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Wednesday, February 08, 2012

Tannoy Unveils VLS Series Passive Column Array Loudspeakers

Tannoy has introduced the VLS Series passive column array loudspeakers offering a balance of performance and cost, when active beam-steering may neither be required nor affordable.

The VLS Series is the first Tannoy product to incorporate FAST (Focused Asymmetrical Shaping Technology), which delivers unique acoustic performance benefits. Central to this is its asymmetrical vertical dispersion, gently shaping the acoustic coverage towards the lower quadrant of the vertical axis. By the nature of a typical application, an “ideal” column loudspeaker should be biased in the vertical plane, toward the audience and away from reflective surfaces above (like ceilings) which are detrimental to intelligibility.

FAST also facilitates quicker, easier installation with less need for tilting or specific concern for optimal mounting height. Mounting is handled via supplied wall brackets.

Tannoy has packaged this performance in a slender and narrow profile, aesthetically refined, powder-coated aluminum chassis with curvilinear aluminum grille. Each model is available in either black or white as standard, with custom RAL finishes available at additional cost and lead-time.

Three models are available – VLS 7 (7 × 3.5-inch LF) designed for speech-only applications, VLS 15 (7 × 3.5-inch LF with 8 × 1-inch HF) and VLS 30 (14 × 3.5-inch LF and 16 × 1-inch HF), both of which are designed for more demanding full-range applications as well as speech.

All are IP64 rated for dust and water ingress and are salt spray and UV resistant as well as subject to rigorous high/low operational temperature and humidity testing.

Specification is aided by the addition of an exclusive Tannoy edition of EASE Focus v2.0 software, allowing systems to be designed with predictable results, along with the ability to specify VLS Series in conjunction with Tannoy’s existing column loudspeakers – including I Series and QFlex.

Tannoy

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Engineer/Producer Matthew Noble Utilizes Metric Halo ChannelStrip On Recent Projects

For more than three decades, Matthew Noble has been at the forefront of pop music as a session guitarist, programmer, songwriter, engineer, and producer, with an engineering client list that includes Rihanna, Diana King, Southside Johnny, and Rod Stewart, among many.

These days, he performs most of his work out of the Loft Studios in Bronxville, NY and in the newly renovated Riverworks Recording in Dobbs Ferry, NY. Recent work with the musical Big River and gospel artist Rell Holland & Experience have put Noble’s new favorite plug-in, the Metric Halo ChannelStrip, through its paces.

“I tried Metric Halo’s ChannelStrip because some other people that I respect were using it,” explains Noble. “My friend Keith Brown, who is a well-known Nashville songwriter, was working on a project with Billie Decker, who is one of the hottest mix engineers in country music. Keith’s enthusiasm for the plug-in, together with his revelation that Billie uses it ‘all over the place,’ was enough to motivate me to check it out.”

Riverworks Recording boasts a huge, luscious acoustical space, which has changed the way both Noble and the producers and artists he works with approach the recording process.

“So much of my work there has involved tracking live instruments, as opposed to the ‘virtual players’ that live inside our modern computers,” he says. “While it’s been a refreshing change, it has also brought with it challenges. For example, getting a great drum sound and a great overall mix with the new expectations for how long things take these days is not easy.

“ChannelStrip has been very helpful because all the functions that I need to access quickly are all in one plug-in. These include the less ‘sexy’ functions, such as phase reverse and multiple trims, in addition to full-blown and flexible dynamics and equalization. Having everything in one plug-in has greatly improved my workflow.”

Noble often puts Metric Halo’s well-crafted presets to use: “The ChannelStrip presets are a great starting point. They’re especially useful in a time crunch, when the client is breathing down your neck. The acoustic guitar and drum presets are often spot on, right out of the gate. When I tweak, the informative GUI lets me know exactly what I’m doing.”

Of course, the best GUI in the world is useless if the algorithms behind it don’t cut the mustard. It’s here that Noble finds it really shines. “ChannelStrip has a great sound,” he said. “Like an SSL, it can be very aggressive and not at all subtle. Despite all its flexibility and sonic muscle, it has remarkably low CPU drain, which means I can use it whenever I need it.”

Metric Halo

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Tuesday, February 07, 2012

In The Studio: Three Mid-Side Processing Tricks (Includes Audio Samples)

A form of processing on stereo sources for practical or creative effects

 
In this article I’ll explain how I use mid-side (MS) processing on stereo sources for practical or creative effects.

Mid-Side?
Two channels of audio can be combined in a way that gives us control over what is the same in each signal, the middle, and what is different, the sides.

The middle is where the kick drum, snare, bass, vocals and a lot of other instruments are, the sides have any hard-panned instruments and spatial effects like reverb.

It can be pretty interesting to listen to music like this, there can be a lot hidden in the side channel.

MS is also a stereo microphone technique using a cardioid microphone facing the source and a bidirectional mic turned 90 degrees away just picking up ambience.

In this situation the two signals would need to be decoded into stereo. The side mic signal is duplicated, polarity inverted and the two side signals are then panned hard left and right. This is not a true stereo mic technique but can sound very nice. The balance of mid and side signals can be adjusted as needed by changing the level of the three tracks.

You can manually encode and decode stereo files to MS and use mono plugins to process mid or side individually. A lot more plugins have an MS mode now. Many of the modules in the T-Racks suite allow mid side processing, as does Ozone, a few compressors and equalizers and a distortion also come to mind.

You can do this for subtle or crazy effects, its a fun way to experiment with plugins and get some unique sounds.

Loud & Wide
For a recent mastering job I used a Fairchild compressor plugin in MS mode (Lat/Vert) to compress the middle and increase the level of the sides. I did this in parallel so I could blend the effect in easily. I was also using this to get a lot of extra loudness. You can call this parallel MS compression.

Compare -

The master without parallel MS compression: listen

With parallel MS compression: listen

With parallel compression soloed: listen

No More Messy Verb
Someone asked ma about clearing up the middle of a mix when using a lot of reverb. Using MS compression on the reverb return can work well. Compress the middle more than the sides and increase the side volume if you want more width.

Here is an example of that on some drums - Steven Slate playing in KONTAKT. The whole kit is sent into Valhalla Room. With the Fairchild after the reverb I’m lowering the middle by 2 dB and raising the sides by 2 dB.

Here is this effect with lots of reverb on the drums: listen

And now with MS compression on just the reverb bus: listen

There is NO compression on the drums themselves, I’m only compressing the reverb return and widening it.

Wacky Effects
Here is an example of what you can do with a stereo loop and any plug-in. This is a little more complicated, and only works if there are hard panned sounds.

