Thursday, August 09, 2012

To Match Or Not To Match? Paying Attention To Impedance

Principles involved in feed a number of inputs from a single signal source

A common task in “audioland” is the need to feed a number of inputs from a single signal source. This may include driving a rack of amplifiers, providing feeds to the press, or distributing a signal around a building or campus.

The methods used to accomplish this range from the profoundly simple to quite complex, and the appropriate method can only be determined after sizing up the situation.

Impedance matching means that an output is terminated with a “mirror” input impedance. This configuration yields maximum power transfer, and more importantly, reduces reflections from a load back to the source.

In multimedia systems, the matched interface is used for very high frequency signals. These include video, antenna and digital interfaces.

One drawback of the matched interface is that active or passive splitters must be used if the source must drive multiple inputs. Otherwise the impedance match is violated and problems result.

One of the most common mistakes in audio is to attempt to apply this interfacing method to the basic analog interfaces that dominate today’s sound reinforcement systems.

In a constant voltage interface, an electronic signal source with a low source impedance (i.e. an output) is used to develop a signal voltage across a high load impedance. The minimum ratio between the source Z and load Z is one order of magnitude (1:10).

This scheme is used universally in the audio industry for passing signals from component to component. One utility of this interface is that it provides the possibility of driving multiple parallel loads from a single source without additional hardware.

The stipulations are as follows:
—The parallel combination of all loads cannot violate the 1:10 minimum impedance ratio.
—The path length (interconnecting cable) must be short when compared to the wavelength of the highest frequency component of the signal.

Figure 1: The relative lengths of an audio waveform, a VHF waveform, and an audio cable. (click to enlarge)

Because the speed of propagation of electricity approaches the speed of light, and audio cables are typically less than a few hundred feet, the second condition is easily met in the bulk of audio applications.

Radio frequency, digital, and video signal wavelengths are much shorter, and the impedance-matched interface must be used in lieu of the constant voltage interface to prevent signal degradation. (Figure 1)

Figure 2 below shows an equivalent circuit of a single source driving multiple loads. Note that even though the load impedances are not the same, this is a parallel circuit so all of the inputs have the same voltage impressed across them.

Signal distribution requires a simple “Y” cable connected from the source to the multiple loads. This is a perfectly acceptable method of distributing the signal from a source to multiple loads.

There is no need for impedance matching if the components involved are typical analog audio products or even digital products if they are being fed an analog signal.

A drawback to Y-cable signal distribution is the lack of isolation between the individual loads and the source. For instance, a short circuit across any of the inputs will kill the signal to all of the inputs.

For this reason (and others), this method is not recommended for driving loads that lie outside of the equipment rack that houses the source. In these cases, load isolation can be achieved by using a distribution amplifier (DA).

Figure 2: An equivalent circuit of a single source driving multiple loads. (click to enlarge)

The DA provides a single high impedance input for the signal from the source, but provides buffered low impedance outputs that can be used to drive the remotely located loads. The load buffering is achieved by using an active stage for each of the DA’s outputs.

A short across any one output is buffered from the other outputs by the active stage (Figure 3). Note that this is not impedance matching since the output-to-input impedance ratio is still at least 1:10.

Figure 3: Isolation between source and loads. (click to enlarge)

It must be strongly emphasized that while the Y-cable makes an excellent signal splitter it should NEVER be used as a mixer. Doing so places the source device under a load, resulting in an increase in output current that can lead to distortion under high signal conditions.

When a mixer is needed - get a mixer.

While the DA solves the isolation issue, we’re not out of the woods yet. Another problem that plagues distribution systems results from multiple ground connections between the various components.

These “shared” ground paths include the AC safety ground, the cable shields, and possibly connections to the building ground through equipment racks, etc.

Noise currents will circulate through these “ground loops” (Mother Nature does this without our permission) and possibly infect the audio signal if this parasitic ground current finds its way onto a circuit board.

Isolation devices can allow the audio signal to be coupled from an output to an input with no physical wire joining the two circuits, eliminating at least one of the ground loops.

Transformer isolation allows the signal to be coupled via induction (Figure 4).

