment, so each customer will get a personalized listening experience.”

Vintage King Los Angeles (VKLA) is displaying a variety of classic and current gear in the state-of-art facility, including eight analog and digital consoles, twenty pairs of monitors, fifty microphones, and dozens of preamplifiers and outboard processors.

Clients can test out nearly a dozen newer consoles such as the API 1608 and the SSL 948, as well as boutique gear from Barefoot Sound, Shadow Hills Industries, Neve, Dangerous Music, Burl, Telefunken, and other top industry brands.

“Our VKLA headquarters is really a new way of giving the customer the most enjoyable buying experience,” explained Shevy Shovlin, Director of Partner Marketing/PR. “We wanted VKLA to be musical, a place that working artists could appreciate. Our long-term goal is to create a real cultural center for the recording scene here in LA.”

VKLA goes beyond the familiar functions of a sales dealership by connecting with the professional audio community through hosting events, panel discussions, and product launches. Plans are in place for a series of educational workshops where seasoned pros will share their knowledge with a new generation of music makers.

Vintage King Audio

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Posted by Keith Clark on 07/17 at 06:28 AM
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Ana Popovic Records In Ardent’s New SSL Duality Studio C

Blues sensation Ana Popovic, a "Best New Artist" W.C. Handy Award-nominee, is recording her next release at Ardent Studios on a Solid State Logic console in Studio C (Memphis, Tennessee).

Blues sensation Ana Popovic, a “Best New Artist” W.C. Handy Award-nominee, is tracking her next release at Ardent Studios in Memphis.  Born in Belgrade, Serbia, Popovic formed her first band when she was 19, gaining a strong following throughout Europe and with her band, Hush, opening for such Blues stalwarts as Junior Wells.

Recorded in Ardent Studios’ new Solid State Logic Duality-equipped Studio C, musicians included Frank Ray Jr., Hammond B3; John Williams, bass; and Tony Coleman, drummer for BB King and producer of the new Popovic album. Longtime Ardent engineer Pete Matthews recorded the new material.

“This was one of my favorite sessions of all times,” commented Popovic. “We cut seven great tracks and the amazing sounding studio C at Ardent made this session stand out from all my previous recordings. Thanks to engineer Pete Matthews, the control room sounded just like the record at all times!  And working in the same room where Van Morrison and Bob Dylan recorded simply adds to the magic of the moment. “

Popovic’s album “Still Making History” (2007) was in the Billboard charts for 19 weeks and her “Blind for Love’ (2009) made it to the #1 radio-played Blues album on US radio.  In 2011, Popovic was winner of the Best Blues DVD at the Blues Matters Awards (UK), and her “Unconditional” was a triple nominee for “Contemporary Blues Album,” “Contemporary Blues Female Artist” and “Best DVD” (“An Evening at Trasimeno Lake”) at the 33rd Blues Music Awards in Memphis.

“We cut some blues, funk and soul, the way it used to be done back in the day, but it’s not a retro record,” remarked Popovic.  “We had the spirit of Albert King and Albert Collins in the room, and we’re coming back in August to finish this project.”

Solid State Logic

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Posted by Keith Clark on 07/17 at 05:33 AM
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Monday, July 16, 2012

A Practical Guide To Key Audio Calculations

Thanks to modern technology, you can do dB calculations without knowing a thing about the mathematics of logs, anti-logs, ratios, exponents, or even much about math

This is a practical guide to doing audio calculations, particularly dB (decibel) calculations, covering most common situations.

You see dB numbers all the time in audio, and probably understand that 3 dB is considered a just noticeable change in volume level.

You may also be aware that dB calculations involve “logs” (logarithms). But perhaps you’re not quite so clear on how to figure out what 24 dB from your mixing console means to your amplifier rated for 1.4V input sensitivity.

Thanks to modern technology, you can do dB calculations without knowing a thing about the mathematics of logs, anti-logs, ratios, exponents, or even much about math. This article is necessarily long in order to cover many situations where you might need to use dB.

But, if you understand the calculations, you will find that most of them repeat the same things in different ways. If you begin to see this, it means you are beginning to understand how to calculate dB.

INTRODUCTION
First, if you don’t have one, you need to buy yourself a cheap scientific calculator. It MUST have several specific functions on it. One is a “Log” function. There are two common Log functions. One is “Log base e” or “natural Log” which you DON’T want. The other is “Log base 10” which is what you need. You can check if it is “Log base 10” by simply entering 10 and hitting the “Log” key. The display should read 1. If it doesn’t, don’t buy it.

Another necessity is a 10x function (also called 10 to the x function or the anti-log function). You can check if this is the proper function by entering 2 then hitting the 10x key. The display should read 100. You also need a +/- (change sign) key. It must also have plus, minus, divide, multiply, and equal (=) keys. You can basically ignore the other functions on it to do dB calculations.

Once armed with this tool, you can now learn to do dB calculations. Actually, you don’t really have to learn much except to push the right buttons on the calculator.

In each example, you will be told exactly what calculator key to hit and what your answer should be. Once you get the right answers as shown, you can substitute your own numbers in the various examples to figure out your own things.

Remember, a calculator is a DUMB device. It will only do what you tell it to do. So you MUST use your brain a bit to see if your answers make sense. This means if you come up with a number like 23841 dB or 20,000,000 watts, it is wrong. Nothing in audio has 23841 dB of anything and 20,000,000 watts is out of the question, unless you are providing sound reinforcement for Space Shuttle launches.