The loop I started with had a hi-hat that wasn’t panned very hard -  I copied it to a new track, filtered out all the lows, boosted some highs and then panned it hard left. Then I recorded the combined original and panned track to a new file.

Here is what I’m starting with: listen

Now that I had something on the sides I could mess around with MS processing.

The first thing you have to do is convert left-right to mid and side. I use the free +matrix MS decoder from SoundHack.com. After that I used a delay plugin to add some filtered echoes just to the middle by disabling the right side input.

In the next insert I used a distortion on just the right side. This brought out a lot more of the reverb than was heard in the original loop. Lastly, second MS decoder was used to bring it back to stereo.

Here is how the loop sounds now with delay in the middle and distortion on the sides: listen

Pretty cool right!? I hope you have found these tricks useful.

Jon Tidey is a Producer/Engineer who runs his own studio, EPIC Sounds, and enjoys writing about audio on his blog AudioGeekZine.com. To comment or ask questions about this article go here.

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Fishman Unveils Triple Play Wireless Guitar Controller

The new Fishman Triple Play Wireless Guitar Controller combines guitar with any virtual instrument or hardware synthesizer to access a wide range of instruments, samples and sounds on stage to expand the depth and impact of live performances.

Triple Play comes with a wireless controller, hexaphonic pickup, and wireless USB receiver. The controller and included software works with industry standard DAWs and vitual instruments and installs quickly on any electric guitar. The system can be easily removed from the guitar because it doesn’t require any permanent installation.

The Triple Play system features several “hold” functions such as sustain, looping, and arpeggiators, along with string or fret splits for multiple instruments.

Also included are menu navigation controls for the included software and a guitar synthesizer volume control. A guitar, mix, synth switch is easily accessible during performances.

A low profile design (less than .5-inch) allows the controller to be left on the guitar and still fit in the case. It operates with a rechargeable Lithium Ion battery (included).

Triple Play’s powered USB wireless receiver interfaces with computers or iOS devices. The system comes with a comprehensive Windows, OSX and iOS software bundle to get users started.

A Triple Play Wireless Guitar Expander option provides additional connectivity for interfacing wireless MIDI signals to computers or iOS devices. It adds a full function USB audio interface with guitar input, bypass and headphone output, MIDI hardware IN and OUT and support for footswitches to extend Triple Play’s capabilities for recording, performing or composing music.

The new Triple Play Wireless Guitar Controller is scheduled for release in June 2012.

Fishman

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Monday, February 06, 2012

Radial Introduces Shuttle Multi-Function Effects Insert Module For 500 Series

Radial Engineering has introduced the Shuttle, a new multi-function effects insert module for the 500 Series frame format and the Radial Workhorse.

The Shuttle offers three insert loops:

—Loop-1 is a front-panel insert that employs 1/4-inch TRS connectors for fully balanced connectivity
—Insert-2 is an unbalanced insert that is also front panel mounted that easily interfaces to standard effects devices
—Insert-3 is available on the Workhorse using the Omniport, which is wired following convention with tip-send, ring-return, making it ideal to interface with a remote patchbay.

All three loops are equipped with an insert switch that lets the user compare the wet and dry signal paths.

The insert points may also be used as inputs to feed a signal into the Workhorse mix bus. This opens the door to using the Workhorse with source devices such as CD players and iPods or with multi-channel fader packs and so on.

The Shuttle also enables those who own a Workhorse to easily integrate older 500 Series modules into the Workhorse mix buss. One mounts the non-Radial module next to the Shuttle, engages the feed function, and the signal will automatically be routed.

“As soon as our engineers started to integrate the Workhorse within the digital studio environment, they immediately noticed a need to simplify the process of patching effects in and out following what studios would normally do using a patch bay,” says Radial sales manager Steve McKay. “And as we delved further down the rabbit hole, we realized that the 500 Series was limited with respect to performing functions such as overdubbing. The Shuttle addresses these limitations while opening the door to creative new patching options.”

Radial Engineering

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Church Sound: How To EQ Speech For Maximum Intelligibility

The problem is unclear words are a distraction from the message

 
Haven’t we all had stories of misheard words? It could have been a song lyric or you misheard your spouse?  Maybe they mumbled a word or it just wasn’t clear what was said. This has been the cause for a few hilarious moments at our dinner table. 

The problem is unclear words are a distraction from the message. 

In the church environment, the pastor’s words must be clear. We can ensure this maximum intelligibility through proper speech EQ.

There are four topics to consider when it comes to the EQ’ing needs for the spoken word. 

1. Microphone location. We are fortunate in that most pastors now use wireless microphones. This means that the distance between the mic and their mouth is pretty consistent.  In the case of the headpiece, this is especially true.

In the case of the lapel mic, remember they should drop their chin to their chest and put the mic directly below that point. Long ago, I was taught “a fist away from the chin.” The point here is that we want the best sound isolation we can possibly get while having a good gain structure in place. 

Remember, the closer to the source, the more the proximity effect comes into the equation and you’ll need to EQ out some of that added bassiness.

2. The speaker’s natural voice. Just as every guitar has a unique sound, so does every person. You want to bring out the best qualities of their voice. You don’t want them to sound like a different person. Their vocal characteristics are also “what you have to work with.” 

This means you’ll need to know how to deal with quiet speakers, bassy talkers, and nasally preachers, just to list a few.  Not everyone has a great radio voice.

3. Presence of background music. Depending on your church, your pastor might talk with a running soundtrack. There is definitely an art to being able to play the right music for this. 

However, any type of music bed means you now have to make a space for the voice amidst the instrumentals. Instrumentals can easily blur the spoken word so you’ll have to plan on tweaking the EQ for the musicians as well. 

4. The environment. Just because a vocal boost at 400 Hz sounds good in one room doesn’t mean it will sound good in another room. One of myreaders runs audio outside…in Egypt. Any EQ work must take the environment into account. The settings for a “quiet room” won’t be the same for an echo-y room or an outdoor venue.

Now that we’ve got those out of the way, let’s turn to…

The Frequency Make-Up Of Speech

Our speaking voice has three frequency ranges that need to be understood:
1. Fundamentals. The fundamental frequencies of speech occur roughly between 85 Hz and 250 Hz.
2. Vowels. Vowels sounds contain the maximum energy and power of the voice, occurring between 350 Hz and 2 kHz.
3. Consonants. Consonants occur between 1.5 kHz and 4 kHz. They contain little energy but are essential to intelligibility.

In short, this means that the “power” of the voice does not equate to the intelligibility of the voice. Think of it like this…just because a person has a booming voice doesn’t mean they are easy to understand. 