Optical isolation uses pulsed light to couple the signal, but usually requires that the signal be converted to a digital format.

The transformer has an advantage in that the signal can remain in analog form.

Figure 4: An isolation transformer. (click to enlarge)

The irony is that the same mechanism that allows a signal to be coupled between two circuits inductively also allows power supply fields to be coupled into ground loops (Figure 5).

We’re faced with the common engineering task of maximizing the effect when it helps us and minimizing it when it is working against us.

Putting all of these mechanisms to work, an active distribution amplifier with transformer balanced inputs and outputs may be the optimum way of distributing an audio signal to multiple components.

Figure 5: Ground loops and power supply radiation form an unwanted transformer in a sound system. (click to enlarge)

The active stages buffer the inputs from short circuits, and the transformers allow ground loops to be interrupted while allowing the signal to pass, while at the same time providing excellent common-mode rejection.

Many DAs also include level controls for each output, allowing the signal level to be optimized for mic or line level devices.

When distributing an audio signal to multiple inputs, don’t overlook the simplicity of simply using a properly wired Y-cable to accomplish the task. If the signal needs to extend beyond the rack, a good DA will easily justify the investment. 

Pat and Brenda Brown own and operate SynAudCon, conducting training seminars around the world.

Posted by Keith Clark on 08/09 at 01:11 PM
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Hosa Technology 2nd Generation Elite Series Mic Cables Now Shipping

Available with Neutrik XX-Series connectors plus a new nylon webbing over the cable’s PVC jacket, the Elite Series delivers performance attributes that are every bit on par—if not superior to—more costly boutique cables.

Hosa Technology is pleased to announce that a significant upgrade of the company’s popular Elite Series microphone cables is now shipping.

Available with Neutrik XX-Series connectors plus a new nylon webbing over the cable’s PVC jacket, the Elite Series delivers performance attributes that are every bit on par—if not superior to—more costly boutique cables.

Available in both Lo-Z (XLR3F to XLR3M) and Hi-Z (XLR3F to ¼ inch TS) configurations, the cable used in the Hosa Elite Series is a vitally important contributor to overall audio performance.

These cables use 20 AWG Oxygen-Free Copper (OFC) conductors that reduce resistance in order to facilitate maximum signal transfer.

Polyethylene dielectrics reduce capacitance for crystal-clear high-frequency transmission while conductive PVC reduces handling noise.

Further, a 95% OFC braided shield is employed for noise-free signal transmission. Take all this and complete it with nylon webbing over the cable’s PVC jacket, which is cut- and abrasion-resistant for a lifetime of trouble-free use, and the end result is a cable audio professionals can place their trust in.

A clean, reliable connection to one’s mic preamplifier or console input is of paramount concern for any audio engineer or recording enthusiast, and the Hosa Elite Series microphone cable cuts no corners in this regard.

The Neutrik XX-Series connectors employ gold-plated contacts for corrosion resistance and superior signal transfer and utilize a zinc die-cast housing for rock-solid reliability.

With a polyurethane gland to prevent cable kinking for longer cable life and chuck-type strain relief for maximum cable retention combined with a sleek, ergonomic design for easy handling, these connectors deliver the ideal blend of performance and long-term reliability.

“These second generation Elite Series cables offer exceptional audio performance and have been field tested for maximum reliability,” said Jonathan Pusey, Hosa Technology’s Vice President of Sales and Marketing. “With a rich feature set and a highly competitive price, we are extremely optimistic that this product line will find favor with a wide range of customers.

“The Elite Series has always been an exceptional product and now, with the addition of Neutrik connectors and abrasion-resistant nylon webbing for a lifetime of dynamic, noise-free sound quality, I’m confident these mic cables offer the best possible combination of performance and value.”

The Hosa Elite Series microphone cables are available in lengths from 3 to 100 feet and carry MSRP pricing that ranges from $42.75 – $203.55.

The Elite Series is in stock and available now.