All the answers given in this article only show the first two digits (numbers) to the right of the decimal point. Because fractions of a dB or a watt rarely have little practical significance in audio, these two digits are shown ONLY so you know you got the right answer.

The answer on your calculator may be 15.84893192 but the answer given here is 15.84. As you will see in one of the examples, chopping off digits like this can lead to slight, but not significant, errors. Note again the last digit to the right in the examples is NOT a rounded off value in the examples, but chopped off. If it was rounded off, the answer would be shown as 15.85.

Most calculators can be set to display only 2 digits after the decimal point. It rounds off numbers to do this so that 15.84893192 becomes 15.85. So, if you set the calculator to do this for figuring out the examples, your last digit to the right may be 1 larger or smaller than in the examples.

For actual audio work, you should only be concerned with dB numbers to the left of the decimal point. Thus an answer of either 15.84 or 15.85 dB should be rounded off to 16 dB when stating the result.

For each example calculation, the actual formula being used for the calculation is also shown in red. The actual calculation procedure is in blue.

The most common “Log” calculations you need are: dB to voltage, voltage to dB, voltage gain to dB, dB to voltage gain, calculating SPL for distances, and converting amplifier watts to SPL changes or SPL changes to amplifier watts. Therefore, these are the only examples given. With a little brainwork, you may be able to apply the example calculations to figure out other things.

DEFINITIONS
Some definitions to know:

Any dB value is a RATIO, meaning it represents one number divided by another. If you simply state something in dB then you are only stating the ratio in between one thing and another. So you might say the difference in two voltages is 6 dB but that only means one voltage is twice the other (6 dB = 2 times voltage). It doesn’t tell you anything about the actual voltages.

If you want to state the actual value of something in dB, most common audio calculations have a 0 dB reference value that is indicated by suffix (a letter or letters following dB). The 0 dB reference value is always used as one of the numbers for the ratio. You should always use a suffix when stating the dB of something as an actual value so that anyone else will know what 0 dB reference is.

So when you say your mixer is putting out + 6 dB, you really need to say + 6 dBu or whatever the 0 dB reference is. Thus +6 dB means you have twice as much voltage while 0 dBu means you have 0.775 volts or 1.55 volts.

For electronic calculations (voltage and wattage) the 0 dB references are: 0 dBu (or dBv) = 0.775 volts
0 dBV = 1 volt
0 dBm = 1 milliwatt (0.001 watts). The standard reference value for 0dBm is 0.775 volts into a load of 600 Ohms. For most audio calculations, simply assume you are NOT using a 600 Ohm load and thus dBm can equal dBu.

For example, a +24 dBm output specification on a mixing console can be used as +24 dBu to calculate voltages. The reason for this is that modern audio equipment will put out the same voltage whether there is a “load” on it or not. Thus knowing the power in a line level audio circuit is of little value and simply complicates what you need to know.

Another example: 6 dBu is the ratio of some voltage divided by 0.775 volts. That voltage is 1.55 volts.
For SPL (Sound Pressure Level) calculations the 0 dB reference is:
0 dB SPL = 0.00002 pascals (a pascal is a measure of pressure, in this case air pressure, just like a meter is for distance).

Another example: 100 dB SPL is the ratio of some sound pressure to 0.00002 Pascals. That pressure is 2 Pascals
These are the symbols used in the formulas:
”x” means multiply
”/” means divide
”^” This symbol indicates what follows is an exponent of the number preceding it. An exponent means “raised to the power of”, as in 10^2 is 10 raised to the power of 2 or more simply stated as “10 squared”. With dB calculations you get “funny” powers like 10^(34/20). Spelled out this is “ten to the power of thirty four divided by twenty”. Don’t be afraid of this. The instructions and your trusty calculator will get you through it without having to fully understand it.

”( )” in the formulas means that everything inside the parenthesis is calculated first to come up with a single number. In the first example below, (24/20) is calculated first to come up with 1.2. Then 10 is raised to the power of 1.2.

dBu TO VOLTAGE
How many volts does a mixing console put out with a maximum rating of +24 dBm?

Although rated in dBm you can “change” this, as stated above, to dBu for this calculation. Using you new calculator you need to find out how many volts +24 dBu is above 0 dBu.
Formula: Volts = 10^(dB/20) x volts @ 0 dBu or 10^(24/20) x 0.775
Enter 24 (dBu)
Hit the divide key
Enter 20
Hit the = key
Your answer should be 1.2
Hit the 10x key.
Your answer should be 15.84
Multiply this by 0.775
Your final answer should be 12.28 volts

The reason you multiply by 0.775 is that any dB number is always a RATIO (one number divided by another). So 15.84 is the numerical ratio of +24 dBu to 0 dBu. Put another way the voltage at +24 dBu is 15.84 times bigger than 0.775 volts so you must multiply 0.775 by 15.84.

The 15.84 is used as a multiplier and will be called this from now on. However, to be mathematically correct, this multiplier is actually a ratio representing one number divided by another.

dBV TO VOLTAGE
Suppose the output was rated as +24 dBV? What would its voltage be?
Formula: Volts = 10^(dB/20) x volts @ 0 dBv or 10^(24/20) x 1
Enter 24 (dBV)
Hit the divide key
Enter 20
Hit the = key
Your answer should be 1.2
Hit the 10x key.
Your answer should be 15.84
Multiply this by 1
Your final answer is, of course, still 15.84 volts
In this case +24 dBV is 15.84 times larger than 1 volt.