Now that you understand the audio dynamics (fundamentals, etc) in a voice and the environmental concerns (background music), let’s turn to…

What You Can Do To Provide The Maximum Speech Intelligibility For Your Pastor

There are three things you can do for tackling the EQ’ing process for the spoken word:

1. Make room for the voice. As I mentioned above, the environment makes a difference in how you EQ the spoken word. We can only control what is coming into the mixing board, so wind and rain aside, let’s talk about music. 

Mixing a large band means making space in the sonic spectrum where each instrument/vocal can sit and sound unique; and of course then blending these sounds together into a tight mix.

The spoken word needs the same treatment when music is played underneath it. This can happen in two ways—

—A. Adjust volume. This can be done using compression or simple volume adjustments. The general rule-of-thumb is the music is there to support the spoken word – to sit underneath it.  Therefore, look to cut volume levels of instruments before you boost the volume of the speaker. You can also use compression to bring volume levels up and down as you wish.
—B. Adjust the mix. Cut the frequencies of the instruments where they are the same as that of the speaker. Boost the spoken word EQ in those areas a little if needed to present the music and the voice as two distinct sounds.

2. Know sibilance and how to avoid it. Sssssssibilance in vocals is when the sound of the letter “S” sounds more like a hissing snake. You can accentuate vowel sounds/add presence by increasing the EQ in the 4.5 kHz to 6 kHz. 

However, the “S” sound lives between 5 kHz and 7 kHz. Therefore, be careful when adding presence because you can easily go from a great sound to a hissy sound.

3. Focus on vocal quality. There is no simple 1-2-3 process to EQ’ing the spoken word. Therefore, take these points into consideration:

—Roll off the low frequencies if the proximity effect is causing unusual bassiness.
—Don’t roll off so much low end as the voice loses some of its umph. Yes, I’m using “umph” as a technical word.
—Boost in the 1 kHz to 5 kHz range for improving intelligibility and clarity.
—Boost in the 3 kHz to 6 kHz range to add brightness. This can help with speakers with poor intonation.
—Boost in the 4.5 kHz to 6 kHz range to add presence. Note that too much boosting in this area can produce a thin lifeless sound.
—Boost in the 100 Hz to 250 Hz for a boomy effect.

In Case Your Head Is About To Explode From Information Overload, Remember:

—The above points can contradict each other. There is no hard and fast rule. Mixing is as much an art as a science. Trust your ears over everything else.
—It’s possible that once you EQ the vocal channel that it’s a little lacking in the low end. Boost it a bit give it that full sound. Again, trust your ears. Close your eyes and ask yourself if it a) sounds natural and b) sounds clear.

Finally

EQ’ing the spoken word is about improving the quality of the sound so it sounds clear, is easy to understand, and sounds natural.

So much of our mix time goes towards the band. Make sure you spend those few crucial minutes working on the pastor’s vocal as well. 

Church was about the sermon long before music, skits, and cool videos rolled onto the scene.

Ready to learn and laugh? Chris Huff writes about the world of church audio at Behind The Mixer. He covers everything from audio fundamentals to dealing with musicians. He can even tell you the signs the sound guy is having a mental breakdown.

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Lexicon Offering Individual Plug-Ins From PCM Native Effects & PCM Native Reverb Bundles

Lexicon has announced the availability of the individual plug-ins from its PCM Native Effects and PCM Native Reverb Bundles, with a total of 14 plug-ins available, including Pitch Shift, MultiVoice Pitch, Chorus, Resonant Chords, Random Delay, Dual Delay, Stringbox, Vintage Plate, Plate, Hall, Room, Random Hall, Concert Hall and Chamber.

“Offering the individual plug-ins from our PCM Native Effects and PCM Native Reverb Bundles represents our commitment to provide Lexicon users with greater flexibility and ease to obtain exactly the sound quality they are looking for from the specific plug-in(s) they need for any project,” says Rob Urry, vice president Harman Professional Division & GM of Signal Processing and Amplifier Business Units.

The PC- and Macintosh-compatible plug-ins are designed to work with popular DAWs like Pro Tools, Logic and Nuendo, as well as with any other VST, Audio Unit or RTAS-compatible host.

Each plug-in can be run in mono, stereo or mono in/stereo out, and on-screen input and output meters are provided for precise level setting.

All Lexicon plug-ins are Native only, and require iLok2 authorization. The individual plug-ins will be available in February 2012.

Lexicon
Harman Pro

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Friday, February 03, 2012

Peavey Debuts Expansion Modules For Powered Loudspeakers

New Peavey Expansion Modules expand the capabilities of the company’s EU Series, Impulse 12D, and the new PVXp powered loudspeakers.

The new 9-Band Graphic EQ Expansion Module aids users in controlling feedback using the built-in, proprietary FLS Feedback Location System, which makes accurate feedback elimination simple and intuitive.

The 9-Band module gives users greater control over shaping and accenting live or recorded vocals and live instruments, as well as increases control over general sound equalization, and removes 60 Hz AC hum.

Meanwhile, the new 3-Band Parametric EQ Expansion Module provides fine-tuning capabilities with gain, frequency and bandwidth (Q) control.

Finally, the new 3-Channel Mixer Expansion Module expands the powered enclosure’s input options by three input channels, with each channel featuring level control, high and low equalization control, and a combination XLR and 1/4-inch input.

9-Band Graphic EQ Expansion Module specifics:
—Nine frequency bands: 63, 125, 250, 500, 2k, 4k, 8k, 16kHz
—FLS Feedback Locating System
—+/- 15 dB boost and cut
—U.S. MSRP $59.99

3-Band Parametric EQ Expansion Module specifics
—Three frequency bands: Low 40 to 800Hz; Mid 200 to 4kHz; High 1kHz to 20kHz
—+/- 12 dB boost and cut on each frequency band
—Q adjustable from 0.1 to 1 on each frequency band
—Hard-wire bypass
—U.S. MSRP $49.99

3-Channel Mixer Expansion Module specifics
—Three channels of additional inputs
—Combination XLR female and ¼” TRS input
—Microphone and line-level input range
—Two-band EQ on each channel
—Switchable phantom power global for all three channels
—U.S. MSRP $79.99

The new Peavey Expansion Modules for powered loudspeakers are made in the USA will be available in Q2 2012 from authorized retailers.

Peavey

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Biamp Systems Promotes Key Executives With Goal Of Bolstering Growth

Biamp Systems is further developing its executive management structure through the creation of three new key positions to provide new focus on product and services innovation; global marketing and sales strategies; organizational scalability; and even more effective service to its customers.