Hosa Technology

Posted by Keith Clark on 08/09 at 09:24 AM
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Our Lady Of Fatima Parish Benefits From Tannoy QFlex Loudspeakers

The installation of a pair of Tannoy QFlex 16 digitally steerable array loudspeakers at Our Lady of Fatima in Seattle, WA has eradicated difficulties parishioners had hearing the Word, while maintaining the parish’s ability to offer historically accurate musical performances.

The installation of a pair of Tannoy QFlex 16 digitally steerable array loudspeakers at Our Lady of Fatima in Seattle, WA has eradicated difficulties parishioners had hearing the Word, while maintaining the parish’s ability to offer historically accurate musical performances.

The existing sound system was installed back in 1968 when the new church was built. Needless to say, it was long overdue for a change.

“It was at least thirty years old,” says music director, Matthew Loucks. “Some speakers weren’t functioning and many had been clustered on one side of the sanctuary. In order to hear speech on the opposite side of the room the system had to be very loud – it was a very lopsided configuration.”

As part of a 2010 audio upgrade, a new pipe organ was purchased, while existing wall-to-wall carpeting covering the 650-seat sanctuary’s was removed replaced with Terrazzo flooring. While that enhanced the sound of the choir and encouraged the congregation to lift up their voices in song, it only exasperated the overall lack of vocal intelligibility during services. The church leadership hired Lynwood, WA-based Morgan Sound to design and install a new sound reinforcement system with the primary goal of speech intelligibility.

“Even when we had carpet, it was impossible to hear articulated speech in the back of the room,” says Stephen Weeks, A/V consultant and project manager for Morgan Sound.

Weeks specified a pair of Tannoy QFlex 26 digitally steerable arrays to dramatically enhance speech intelligibility and subtly reinforce musical performances by resident ensembles and visiting performers like The Vienna Boy’s Choir.

“The QFlex array speakers sound natural, clean and clear; particularly in the articulation of high frequencies,” explains Weeks. “QFlex just does a better job in the 4k to 20k range than their competitors and we could manipulate the beam substantially, so less energy is directed at the front of the room and, as you go further back, it gets louder.”

The compact footprint of the QFlex 16 also appealed to aesthetic concerns the church leadership had voiced to Weeks. Mounted discreetly on either side of the altar behind acoustic cloth on an imposing white rock wall their visual impact was minimal while their sonic impact substantial.

“I have nothing but high praise for them,” Loucks says. “There are only two speakers, but they do the job extremely well. QFlex saved us from the hassle and cost of applying acoustic treatments, and they allowed us to preserve the reverberant environment we wanted for music.

“When you’re reading from scripture, you want the Word to be understood all the times. Now, even when there are only a handful of people at our daily 8 AM mass, speech is highly intelligible.”

Morgan Sound

Posted by Keith Clark on 08/09 at 08:55 AM
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Sennheiser & Full Compass Sponsoring Recording Clinic Led By Leslie Ann Jones Of Skywalker Sound

Grammy Award winning engineer to Illustrate vocal mic techniques and best practices

Sennheiser and Full Compass Systems are co-sponsoring an audio recording clinic on Tuesday, September 11 at the Full Compass facility in Madison, WI, featuring Grammy Award-winning sound engineer Leslie Ann Jones, who will demonstrate vocal recording techniques and cover best practices when recording live vocals in the studio.

Attendees will be provided with a pair of Sennheiser HD 449s, enabling them to monitor both recording and playback.

The event will feature door prizes including a K-array Piccolo audio system, a Neumann TLM 102 microphone and a TRUE Systems P-SOLO microphone preamplifier.

Leslie Ann Jones, who is director of music recording and scoring with Skywalker Sound, has been a recording and mixing engineer for over 30 years.

She began her career at ABC Recording Studios in Los Angeles in 1975 before moving to Northern California in 1978 to accept a staff position at the legendary Automatt Recording Studios. There she worked with such artists as Herbie Hancock, Bobby McFerrin, Holly Near, Angela Bofill, and Narada Michael Walden, and started her film score mixing career with “Apocalypse Now.”