GAIN
Your amplifier puts out 70V with a 1.4V input. How much gain does it have?
Formula: dB = 20 x Log (volts1/volts2) or 20 x Log (70/1.4)
Enter 70 (volts1)
Hit the divide key
Enter 1.4 (volts2)
Hit the = key
Your answer should be 50.00
Hit the Log key
Your answer should be 1.69
Hit the multiply key
Enter 20
Hit the = key
Your answer should be 33.97 dB gain

Suppose you only knew your amplifier had 33.97 dB voltage gain (we’ll round this up to 34 dB). What would its maximum output voltage be?
Formula: Multiplier = 10^(dB/20) or 10^(34/20)
Enter 34 (dB)
Hit the divide key
Enter 20
Hit the = key
Your answer should be 1.7
Hit the 10x key
Your answer should be 50.11

The voltage at the output will be 50.11 times bigger than the voltage at the input. (Note: this number appeared during the first calculation for this amplifier as 50.00 but we rounded up the gain from 33.97 dB to 34 dB.)

Take the next step. The input sensitivity on the amplifier is 1.4 volts.
Formula: Volts Out = Multiplier x Volts In or 50.11 x 1.4
With the 50.11 (multiplier) still displayed, hit the multiply key
Enter 1.4 (volts in)
Hit the = key
Your answer should be 70.16 volts

Thus, your amplifier will put out 70.16 volts with a 1.4 volt input with the input control at maximum. If you wanted to put 8 volts in your amplifier it will clip unless you turn the input control down. But, your control is marked in dB, so how far do you turn it down? Not a problem to figure out. Read on.

VOLTAGE TO dB
The question is what is the difference in dB between 1.4 volts and 8 volts?
Formula: dB = 20 x Log (volts1/volts2) or 20 Log (1.4/8)
Enter 1.4 (volts1)
Hit the divide key
Enter 8 (volts2)
Hit the = key
Your answer should be 0.17
Hit the Log key
Your answer should be -0.75
Hit the multiply key
Enter 20
Your answer should be -15.13 dB

So guess what? Turn your input control down to the -15 dB point and now when you put 8 volts in, the amplifier will put out its full 70.16 volts. Why? Back to the calculator.

You amplifier has 34 dB gain. You turn your input control down 15 dB so now it effectively has 34 dB - 15 dB = 19 dB gain between the input jack and the output.
Formula: Multiplier = 10^(dB/20) or 10 ^ (19/20)
Enter 19 (dB)
Hit the divide key
Enter 20
Hit the = key
Your answer should be 0.95
Hit the 10x key
Your answer should be 8.91

Your output will now be 8.91 times bigger than the input. Your input is 8 volts so multiplying 8 times 8.91 gives you 71.30. Well, that’s not exactly the 70.16 volts we got before. Why? Because the numbers were chopped off in these calculations. If you used the numbers with all the digits, the answer would have come out as 70.16621271 volts which is the precise answer arrived at previously. Is this difference significant? No. You can find out why it isn’t by using the next example calculation to find the dB difference between 71.30V and 70.16V.

Suppose you have one device that has a maximum 15.5 volt output and the device you wish to drive with it accepts a maximum input of only 7.75 volts. What is the dB difference?
Formula: dB = 20 x Log (volts1/volts2) or 20 x Log (15.5/7.75)
Enter 15.5 (volts1)
Hit the divide key
Enter 7.75 (volts2)
Hit the = key
Your answer should be 2
Hit the Log key
Your answer should be 0.30
Hit the multiply key
Enter 20
Hit the = key
Your answer should be 6.02 dB
The output of the first device is 6 dB more than what the second device can accept.

You’ll notice that in the first voltage to dB calculation you ended up with a minus dB number and in this one a plus or positive dB number.
+dB AND -dB

In the last example, if you used the formula 20 x Log (7.75/15.5), your answer would be -6.02 dB. It simply depends whether a larger number is divided by a smaller one (answer is always +dB) or a smaller one is divided by a larger one (answer is always -dB). The basic number will be the same no matter which way you divide. Try reversing the 7.75 and 15.5 in the last example. You should get -6.02 as the answer.

Also, if you calculated the dB difference between the 71.30V and the 70.16V for the amplifier outputs, you would have gotten 0.139 dB or -0.139 dB, depending on which number you divided by which. This is why the difference was not significant: you can not hear a 0.139 dB difference.

The main reason for dividing a smaller by a larger number when calculating dB is to figure out the LOSS in dB. You will see this in the SPL and amplifier calculations below.

VOLTS TO dBu or dBv
You have a device that puts out 8.8 volts. What is that in dBu?
Formula: dBu = 20 x Log (volts1/volts2) or 20 x Log (8.8/0.775)
Enter 8.8 (volts1)
Hit the divide key
Enter 0.775 (volts2)
Hit the = key
Your answer should be 11.35
Hit the Log key
Your answer should be 1.05
Hit the multiply key
Enter 20
Hit the = key
Your answer should be 21.10 dBu

To find dBv simply substitute “1” for “0.775” in the calculation. Your answer should be 18.88 dBv.