In support of these goals, Biamp has made the following executive appointments: Graeme Harrison has been appointed to the position of executive vice president of marketing; Matt Czyzewski has been named the executive vice president of operations; and Ron Camden has been named the vice president of worldwide sales.

“Our vision for the future requires an alignment of our organization with our goals for bringing fresh thinking and innovative products and services to our customers,” states Steve Metzger, president and CEO, Biamp Systems. “I’m very excited about the changes we’re making. Graeme, Matt, and Ron all have a wealth of experience and are extremely talented people, and their promotions will have far reaching and positive consequences for our company and our customers.”

In his new role as the executive vice president of marketing, Graeme Harrison will oversee the Biamp worldwide sales, marketing communications, and product management groups. Harrison has worked for Biamp Systems for 20 years and first started as the company’s regional manager in Europe serving Europe, Middle East, Africa and India. He then transitioned to international sales manager and most recently to vice president of international sales.

Matt Czyzewski has been with Biamp for 15 years and has more than 25 years of industry experience. Czyzewski will assume the role of executive vice president of operations, moving from his previous position of vice president of business development at Biamp. Prior to his last position, Czyzewski was the vice president of engineering. His new position will oversee the technical operations at Biamp.

Ron Camden becomes the new vice president of worldwide sales. Camden has more than 25 years of experience working in AV technologies and is passionate about sharing innovations and identifying trends. For the past 17 years, he has been the vice president of North American Sales. In his new position, Camden is charged with developing global sales strategies, and leading the worldwide sales team.

Biamp Systems

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Wednesday, February 01, 2012

Oslo Audio First Rental Company In Norway To Invest In L-Acoustics KARA

Oslo Audio has become the first rental company in Norway to invest in L-Acoustics KARA WST line source loudspeakers, ordering 24 cabinets from local distributor Scandec Systemer.

The KARA loudspeakers are powered by L-Acoustics LA-RAK amplified controllers and supplemented with SB18 subwoofers.

“KARA has a flexibility that will enable us to use the system for pretty much all venues in Norway,” says Paal Klaastad of Oslo Audio. “For us the audio performance of the KARA system was never in question. As a long-time L-Acoustics user, we are confident that the sound quality is first class. The reputation of the brand ensures that the end users are also confident of the system’s performance.

“The scalability of the system, its integration with the LA-RAK platform and the ease of rigging and handling makes us confident that this will provide a good return on investment for years to come. We look forward to putting the system to use, and to collaborating with other network agents in Scandinavia.”

Oslo Audio’s new KARA loudspeakers were used for the first time at the 10-year anniversary concert of Crystal Canyon Studios, with a lineup of Kåre & The Cavemen, Ulver, Paperboys, Kitchie Kitchie Ki Me O and André Holstad.

That system consisted of the 24 KARA cabinets with 12 SB118 subs and six 115XT HiQ coaxial monitors, powered by LA-RAKs.

L-Acoustics

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Monday, January 30, 2012

Summit Ridge Christian Fellowship Mobilizes With Powersoft Amplifiers

Summit Ridge Christian Fellowship located in Spokane, Washington, recently upgraded their worship services with a new sound system powered by Powersoft amplifiers.

The church leadership worked closely with AGI Professional, located in Eugene, Oregon, to design a system that would accommodate not only their contemporary services but also their requirement for the system to be portable with easy set-up.

“Summit Ridge is part of a growing segment of churches that doesn’t own a brick and mortar building but instead leases space for their regular services,” explains Kyle Anderson, owner of AGI Productions. “They currently use another church space that doesn’t have a big enough sound system to accommodate their needs – which is where we came in.”

Anderson specified a left-right system that consists of two Fulcrum Audio FA12 12” coaxial loudspeakers pole-mounted on 13 ft. heavy-duty truss stand with a steel crank-up. Two Fulcrum Audio Sub115 15” direct radiating subwoofers are positioned directly below the loudspeakers for driving low end.

A single four-channel Powersoft M50Q DSP+ETH provides all of the power necessary.

“With each channel providing 750W at 8 ohms, it was a no-brainer,” Anderson adds. “We used channels 1 and 3 for the subs and 2 and 4 on the mid/high boxes and had plenty of headroom.”
Anderson continues, “All we had to do was synch the DSP with the manufacturer’s FIR filters – do a little tuning here and there and we were good to go. We locked it down and provided the crew with a “show EQ” in their portable racks that they use if they run into a situation that requires minor modifications.”

The 4-channel M50Q DSP+ETH is equipped with four inputs and outputs all in one rack unit. Each channel offers five bi-quad filters for system equalization, two crossovers, eight bi-quad filters, and RMS and peak limiters eliminating the need for outboard equalizers. Presets can be created, stored, and completely monitored utilizing the Armonía Pro Audio SuiteTM software with computer and amplifiers communicating over standard Ethernet.

“Our clients were amazed that they didn’t need a huge amp rack to get the power the system provides,” Anderson concludes. “They were so delighted that they have spread the word to some other churches in the area that were looking for upgraded systems as well. It was a win-win for everyone involved.”

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Friday, January 27, 2012

Everything You Wanted To Know About Sound Level Meters (SLMs)

The primer: what, how, why, what's available, techniques, applications and more

A sound level meter (SLM) is a device used to make frequency-weighted sound pressure level measurements displayed in dB-SPL.

0.0 dB-SPL is the threshold of hearing, and is equal to 20uPa (microPascals). This correlates to what one would aurally perceive when in a deep cave or in a large anechoic chamber.

Packaged as a single-function handheld test device, SLMs are intended to be held at arm’s length during measurements (to reduce the effects of the body on the measurements) or secured to a tripod stand for more stability.

All SLMs feature an omnidirectional measurement quality condenser microphone, a mic preamp, frequency weighting networks, an RMS detector circuit, averaging circuits, the meter display, AC and DC outputs used to feed other measurement devices or for recording (see Figure 1, below).

Most SLMs have the same set of user adjustments, including SPL range selection, A and C weighting filters, slow and fast detector response, and minimum or maximum SPL.

The SPL range switch provides a balance between minimizing the preamp noise level and measuring a wide range of sound pressure levels.

Most of the commonly available SLMs measure from about 30 to 130 dB-SPL and do this in 3-4 ranges.

 
The more advanced and expensive SLMs feature removable microphones, 1-octave and/or 1/3-octave filter sets, additional weighting filters including B, D and Flat or Linear (no filter), additional detector response options (Impulse and Peak), averaging (over time) and data logging or storage (either on-board, as computer files or both).