From 1987 to 1997, Jones was a staff engineer at Capitol Studios located in the historic Capitol Records Tower in Hollywood, where she recorded projects with Rosemary Clooney, Michael Feinstein, Michelle Shocked, BeBe & CeCe Winans, and Marcus Miller, as well as the scores for several feature films and television shows.

In 2003, Jones was nominated for a Grammy Award for Best Engineered Recording, Classical, and received a Grammy Award for The Kronos Quartet’s recording of Berg: Lyric Suite, which won Best Chamber Music Album. This year, she won a Grammy Award for Best Engineered Album, Classical for Quincy Porter: Complete Viola Works by Eliesha Nelson & John McLaughlin Williams.

Go here for more information on this event. Seating is limited.

Full Compass Systems

Posted by Keith Clark on 08/09 at 06:51 AM
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Audio-Technica U.S. Appoints Javier Tiburcio As Support Specialist For Mexico, Central America

Will be responsible for providing product training and sales support to A-T distributors and customers

Audio-Technica U.S. has appointed Javier Tiburcio as training & sales support specialist for the territories of Mexico and Central America.

He will be responsible for providing product training and sales support to A-T distributors and customers in Mexico, Central America and South America, as he works as part of the team to grow the brand and increase sales within these regions. The announcement was made by Philip Cajka, Audio-Technica U.S. president and CEO.

Tiburcio has broad experience in microphone sales and training. A native of Acapulco, he has lived and worked his entire career in Mexico, and will continue to be based there. 

His previous positions at Hermes Music and Grupo Imis have included sales and marketing responsibilities for the A-T brand in Mexico.

An experienced musician and audio engineer, Tiburcio studied music at the School of SUTUM, and digital audio and audio for broadcasting at CECAT, gaining expertise in microphone placement for live sound and recording.

He has visited music industry dealers throughout Mexico and conducted clinics and training seminars at a variety of music expos with musicians like Moderatto Drummer Elohim Corona.

“Javier is a tremendous asset for us to expand our sales and training strategies in Mexico, Central America and South America,” said Cajka. “He comes to Audio-Technica with a wealth of experience, and we look forward to all of the great things we can accomplish together.”


Posted by Keith Clark on 08/09 at 06:17 AM
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Furman Now Shipping Contractor Series CN-15MP MiniPort

Extends SmartSequencing technology to components outside the equipment rack

Furman is now shipping its Contractor Series CN-15MP (15A capacity) MiniPort.

Designed for components outside the equipment rack, it combines Furman’s SmartSequencing technology with power protection and optional compatibility with Panamax/Furman’s BlueBOLT hosted remote power and energy management platform.

The CN-15MP features one pair of AC outlets with configurable delay on/off options, while Extreme Voltage Shutdown circuitry protects connected equipment against under/overvoltage conditions.

When connected to a Furman SmartSequencer (CN-1800S or CN-2400S), the CN-15MP’s SmartSequencing technology allows bidirectional, safe sequenced power on/off of remotely located equipment with the simple press of a button or turn of a key.

Remote control/monitoring is available via Panamax/Furman’s BlueBOLT cloud-based platform or third-party control systems when utilizing a BB-RS232 adaptor with the SmartSequencer.

The CN-15MP can also be integrated with legacy (non-Contractor Series) and non-Furman power conditioners/sequencers via remote terminal blocks.

“Packed with features for the ultimate in flexibility, the CN-15MP provides a convenient way to extend the benefits of our SmartSequencing technology to remotely located equipment,” says Dave Keller, senior vice president of sales and marketing for Panamax/Furman. “When combined with our SmartSequencers, installers can also take advantage of our cloud-based BlueBOLT platform for remote outlet control, energy monitoring, email alerts during power events, and more.”

The CN-15MP is available now to Certified Furman Contractor Series resellers. More information is available at


Posted by Keith Clark on 08/09 at 06:09 AM
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Cross Pointe Ministries Selects WorxAudio Technologies

To ensure a high level of speech and music reproduction capabilities throughout the seating areas, the church installed a new sound system consisting of WorxAudio X1i-P line array and TL218SS-P subwoofers.

The Cross Pointe Ministries campus recently celebrated the opening of their new building that provides seating capacity for over one thousand worshippers.