SPL CALCULATIONS
You do SPL calculations using exactly the same formulas as for voltages. Both SPL and voltages are “pressures” and the factor “20” is used for both. Your loudspeaker puts out a maximum of 120 dB SPL at 3.3 feet = 1 meter. What is the SPL at 60 feet?

First, you must understand that SPL drops 6 dB for each doubling of distance. Why is this? Calculate the multiplier for a pressure change of 6 dB.
Formula: Multiplier = 10^(SPL/20) or 10^(6/20)
Enter 6 (dB SPL)
Hit the divide key
Enter 20
Hit the = key
Your answer should be 0.3
Hit the 10x key
Your answer should be 1.99 = about 2

So the multiplier for a 6 dB SPL change is 2. This means if you move twice as close to the loudspeaker, it will be 6 dB louder.

Now recalculate this using another function on your calculator called the change sign key. This changes a number in the display from a plus to a minus number.
Formula: Multiplier = 10^(-SPL/20) or 10^(-6/20)
Enter 6 (dB SPL)
Hit the “+/-“
Your display should change to -6 (dB SPL)
Hit the divide key
Enter 20
Hit the = key
Your answer should be 0.3
Hit the 10x key
Your answer should be 0.50 = 1/2

So the multiplier for a -6 dB SPL change is 1/2. This means if you move so you are only 1/2 as close (meaning twice as far) to the loudspeaker, the SPL is 6 dB less.

Back to the problem. Your are 60 feet away and you know what the SPL is at 3.3 feet = 1 meter. You already know this should be a -dB number so you are going to divide the small number by the big one.

Distances are equivalent to voltages so you simply divide the two distances. You can do this for any two distances and substitute the SPL at one of the distances for the Max SPL in this example.

If this SPL is for the further distance, you must divide the further by the nearer distance. Always divide the distance you are finding the SPL for into the distance for which you know the SPL.

Otherwise, you will find your answer will not make sense, such as having more SPL at a further distance. Sound just doesn’t seem to work this way.
Formula: dB SPL = Max SPL + (20 x Log (distance1/distance2)) or 120 + (20 x Log (3.3/60)
Enter 3.3 (distance1)
Hit the divide key
Enter 60 (distance2)
Hit the = key
Your answer should be 0.055
Hit the Log key
Hit the multiply key
Enter 20
Hit the = key
Your answer should be -25.19
Hit the + key
Enter 120
Hit the = key
Your answer should be 94.80 dB SPL

Note: the SPL formula is only valid outdoors and for the direct sound from the loudspeaker indoors. Indoors the reverberation of the room will take over at some distance from the loudspeaker and the sound level will remain more or less constant the further you move away. This is a subject beyond the scope of this article, but it will at least give you a rough idea of what to expect indoors.

AMPLIFIERS
There are really only two dB calculations you generally need for amplifier outputs.

One is how many dB SPL is there between two wattage levels.

The other is what is the difference in watts you need to achieve a given change in dB SPL.

Because amplifier watts are power, that number 20 you’ve been using thus far in the calculations changes to 10 for doing the “log” calculations.

Otherwise, the formulas are the same. Through a quirk in mathematics, you can use the dB answers you get when calculating differences in wattages as representing the change in SPL from the loudspeaker.

While this is not mathematically correct, it is a perfectly practical thing to do. Substitute some other numbers in the following example, such as 2 watt and 4 watts. Then use 2000 watts and 4000 watts. You will find the answer is exactly the same (but the price tags are definitely not!!).

WATTS TO dB (or Watts to SPL)
You have a 100W amplifier and want to change to a 350W amplifier. How many more dB will you get out of your loudspeaker?
Formula: dB = 10 x Log (watts1/watts2) or 10 x Log (350/100).
Note the larger number is divided by smaller will give you +dB. You DO expect higher SPL, don’t you?
Enter 350 (watts1)
Hit the divide key
Enter 100 (watts2)
Hit the = key
Your answer should be 3
Hit the Log key
Your answer should be 0.54
Hit the multiply key
Enter 10
Hit the = key
Your answer should be 5.4 dB

This is the power difference in dB between these amplifiers. If you were going from a 350W to a 100W, the answer would be -5.4 dB as the smaller number would be divided by the larger. Try it.

You could go through correct mathematical conversions, but because we are being practical, the SPL from your loudspeaker will be 5.4 dB greater with the 350W amplifier than the 100W amplifier. Or, it will be -5.4 dB less using a 100W amplifier instead of a 350W amplifier.

dB TO WATTS (or SPL to Watts)
Suppose you want to increase your SPL by 8 dB and you have 100 watts to start with. How many more watts would this take?
Formula: Multiplier = 10^(dB/10) or 10^(8/10)
Enter 8 (dB)
Hit the divide key
Enter 10
Hit the = key
Your answer should be 0.8
Hit the 10x key
Your answer should be 6.30
So you need 6.30 times as many watts to get an 8 dB SPL increase. How many watts is that?
Formula: New Watts = Multiplier x watts or 6.3 x 100
Enter 6.3 (multiplier)
Hit the multiply key
Enter 100 (watts)
Hit the = key
Your answer should be 630W
Thus, if you start with 100W, you need 630W to increase your SPL by 8 dB.

SUMMARY
The above examples should give you valuable tools you need to calculate dB in the most of the ways you need for audio signals. Use them wisely.