Almost all SLMs are designed and specified to perform to one of four internationally standardized levels of accuracy:

 
Note that these tolerances are at 1 kHz, the standard calibration frequency for SPL measurement. In order to ensure the flatness of the SLM there are additional tolerances specified for various frequency bands and microphone classes as well.

Class-0 SLMs are employed primarily to calibrate other SLMs and may be used for very high precision noise measurement in controlled spaces and/or for academic research.

Class-1 and Class-2 SLMs are most widely used by acousticians, sound system professionals, industrial designers/ manufacturers and researchers in academia and government. Measurements made with these levels of accuracy are generally acceptable as evidence in the resolution of legal disputes.

Class-3 SLMs are restricted to noise survey meters and dosimeters.

Microphone Sizes
Most general purpose SLMs are provided with a 1/2-inch free field microphone, permanently attached to the SLM body.

Higher quality SLMs with removable microphones may be outfitted with capsule sizes ranging from 1/8-in to 1-in.

The smaller microphone capsules (1/8-inch and 1/4-inch) have three primary advantages:
—They are capable of higher SPL measurements;
—They remain omnidirectional up to higher frequencies;
—And they present less disturbance of the air due to their reduced physical size (the mic itself has less impact on the measurement).

Larger 1-inch capsules exhibit less self-noise and are therefore best suited to measurement in very quiet spaces, of very low-level noise sources, and of low-frequency energy.

Microphone Classes
One can also choose between three different classes of measurement microphones: Random Incidence, Free Field, and Pressure.

In reality, these three classes of microphone are all fundamentally “pressure response” microphones and are designed to sense pressure - not pressure gradients or particle velocity.

The difference is primarily in how they pick up sound at high frequencies, and this relates to how they are positioned relative to the sound source.

At higher frequencies (where wavelengths become so small as to equal, or be smaller than, the size of the microphone diaphragm) the direction in which the microphone is aimed - relative to the direction of the sound being measured - will have an effect on the accuracy of the measurement.

Random Incidence microphones are intended for use in a diffuse sound field where the sound is arriving from all directions due to a high level of reflections.

This type of microphone must be angled approximately 70 degrees to the sound source in order to yield accurate measurements.

Free Field microphones are designed for use in an open space (free of reflections) and should be held at 0 degrees to (pointed directly at) the sound source.

Pressure microphones are designed specifically for measurements in couplers and closed cavities such as plane wave tubes, ear cavities in dummy heads and for noise measurements in air ducts.

Some SLMs provide a switched correction filter allowing one type of microphone to be used in place of the other.

Note: In general, a “measurement quality” microphone differs from a recording or sound reinforcement microphone in its flat response, the absence of an onboard preamp, and its less rugged construction.

Detector Averaging Time
The standard averaging time choices provided by the majority of SLMs are Fast and Slow.

The Fast setting has a 125milliseconds (mS) averaging time (developed in 1950 to match the ballistics of the moving coil meters used at that time) and is useful for steady state (not dynamic) sound.

For measurement of more dynamic sound (when the Fast setting results in fluctuations of 4 dB or more), the Slow setting provides a 1 second averaging time resulting in much more consistent and stable readings.

It is this setting (along with an A-weighting curve) that is most frequently used for measurement of pop music SPLs.

Some higher priced SLMs also provide Impulse response time (four times faster than the Fast setting) used for extremely dynamic and short term (impulse) sound and peak response time, which accurately responds to signal pulses as short as 50 microseconds.

Frequency Weighting Filters
Because we humans do not hear frequencies in a linear manner (as illustrated in the often referred to Fletcher-Munson curves, aka: equal loudness contours), sound level measurements made with a flat response do not accurately reflect how we perceive sound.

When measuring pop music sound reinforcement systems (with the program signal containing generous amounts of bass energy), broadband readings would be artificially high without the frequency shaping provided by the A-weighting filter.

C-weighting, with its more gradual low-and high shelving filters, is used for very high SPL sound system and noise measurements.

On SLMs without a Flat setting, the C-weighted setting may be used to feed a spectrum analyzer such as Rational Acoustics Smaart for frequency analysis of the broadband sound.

Table 2 below, and then Figure 2 below that, show the frequency response for A and C weighting filters, plus two other weighting filters employed for more exacting and specific noise measurements.

 
All weighting curves are equal in level at 1,000 Hz, the standard reference frequency that SLMs are calibrated to using acoustic calibrators.

Note On SPL Annotation: When communicating or recording (speaking or writing) SPL values, it is necessary to annotate what - if any - weighting curve has been employed. A-weighted SPL is written or stated as “XX.X dBA” and C-weighted measurements as “XX.X dBC”. Such annotation eliminates the need to provide the “SPL” suffix.

The absence of weighting notation implies that a Flat setting (no weighting filter) was used and in this case the suffix “SPL” must be used to differentiate from other, non-acoustic decibel values.

Historical Note: The A, C and B frequency weighting curves were first developed for specific sound levels based on equal-loudness contours.

Because equal-loudness contours were developed from research utilizing pure tones, it later became evident that the weighting curves chosen for SLM measurements at certain levels do not correlate very well to how we actually hear complex sounds.

 
More recently it has been found that the A-weighting curve provides a realistic approximation of how we hear over a fairly wide range of sound level measurements.

Display
With very few exceptions, modern SLM’s are provided with LCD displays with 3- or 4-digit readouts. Most of these provide 1/10-dB resolution except in the case of 3-digit displays when they go over 100 dB-SPL.

Prior to LCD technology, analog meters were employed and these suffered from all the inherent shortcomings of this method of display: slow rise and settle time, susceptibility to damage from pinging (over loading) and overall mechanical fragility.

Another shortcoming of analog meters is the difficulty in establishing stable readings due to the range selecting that is common in most SLMs. One must take the range that is selected and then add or subtract from the meter position accordingly.

Measurements made when the sound pressure level has changed and is out of range are prone to be inaccurate, as well. Some feel that fast impulse noise measurements are easier to read with an analog SLM than with a “digital” SLM.

Both styles of display are difficult to read in dim lighting but some SLM’s feature backlit LCD’s for this condition. And either type of display may be quite difficult to read while at arm’s length if the display size is too small.

Equivalent Continuous Sound Level
Although we live sound folk generally believe that we are able to accurately determine the overall sound pressure level when we view the displayed real time values on our SLM’s, these readings are actually no more than a glimpse, or snapshot, of the varying sound pressure levels that we are exposed to.

Due to the dynamic content of the sounds we normally work with (this applies to both speech and music) and our own human shortcomings, we are unable to mentally integrate the frequently changing SPL’s and determine an accurate mean SPL value.

Research into hearing safety and annoyance and the standards that have been developed from this research are based on the accumulative exposure to sound energy over periods of time.

To this end, integrating SLMs have been developed to provide measurements that are conducted over a selected time period and averaged so that they provide an accurate equivalent energy value, designated as Leq (LAeq or LCeq when weighting is employed).

Integrating SLMs also provide Lmax and Lmin values, and the user can choose the time period that the Leq averaging takes place (normally from several seconds to 24 hours).

More and more organizations and governments are adopting this as a more accurate means for establishing risk of hearing damage, community noise standards, annoyance issues, etc.

Although stand-alone integrating SLMs are considerably more expensive than basic SLM’s, some software is available to provide this statistical averaging from the AC output of any SLM.  and both the TerraSonde Audio Toolbox and Neutrik Minilyzer provide this function along with other SPL measurements and other useful audio system measurements.

Relative Vs Absolute Measurement
Absolute measurements are those made with calibrated SLMs that provide actual sound pressure levels referenced to 20uPa (0 dB-SPL).

Such absolute SPL measurements may be used with and accurately compared to those provided by other calibrated and equal quality SLMs.

Conversely, measurements made with un-calibrated SLMs or those with of low quality (including less then linear frequency response) do not accurately indicate actual sound pressure levels.

SLMs that have been incorrectly user-calibrated or have simply lost their calibration over time also do not provide accurate absolute SPL values.

Relative measurements are simply comparisons between sound pressure levels and do not require absolute calibration nor other “precision” performance characteristics such as a flat frequency response.

Non-calibrated and lower-quality SLMs do provide meaningful comparative measurements as long as the same SLM is employed for these measurements and the measurements are made in the same manner.

Example: A Radio Shack SLM, which is specified as providing +/-2.0 dB-SPL accuracy and which has a less-than linear frequency response, is still able to provide realistic indication of the changes in relative sound pressure levels even though it does not provide accurate measurement of absolute sound pressure levels.

Likewise, the SLM function contained in Smaart may be used “as is” (without calibration) for accurate relative SPL measurement.

One relevant and valid relative SPL measurement that can be made in sound reinforcement work is determining the even-ness of coverage provided by a loudspeaker system.

In this case virtually any SLM is able to indicate any deviations that may exist in the sound pressure levels as the SLM is moved about the coverage area of the loudspeaker system.

In a similar manner we can observe changes in relative SPL as the gain of a sound system is brought up and down and we can accurately determine changes in SPL as a noise source (machinery, for example) is turned on and off.

Available SLMs
As stated previously, SPL measurements made with Class-2 and Class-3 SLM’s provide sufficiently accurate (within +/- 1.0 dB) readings for what we require in sound reinforcement work.

Until recently, SLM devices that met this level of accuracy were prohibitively expensive ($1,200 and up).

Most of the old classic SLM’s are cumbersome to set up and, due to their inherently fragile construction, are seldom used for touring sound or on-site installation work.

Without question the most common SLMs used by live sound folks have been those made by Radio Shack. These appear to represent a very good value, are easy to use and are readily available almost anywhere in North America.

In researching for this article I have found that, along with all of the sound system operators and contractors who own a Radio Shack SLM (or two), there are large numbers of other users including university physics/acoustics and biology departments, municipalities, hi-fi enthusiasts, and theater system installers and property management companies.

And there are at least a few websites devoted to using the Radio Shack SLM for tweaking hi-fi, project studio and sound reinforcement loudspeaker systems and for simple noise measurement.

The original Radio Shack model 33-2050 utilizes an analog meter display and has been in production for the past few decades, and several years ago, the model 33-2055 was introduced, providing a “digital” (alphanumeric) LCD display, recessed and better-protected switches and a maximum/minimum measurement function.

Otherwise, these two models employ the same microphone, base circuitry and battery complement.

 
Alternative SPL Measurement Tools
One alternative to the stand-alone SLM’s described in this article is the SLM function included with some of the better portable RTA’s (real time analyzers) that are often employed for live sound and acoustic measurement.

Models from Ivie, Audio Control, and GoldLine all provide accurate broadband sound level measurement (Class-2 or Class-1, depending on the microphone used) plus octave and 1/3-octave spectrum analysis.

This combination of measurements is very useful for determining both the SPL and the frequency components of the sound/noise under con sideration.

 
Most, if not all, of the current crop of computer-based electroacoustic measurement system provide an SPL function along with their frequency and time-related measurements. Of these, SIM from Meyer Sound and TEF from Goldline are sold with microphones and are calibrated at the factory.

Smaart (any version), SpectraFoo, SysSid, PC-RTA and other similar systems utilize third-party microphones and preamps that must be calibrated by the end-user, using an acoustic calibrator or piston.

Once calibrated, these systems provide very good accuracy and feature a number of variables (weighting filters, etc).

Unfortunately, they are not really “handheld” nor nearly as portable as stand-alone SLMs. And (most importantly) they do not retain their calibration from site to site and therefore require an acoustic calibrator for continued use in measuring absolute SPL’s.

Another product that is a variation of a SLM deserves mention here. SPL “management” systems have been available for perhaps a decade and are directly related to the issues of sound system volume levels.

These simply employ microphones and a software-based SLM system that provides either an obnoxious visual warning or more extreme reactive functions (such as muting of the sound system signal!) when the predetermined SPL limit has been reached or surpassed.

Calibration
In situations where the accuracy of sound level measurements must be recorded and/or verified, it is necessary to calibrate the measurement system both before and after the measurements are made.

Software-based electroacoustic measurement systems that feature an SLM function also need to be calibrated if used for absolute SPL measurements.

An acoustic calibrator provides the standard 94 dB or 114 dB (or both) 1 kHz test tone and should also include the proper size opening or adaptor rings for the microphone(s) you intend to employ.

As is the case with SLMs, calibration devices are manufactured to several levels of tolerance (see table near beginning of article) and they have historically been prohibitively expensive.

Recently there have been several acoustic calibrators introduced that provide very good accuracy for sound system and acoustics measurement.

SLMs may also be calibrated with a pistonphone, but the cost of this device is prohibitively high and the high level of accuracy it provides is not practical for most general sound system related measurements.

Why We Use Them
The simple and snappy answer as to why we use SLMs: so we can know what’s going on.

The need for accurate and repeatable sound level measurement is due simply to the tendency for humans to aurally perceive sound in a very subjective manner. This applies to all aspects of sound perception in the audio field: frequency content, mix “correctness”, system EQ, dynamics, etc.