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Posted by Keith Clark on 07/16 at 01:34 PM
AVFeatureBlogStudy HallAmplifierAVEducationLoudspeakerProcessorSound ReinforcementAudioPermalink

Extron Announces New High Resolution VGA Line Driver with EDID Minder

Extron Electronics has introduced the new Extender Plus VGA and Audio Line Driver with EDID Minder.

This one input, one buffered output line driver extends VGA-QXGA and HDTV component video along with audio signals up to 250 feet (75 m).

Unbalanced computer stereo audio is converted to balanced, line level stereo audio to eliminate noise usually associated with unbalanced audio when distributed over long cable runs.

It includes EDID Minder, which automatically manages EDID communication between connected devices to ensure that the source powers up properly and reliably outputs content for display.

The Extender Plus is available in Decora-style and AAP form factors, providing convenient AV access and signal extension for a wide variety of environments.

“VGA line drivers are standard in nearly every professional AV system design, but now they need to do more than just boost AV signals,” says Casey Hall, vice president of aales and marketing for Extron. “The new Extender Plus products expand the capabilities of the popular Extron Extender Series by including EDID Minder, a valuable technology that ensures reliable EDID management for computers and displays.”

To maintain signal integrity over long distances, the Extender Plus provides video amplification and peaking control to compensate for attenuation that can occur in long cable runs. Proper signal compensation supports a more detailed image with greater contrast.

Additional integrator-friendly features include an EDID capture mode, selectable resolutions and refresh rates, and real-time status LED indicators for system monitoring. Both versions also include an energy-efficient, external universal power supply for worldwide compatibility.

Extron Electronics

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Posted by Keith Clark on 07/16 at 01:08 PM
AVLive SoundNewsProductAVInstallationInterconnectNetworkingAudioPermalink

Peavey MediaMatrix NION Training Seminar Scheduled For Bay Area In July

Peavey Commercial Audio will host a two-day MediaMatrix NION Certification training course in Emeryville, Calif., during July, where AV designers, consultants, contractors and end users can learn the best practices for designing, deploying and implementing MediaMatrix audio distribution, processing and control systems.

The MediaMatrix NION Certification training course instructs AV professionals on the fundamentals of MediaMatrix, a flexible and scalable audio networking system on the market, as well as how to design and program projects in NWare; how to set up a NION processor; how to create end-user GUIs for nTouch 60 and nTouch 180 touch screens and PC kiosks; and how to integrate the XControl into a MediaMatrix installation.

The MediaMatrix NION Certification training seminar will be held at Meyer Hall on the campus of Ex’pression College in Emeryville on July 30-31, from 8:30 a.m. to 5 p.m. each day.

Each successful student will receive a completion certificate that can be submitted to InfoComm for receipt of 7.5 renewal credit hours for InfoComm CTS, CTS-I and CTS-D.

For full class descriptions and registration, please visit mm.peavey.com/education or http://www.peaveycommercialaudio.com/education.

Peavey has educated thousands of AV system designers, integrators and end users from around the world since MediaMatrix was introduced in 1993.

MediaMatrix courses are also offered online at mmtraining.peavey.com. Completion of the online Peavey MediaMatrix Basic or Advanced course earns two hours of credit toward InfoComm’s CTS and CTS-D Certification Renewal.

Peavey Commercial Audio

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Posted by Keith Clark on 07/16 at 12:07 PM
AVLive SoundChurch SoundNewsProductTrainingAVDigitalEducationNetworkingProcessorAudioPermalink

C3 Productions Brings L-Acoustics KARA Line Source Loudspeakers To Brighton

The south coast-based production and hire company has added 12 KARA cabinets with eight SB18 subwoofers

UK-based L-Acoustics Certified Provider SFL Group has supplied a KARA WST line source system to C3 Productions of Brighton.

SFL Group also demo’d the system for C3 prior to purchase in several London venues and provided training at its Reading premises.

The south coast-based production and hire company has added 12 KARA cabinets with eight SB18 subwoofers, three LA8 amplified controllers and flying hardware to its inventory.

This is C3’s first L-Acoustics purchase, and the company chose KARA because it needed a medium format high quality line array to use across live music, performance arts and corporate live events.

“Our mission statement is to provide our customers with the latest and highest performance technology and this system enables us to deliver that,” says C3 director Jon Crawley. “We investigated all the systems currently available on the market and felt that KARA offered the best combination of cost, performance and gave us a unique product in the local area. It is small enough to be discrete for theatre shows but powerful enough to deliver a full-on rock show.

“We have always been fans of the K1 system, and KARA has a similar clean, powerful and dynamic sound, which suits our core live music business. The achievable levels from the system are hard to believe given the size.”

C3’s KARA system made its debut at one of Brighton’s main annual events, the Great Escape festival, which showcases new talent from around the UK. It was used on the NME stage in the 2,000 capacity Corn Exchange, which featured emerging and existing artists such as Dry the River, Mystery Jets and Bookashade.

“There were some very difference music styles but the system sounded incredible throughout and we only got positive responses from all touring parties,” continues Crawley. The system has since been used at the Jubilee celebrations, for a theatre production on Hastings seafront and will be on the main stage at BLOC weekend in London in July with artists such as Snoop Dogg and Orbital.