At some point or another we will disagree on the relative loudness being produced plus these other subjective sound qualities.

Whereas other electroacoustic measurement tools remain very subjective in how we interpret the data they provide, a reasonably accurate SLM provides an irrefutable means to objectively verify how loud the sound is (or is not).

SLM measurements may also be used to verify exactly how much the sound level has been changed (as noted in the section on relative versus absolute measurement).

Anyone involved in live sound system work is aware that there is an ongoing and increasing problem with the issue of volume levels produced in public performance, and specifically, in pop music reinforcement.

It’s wise for anyone involved in this field to become familiar with the actual measured numbers, not to mention the need to continually monitor what we are doing. Issues of personal health and legal liability are an ongoing and increasing concern.

Measurement of unintended noise levels and/or the effectiveness of acoustic treatment are additional exercises that most live sound folks eventually will become involved in.

How We Use Them
For basic every day sound system level measurement, using an SLM is fairly straightforward.

The standard SLM employs a free field microphone and is intended to be held at arm’s length and tipped up slightly towards the sound source.

As you move away from the source and into the reverberant or diffuse field, or when measuring in a multiple source sound field such as on stage, the angle of incidence of the microphone becomes much less of a concern.

When measuring noise levels at installation sites, some care should be taken. If measuring the noise level of a rack room, for example, be sure that the SLM is positioned away from walls, floors and other large boundaries.

Signal-to-noise levels are seldom an issue for loudspeaker system measurement because we can simply turn the system up, more than far enough above the noise floor.

But when measuring the noise from a loudspeaker system (hum, hiss and/or the fan noise that may exist in self-powered systems), or from a mixing console and its power supply, etc., or from other common noises such as HVAC systems, vacuum cleaners, power tools, etc., the unintended noise must be at least 10 dB-SPL lower than the noise being measured.

When measuring and logging levels to help resolve a dispute, both A and C weighted measurements should be provided, as should the distance from the source and any other detailed notes.

Also don’t forget that a non-annotated measurement (“00.0 dB-SPL”) will be interpreted to be flat (unweighted).

General Measurement Guidelines
When conducting sound pressure level measurements there are a few basic rules that apply:

• Keep the SLM at arm’s length, aimed in the direction of the sound source(s) and positioned away from room boundaries. By the way, the mixing surface on a large (40-plus channel) mixing console will effect SPL measurement when the SLM is laid down on the mixing surface or angled against the meter bridge.

• For almost all sound system measurements, use the A-weighting filter and Slow response setting.

• Be sure that you’re measuring within the range that you have selected. SLMs with multiple range positions and those with analog displays, in particular, may provide incorrect readings when the SPL is over/under the selected range.

• Prior to making SPL measurements, ensure that the SLM has been turned on for a few minutes and has reached room temperature.

• Wind and air-blowers will effect SPL measurements.

• Be aware of where you are taking your measurements. For example: if you measure from within the area where the bass energy from the sound system has combined (aka “power alley”) or within a standing wave null, the readings will be non-representative of the SPLs for the majority of the audience.

Tom Young is principle consultant for Electroacoustic Design Services in Connecticut, and he has designed hundreds of systems for churches and similar spaces. Tom’s also the moderator of the Church Sound Community Forum here on ProSoundWeb.

More articles by Tom Young on PSW:
Time Is On Your Side (If You Want It To Be) With Sound System Alignment
Microphone Techniques For Drums In Contemporary Worship Environments
The Aux-Fed Subwoofer Technique Explained

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Ashly Audio Ampliers, DSP Key System Upgrade At North Carolina Event Center

The Western North Carolina Agricultural Center (WNCAg) in Fletcher, NC, hosts a wide range of events, such as equestrian shows, dog shows, and gun shows that may run in parallel with conventions, meetings, and workshops.

WNCAg recently updated the Davis Arena, which offers 45,000 square feet of exhibition space plus meeting rooms and hospitality suites, with upgraded sound reinforcement.

Local A/V integrator B&R Audio deployed Ashly Audio amplifiers and signal processors, including a Pema multi-channel amplifier with on-board Protea DSP processor. To give the non-technical staff a transparent user interface, the company provided WNCAg with Ashly’s new FR-8 and FR-16 networkable remote fader controls.

“The center needed a very flexible sound reinforcement system that would allow any member of the staff to play music from a variety of input sources and deliver announcements that would be clearly intelligible,” says Bruce Jensen, president and owner of B&R Audio, who led the design and installation. “In a nutshell, they asked for input volumes and mutes as well as output volumes and mutes to a number of zones, including several far-flung buildings. But perhaps even more importantly, they wanted something that would not intimidate even the most technologically bashful user.”

Jensen had worked with Ashly Audio before and caught wind of the FR-8 and FR-16 before they were officially released.

“Ashly’s FR-8 and FR-16 remote controllers were perfect for this installation,” he continues. “And because Ashly’s support is so complete and reliable, I had no fear that I’d be stranded with a semi-functional beta if I pressed the company for an early model.”

The FR-8 and FR-16 offer eight and sixteen faders and mutes, plus a master fader. The integrator has full control over which input, output, or mixer volumes are controlled by each fader, with flexible range limits and defaults.

An Ashly ne24.24M DSP Matrix Processor handles all of the complex input processing, matrix logic, and output processing behind the scenes. Jensen was careful to set the fader limits and the ne24.24M’s output protection limiting in such a way that even a complete novice can do the system no harm.

Four Ashly Audio amplifiers provide power to the Davis Arena’s distributed loudspeakers. An ne2400.70 2-channel networkable amplifier rated at 1200W into 70V feeds an outdoor collection of Community R1-64Z weather-resistant horn-loaded loudspeakers.

An ne800.70 Network Amp covers the bathrooms and corridors, with ne1600pe.70 and ne2400pe.70 Networkable Amplifiers powering the main exhibition hall.

All indoor loudspeakers are down-firing SoundTube models, which do a good job of keeping energy off the reverberant walls. Both main exhibition hall amplifiers deliver loudspeaker conditioning with integrated DSP.

“Like everything with Ashly, when I buy the amps, I know I’m also getting the company,” says Jensen. “The stuff is solid, and on those rare instances when there is a problem, the people at Ashly are on it right away. Ashly installations always run smoothly.”

Because several of the buildings that require a music and/or announcement feed from the Davis Arena are quite distant, Jensen convinced the WNCAg for forego the miles of speaker cable (literally) they had been using in favor of fiber optics. A MultiDyne DAM-1000 provides conversion on either end of a pilot feed.