C3 Productions
SFL Group
L-Acoustics

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Posted by Keith Clark on 07/16 at 08:25 AM
AVLive SoundNewsAVBusinessLine ArrayLoudspeakerSound ReinforcementAudioPermalink

Friday, July 13, 2012

Church Sound: Establishing Proper Gain Setting On A Mixing Console

One of the primary keys to a good sounding system

The purpose of any sound reinforcement system is to amplify a source, whether that is a single speaking voice, a group of singers, a collection of musicians, a piece of pre-recorded music, or other sound sources, and to deliver that to the listener.

Each individual piece of equipment in the signal chain between the source and the loudspeaker can affect the level of the signal.

That means that the mixing console, outboard effects processors, crossovers, and amplifiers (obviously) can all add to or subtract from the signal level.

The mixing console can almost be considered to be an audio system in itself, containing microphone preamplifiers, equalizers, and perhaps even dynamics processing.

Each includes a gain stage that can amplify the signal. Then there are the input channel faders, subgroup faders, and the master faders. Again, each can add gain to the signal.

Each component part of the mixing console, and piece of audio equipment in the following signal chain, also adds noise to the system, since all electrical circuits produce some level of noise. The amount of noise depends on the circuit design, and may be negligible in the case of a digital device or may be a great deal, especially where analog gear is involved, and particularly with older or poorly maintained equipment.

If you turn everything up in your system with no signal going through it then what you hear coming out of the loudspeakers—that hiss—is the sum of the noise being introduced by everything in the signal path (the equivalent of the tape hiss we mentioned).

The level of hiss that you hear is known as the noise floor. It may also include buzzes from grounding problems or other interference, but those are issues for a separate article.

At the other end of the scale, too much gain introduced by any device in the signal chain will “clip” the signal, which means that the signal peaks are flattened by the circuit’s inability to handle the level, resulting in a distorted sound.

With analog circuits a little distortion is tolerable. Digital clipping, however, just plain sounds nasty, and is to be avoided at all costs.

There is a window in which the system operates at its optimum, where the noise floor does not mask a signal that has been set too low and high-level signals do not distort.

The fundamental aim when setting the gain through a mixing console is to ensure that the loudest signal passes through at a level just below clipping, while also allowing for signal peaks. That extra leeway is known as headroom.

Let’s take a basic signal path, a microphone connected to the input of the mixing console, the output of which is connected to an amplifier and a loudspeaker. To set up the console, begin by turning down all the input gain (also known as trim or input level or input sensitivity) controls. Disengage the pad switches, if there are any.

Pull all the faders—inputs, subgroups, and masters—all the way down. Set the equalizer level controls to the 12 o’clock position or switch the EQ out of the signal path. Turn all of the auxiliary sends down and set the pan (also known as balance) controls to the 12 o’clock position.

On the relevant input channel, select PFL or Pre Fade Listen. This switch routes the input signal to the meter (and usually the headphone output) from before the fader, so the fader position has no effect on the signal.

The loudest level of voice or instrument that is likely to occur during the service or performance should be produced during this setup.

As the person speaks, sings, or plays their instrument, adjust the channel input gain control while watching the LED or VU meter until the loudest sounds peak at between +6 dB and +9 dB on the meter scale.

On a console outfitted with bargraph meters this zone is typically delineated by yellow LEDs. Avoid setting the level so high that the red LEDs illuminate, or the VU meter needle enters the red area, which will result in a distorted signal.

To allow yourself some wiggle room later, back the level off slightly.

By the way, note that not all models of mixing console offer the same amount of headroom.

Once you’re familiar with your console you will know just how much headroom you have before distortion and can setup the levels accordingly.

Repeat the procedure for each of the input sources. Remember that gain can be added at other stages in the console. If you add EQ, for example, you will need to re-check the level using PFL and the meter.

Those numbers around the EQ controls refer to the dB value added or subtracted in the selected frequency range. Since maximum EQ gain is typically 15 dB, that will affect the signal level and may require the input gain control to be reset lower to compensate.

A mixing console typically gives its best signal-to-noise performance—where the amplitude of the signal is proportionally at its greatest compared to the amplitude of the inherent circuit noise—with the input faders at 0. This is also known as ‘unity’ and is often marked by a heavier line or shading next to the fader. At this position, the fader is not adding to or subtracting from the signal level. 

Also, fader scales are generally logarithmic, which means that small movements at the bottom end of the fader travel will have more of an effect than higher up the scale.

You also don’t want to be at the top of the scale, as that will prevent you from being able to add any more level, if required.

Audio subgroups may be used to control a group of instruments or voices on one or two faders. If your console has subgroups, these faders, which receive their inputs from the group bus assignment switches on the input channels, should also be set at 0. If you find while you are mixing that they are more than a little above or below the unity position then you should adjust the input gains slightly.

An audio bus is a means of getting the audio from an input to an output of your choice. Think of an input channel as being similar to a narrow side road that dead ends. Traffic on that road, and other roads, may all need to go to the same destination. To do that, they must all join a larger road—a bus.

Each bus links each input channel with a master channel or section that typically features an output. So, for example, subgroup assign switch #1 on each input channel routes that input signal to subgroup #1, through the subgroup fader, and to an output connector.

In turn, that subgroup may also be assigned to the stereo or LCR (left/center/right) master output via another set of buses, through the master fader or faders, and out of the master output connectors.

Consoles may also include any number of auxiliary send and matrix buses, all fed by switches or level controls on each input channel or the subgroup channels.

As with the input levels, you do not want to overload the console buses and cause distortion. Maintaining a good gain structure that essentially passes the signals through at unity, as described above, will help you to achieve this.

With all of the console gain stages operating properly together, and at the optimum signal-to-noise ratios, you have one of the primary keys to a good sounding system.

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Posted by Keith Clark on 07/13 at 02:07 PM
Church SoundFeatureStudy HallConsolesDigitalSignalSound ReinforcementAudioPermalink

Power Viruses: The Source of Electronic Influenza & The Cures

Prevention of power viruses is possible only when prevention is systematic

Software viruses are common occurrences in the computing world.

In fact, almost everyone is familiar with their potential for damage, and the news media routinely reports the details of system disruption related to their appearance.

It’s a reasonable term to apply to these rogue programs. They enter a system unseen, often incubate in silence, and eventually come to life with results that range from merely annoying to disastrous.

Electrical disturbances are quite similar.

In fact they could reasonably be called “power viruses” since they, too, are unseen and can cause serious and expensive electronic system failure.

POWER VIRUS ORIGINS
Power viruses are contracted the same way as other viruses. They’re passed along - often by your system’s electrical neighbors. Plug your system into the wall, turn it on, and look out. You’ve just been exposed to an epidemic, and there are a lot of very sick electrons looking to cause problems.

Some of them may take time to cause noticeable damage. Others are immediately catastrophic (like a lightning strike).

How do power viruses affect an electronic system? What can you do to prevent power viruses in the first place?

Start by understanding them, and then you can tackle the job of immunizing your system against their harmful effects.

There are six main power viruses that can invade a system. The symptoms they cause can vary as can the proper course of treatment.

Voltage Spikes & Impulses
This virus is mostly the result of electrical equipment inside your facility. Electrical loads like elevators, motors, relays, induction furnaces, copy machines, and similar devices can cause sudden large increases in voltage inside the electrical system.

Conditions outside your facility can be to blame, too. Switching activities by the electric utility and lightning strikes can cause transient impulses so intense they literally “blow up” sensitive micro-circuitry.

This virus is deadly to electronic systems - but not always immediately. Sometimes voltage spikes and impulses are relatively small in amplitude. In these cases, the virus weakens the system components over time leading to deteriorating health and eventual failure.

Other times the impulses may be so large that they cause immediate system failure.

Electrical Noise
Like voltage spikes and impulses, electrical noise is generally created inside the facility by the system’s electrical neighbors. Almost every electricity- consuming device contributes its share of electrical contamination.

Things like appliances, photocopiers, laser printers, and electronic lighting ballasts are all noise sources that can cause computers to lock-up, lose data, or behave unreliably.

Even computers themselves generate electrical noise. It’s truly a paradox that our computers often infect other computers with power viruses.

Common Mode Voltage Problems
Traditionally, this power virus hasn’t received much attention.

But detection of common mode voltage problems is easier and, as a result, more system problems are being traced to its existence.

The condition is characterized by unwanted voltage measured between neutral and ground (the white wire and the green wire or conduit) in the electrical system.

In fact, the common mode voltage virus is probably the most serious power virus infecting electronic systems today.

It occurs as a result of high impedance safety grounds, neutral conductors shared with other circuits, and branch circuit lengths that are excessive.

When the electrical noise virus (already mentioned) appears between the neutral and ground conductors it becomes a common mode virus with the ability to cause lost files, system lockups or re-boots, communication errors, and “no problem found” service calls.

Voltage Regulation

This virus is characterized by abnormal variations in the electrical circuit’s nominal operating voltage (120 volts, for example). These variations are generally greater than ±10 percent of nominal voltage and may last for several line cycles or more.

Traditionally this virus has been referred to as the “sag” or “swell,” and it is typically caused by large loads turning on and off, and overloaded branch circuits or distribution transformers. In some cases, voltage regulation viruses can be the responsibility of the power utility.

If an electronic system requires tightly regulated voltage (most of today’s systems don’t) the voltage regulation virus is likely to cause system lock-ups and unreliable operation in addition to damaged or destroyed components.

Blackouts
Blackouts are the most visible and easily identifiable of all the power viruses. And they have the most obvious cause and effect relationship.

One moment power is present—the next moment it’s not and your system is dead in its tracks as a result. The effects of unanticipated power loss are obvious. This is especially true if the system is a network or some other “fault intolerant” architecture.

Fortunately, in spite of what most UPS manufacturers advertise, blackouts account for comparatively few occurrences of all the power viruses.

Backdoor Disturbance
This virus (as its name implies) infects your system via a secondary path.

Even though they’re not an AC power connection, things like serial ports, telephone lines, network cabling, and I/O connections can all permit power viruses to invisibly enter a system.

This virus causes driver chip failure and communication errors.

The back door disturbance virus is often unrecognized.

Without treatment, serious damage can occur, and lost productivity can result in substantial financial losses as well.

AN OUNCE OF PREVENTION

There’s an old adage that “An ounce of prevention is worth a pound of cure.” Nothing could be closer to the truth when it comes to power viruses.

We’re familiar with the damage that results from software viruses and we’ve all experienced the debilitating and sometimes deadly results of real life viruses like influenza. We go to great lengths to avoid both.

In our personal lives, we get vaccinations, eat healthy diets and exercise (most of us), shun contact with infected people, and generally avoid living the type of high risk life style that leads to illness.