“They weren’t sure that they wanted to commit to fiber optics, so we did one trial run,” Jensen says. “They love it, and we’ll be upgrading the other distant feeds in the near future.” Every aspect of B & R Audio’s installation is future-proof. Jensen continued, “One of the nice features of the Ashly ne24.24M is that the input and output count is totally modular. So when the needs of the event center change, we can accommodate those changes without having to rip everything out.

As part of the renovation, the WNCAg added three classroom spaces that are large enough to require sound reinforcement.

“It was the perfect application for Ashly’s Pema Multi-Channel DSP Amplifier,” Jensen explains. “Because the Pema has everything I needed inside a two-rack space unit, the installation was simple, quick, and reliable.”

The Ashly Pema 4250 four-channel delivers input conditioning, switching logic, and loudspeaker conditioning (again for SoundTube loudspeakers). Similar to the FR-8 and FR-16 in the big room, each classroom has an Ashly neWR-5 wall-mounted remote control that allows users to easily select inputs and volumes.

Ashly Audio

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Wednesday, January 25, 2012

Behringer Introduces FIREPOWER FCA610 & FCA1616 Recording Interfaces

At the NAMM 2012 Show, Behringer introduced the FIREPOWER USB/FireWire audio interfaces, the FCA610 and FCA1616.

The FCA610 and FCA1616 incorporate 24-bit/96 kHz A/D-D/A converters, support Windows XP/Vista/7 plus Mac OS X, and provide onboard phantom power for use with condenser microphones. 

Due to its small size and low-latency operation, 6-input/10-output architecture, and two XENYX mic preamps, the FCA610 is a tool for traveling musicians who record and edit on their laptops. The portable FCA610 can receive power from a computer’s 6-pin FireWire bus or via the included external power supply.

Built-in MIDI I/O allows for easy connectivity with keyboards and other outboard MIDI hardware. All standard I/O formats are supported, including analog and S/PDIF (both coaxial and optical).

The half-rack-space FIREPOWER FCA610, which stows easily in a travel kit, can also be used as a premium 2-channel mic preamp and A/D-D/A converter.

With an expanded 16-channel I/O, four XENYX mic preamps and ADA8000 ADAT connectivity, the FCA1616 is more suitable for permanent applications as well as live performance multi-track recording rigs.

All standard I/O formats are supported; including analog, S/PDIF (coaxial and optical), ADAT and S/MUX, and a built-in MIDI I/O allows the user to connect keyboards and other outboard MIDI hardware.

The single rack-space FCA1616 also features eight analog Inserts for use of external effects such as compressors, gates and EQs. A dedicated power supply comes with the unit.

Included with both FIREPOWER interfaces is a massive software download at behringer.com that includes the widely popular Audacity audio editor, as well as a selection of audio software such as Podifier, Juice, Podnova and Golden Ear. Also included are more than 100 virtual instruments and 50 FX plug-ins.

FIREPOWER Features:
• Low-noise, high-headroom audio interface with 24-bit/96 kHz resolution
• Operates as multi-channel audio and MIDI interface via FireWire and USB2.0
• XENYX mic preamps with individual switches for Phantom Power, Pad, Low Cut and Hi-Z
• Direct Monitoring and Main Volume control on hardware front
• Two headphone outputs with individual volume control, mono and source signal select for flexible monitoring purposes
• Level control of stereo or 7.1 active loudspeaker systems with a single knob turn
• Smooth cross-fading between inputs and DAW playback signals
• Status and signal presence indication for all analog and digital I/O
• Standard port for Kensington security lock

Behringer

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Tuesday, January 24, 2012

Radial Engineering Announces The New Firefly Tube Direct Box

Radial Engineering has introduced the Firefly tube direct box, a fully discrete class-A unity gain amplifier designed for both studio and live performance.

Radial company president Peter Janis explains: “We have always wanted to launch a tube DI to round out our product range. But over the past two years, we have been sidetracked as we developed the Workhorse and our many 500 series modules. We finally got back on this project and are pleased to bring the Firefly DI to market.”

The Firefly begins with two inputs, each of which features a separate level control to enable the artist to set each instrument with optimal gain.

Switching between inputs can be can be done using the front panel switch or via the optional JR2 remote footswitch. The instrument signal is immediately routed to a tuner output that is always on.

When used with the JR2 footswitch, the Firefly may also be muted remotely for quiet on-stage tuning. Both the footswitch and front panel are equipped with LED indicators for status monitoring.

Following the Radial JDV, the Firefly’s front end circuit is 100 percent discrete class-A and is void of any circuit stabilizing negative feedback. This produces a more open, less constricted sound. 

The Firefly is also equipped with Drag Control load correction that enables the artist to adjust the load on the magnetic pickup for a much more natural rendering. When bypassed, the load jumps to 4 meg-ohms enabling the Firefly to be used with piezo pickups such as common with upright bass and other acoustic instruments.

The exceptional warmth and detail is achieved by combining Radial’s unique front end with an all new12AX7 tube drive circuit. Contrasting the input sensitivity with the output drive enables the artist to fine tune the grit or edge to give the sound more character. 

Firefly comes shipps with two 12AX7 tubes, a select premium tube for audiophile performance and a low-fi version for added growl.

A fully variable high-pass filter enables the engineer to set the bass cut-off frequency for optimal layering. This ‘Nashville trick’ lets you set the cut-off to better match the size of the instrument whereby a lower cut-off would be used on contrabass, slightly higher on acoustic and higher again on fiddle or mandolin. By setting a different cut-off for each instrument one can eliminate resonance while still retaining the character.

Connectivity is extensive: The rear panel begins with two stacked 1/4-inch instrument inputs. A second set of stacked 1/4-inch jacks presents the user with a buffered thru-put that delivers either the original instrument’s tone or the output from the tube circuit.

Below, an insert jack enables one to add in effects in series with the tube drive circuit and apply the effects to the overall sound.

The third set of stacked jacks feature a tuner output and a TRS jack for the JR2 remote control. The Radial transformer coupled XLR output is outfitted with a ground lift switch and a 180-degree polarity reverse. This can be helpful when controlling feedback or interfacing with older vintage gear.

Power is supplied via an exterior switching supply for 100- to 240-volt operation and delivers a variable output that ranges from a typical unity gain DI level to a full +4 dB line level for direct recording.

The Firefly is road ready with 14 gauge steel construction plus a protective zone around the controls and switches. This is augmented with steel cased switches and potentiometers plus a double sided military grade PCB for added life.

The Firefly comes with a carry handle that may be removed should rack mounting be needed for touring using optional rack-mount kits.

Radial Engineering

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