Where our electronic systems are concerned, we’ve learned to practice “safe computing.” We back up our data regularly; avoid logging onto’ questionable bulletin boards and networks, or sharing data sources (USB, CD-ROM, etc) of unknown origin. We also run anti-virus programs on a routine basis.

It doesn’t require a huge leap of logic to ask, “Why don’t we practice safe computing where power viruses are concerned?”

They have the same potential effect where our systems are concerned. They enter unseen. They can cause damage ranging from annoying to catastrophic. And like most other viruses, prevention is possible if you understand the basics.

There are five simple devices you can use to prevent the equivalent of electronic influenza. All five are required for complete immunity.

THE MAGIC PILL

If there’s a magic pill to prevent power viruses, it’s understanding that prevention must be practiced as a “system.” What that means is that certain prevention techniques must be used together.

Voltage spikes are addressed with a surge diverter and electrical noise with a noise filter. Each of these by themselves, however, is capable only of weakening or slowing down a virus not eliminating it.

Isolation transformers eliminate common mode voltage problems. When surge diverters and noise filters are added to the isolation transformer, the resulting “system” kills all three viruses.

Uninterruptible power supplies eliminate blackouts, but in spite of many manufacturer claims, most aren’t capable of preventing other viruses. Once again, the UPS must be used with the other parts of the system to achieve total virus immunity.

The backdoor disturbance can be addressed several ways. Fiber optic connections are one means of electrically closing the back door, but if ordinary copper wiring is used for communication lines, it may be necessary to employ special surge diversion techniques for these connections.

Luckily the voltage regulation virus is no longer a serious hazard. Once upon a time, this virus was responsible for many system failures. However, most of today’s systems use switch mode power supplies. This technology was designed as a way of reducing both power supply size and cost while simultaneously increasing electrical efficiency.

To achieve these goals, switch mode supplies are designed to consume electrical power differently than their predecessors. These operational differences have created a beneficial by-product where voltage regulation is concerned.

As a result, most systems enjoy substantial immunity to the voltage regulation virus. Additional preventative measures (voltage regulators, etc.) are unnecessary.

CONCLUSION

Power viruses are an appropriate description of the power quality problems that can plague electronic systems. Like other viruses, they are invisible - often announcing their presence only after some initial damage has already been done. Their effects can be a minor annoyance like a lockup or system error or they can be catastrophic like a blown up integrated circuit or power supply failure.

Our dependence on sophisticated technology has created an increased awareness regarding the need to safeguard system integrity. Software viruses have led to the introduction of “antivirus” programs and system data is routinely backed up to prevent loss.

Part of this “safe computing” lifestyle should be the prevention of power viruses, too. This is possible only when prevention is systematic.

Voltage spikes, electrical noise, and common mode voltage is eliminated by a package that contains an isolation transformer, surge diverter, and noise filter. UPS and data line protection can be added to the system as applications demand.

Also see Protection Or Quality? With AC Power For Systems, It’s Vital To Know The Difference.

Dennis Ver Mulm works with POWERVAR, based in Lake Forest, Illinois.

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Posted by Keith Clark on 07/13 at 12:51 PM
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Joe Tessone Joins Sensaphonics As Audio Engineering Consultant, a.k.a., “Sound Guy”

Joe Tessone has joined Sensaphonics as the company’s new audio engineering consultant, a.k.a., the new “Sensaphonics Sound Guy,” where he will advise customers on the implementation of in-ear monitoring, handling questions ranging from the basic to the highly technical, helping ensure a positive IEM experience.

Joe Tessone, 27, is a 2007 graduate of the Audio Arts and Acoustics program at Columbia College Chicago, with specialties in Audio Design and Production.

In addition to his duties at Sensaphonics, he owns a commercial studio, Mystery Street Recording Company, and is the Audio Archivist at the Old Town School of Folk Music in Chicago.

He is also a performing musician, most recently as singer/songwriter for the eclectic punk band Behold!, and as banjo/ukulele player for an Americana music group, the Rust Belt Ramblers.

Tessone is also passionate about occupational health risks and workers’ rights, which is a perfect fit with Sensaphonics’ focus on hearing conservation as an industry-wide issue among both engineers and musicians.

“I’m a big believer in healthy hearing. Our ears are the most important tools we have. Just as it would be crazy for a welder to work without eye protection, it is just as dangerous for us in the music industry to work without protecting our ears,” he says. “I own two pair of custom Musicians Earplugs, the ER-15, and always have one with me.”

Sensaphonics president Michael Santucci states, “Having a sound engineer on staff is a big help to our customers, and Joe Tessone really stood out as the obvious choice. Being a performing musician with strong engineering skills and customer focus means that Joe has the breadth of knowledge and perspective required for the role of Sound Guy.”

Tessone views his new job as a natural next step in his personal development within the music industry. “Working with the industry leader in both in-ear technology and hearing conservation is a great opportunity,” he says. “One thing I learned at Columbia College is that all of us in the audio industry really need to support each other. Being the Sound Guy at Sensaphonics will allow me to share knowledge and solve problems for other musicians and engineers. I’m very happy to be here.”

Tessone is available to answer IEM questions via phone and email, /To contact him, go to www.sensaphonics.com and click on “Ask the Sound Guy.”

Sensaphonics

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Posted by Keith Clark on 07/13 at 09:35 AM
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