To ensure a high level of speech and music reproduction capabilities throughout the seating areas, the church installed a new sound system consisting of TrueLine components from WorxAudio Technologies.

Terry Hayes, owner of Tupelo, MS-based Pro Concert Music, Inc., was contracted to design and deploy the church’s new sound reinforcement system.

After discussions with church management to ascertain their goals for the system, Hayes deployed a central cluster consisting of six WorxAudio Technologies X1i-P powered line array elements augmented by two WorxAudio TL218SS-P powered subwoofers—one on each side of the stage area.

“In addition to consistent sound coverage, we were very concerned about the visual aesthetics of the space and, on that note, wanted to minimize line of sight issues,” explained Hayes. “For this reason, we elected to install a central loudspeaker cluster.

“These speakers needed to have broad horizontal dispersion and this was a key factor in our selection of the WorxAudio X1i-P. With a dispersion pattern of 160 degrees horizontal by 10 degrees vertical, we determined that a 6-element hang would properly fill the space.”

“I provided the WorxAudio Technologies support staff with the measurements of the room and emphasized the need to ensure coverage across the sanctuary’s 120-foot width,” Hayes continued. “After WorxAudio modeled the space in EASE Focus 2, I proceeded to install the system based upon their recommendations.

“I emphasized the fact that this building is metal construction with a concrete floor, as I was very concerned about echoes. WorxAudio’s TrueAim Grid made the process of installing the loudspeakers a remarkably easy task.

“The bottom of the cluster is positioned 21 feet above the congregation floor and, thanks to the precise throw of the loudspeaker enclosures, echo was held to a minimum. With the congregation in the church, there was no need for acoustical treatment!”

In addition to the six forward facing loudspeakers, there is an additional X1i-P mounted on the rear of the TrueAim Grid. This loudspeaker is positioned down toward the stage and serves as an on-stage monitoring system. It is completely hidden from the congregation’s view.

“The two WorxAudio TL218SS-P subwoofers do a terrific job of providing the low end punch for the system,” Hayes added. “With two 18-inch transducers in each enclosure, there’s plenty of low frequency sound from this system.

“Music plays a prominent role in worship services here, so these subs are a vital part of the overall sound. The fact that the entire system is self-powered was yet another factor that figured prominently in my selection of the WorxAudio equipment, as it minimized noise and power loss from long cable runs and the amps with DSP are part of each enclosure’s design.

“As a result, there was no need to find a space to house a rack of heavy power amps.”

Being a member of Cross Pointe Ministries, Hayes knew that any shortcoming of the sound system would haunt him on a weekly basis. He knew he could count on WorxAudio’s customer support services to help him design and deploy the right system.

“WorxAudio’s support was terrific,” Hayes reports. “They did a great job of modeling the room and had my back every step of the way—all without incurring any additional expense. Speech intelligibility is terrific and the sound remains consistent throughout the entire space. The client is thrilled and I’ve had several other churches come by to listen and observe the new system. I can say with confidence, they’re the best speakers I’ve ever heard!”

WorxAudio Technologies

Posted by Keith Clark on 08/09 at 06:02 AM
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CPL Takes Delivery Of Its First Yamaha CL5 Digital Console

Also have sights set Yamaha’s new CL1 and CL3 variations

West Midlands, UK-based AV specialist Central Presentations Ltd. (CPL) is one of the first UK rental companies to take delivery of the Yamaha CL5 digital mixing console, introduced earlier this year.

CPL already has a large stock of Yamaha LS9 and other desks. One reason the CL5 was chosen is that it runs on the same operating principals as these, so all their engineers who are used to working with Yamaha digital consoles will be able to use the CL5 immediately, and quickly familiarize themselves with its many features.

The CL5 system was supplied to CPL by the Birmingham office of LMC Audio Systems.

“The compact size and weight for 72 channels is a ‘no brainer’,” explains CPL managing director Matthew Boyse, who adds that it is “perfect“ for so many of the corporate events and presentations that CPL services. “It is a truly multi-purpose console that will fit neatly into any space and transport very easily.”

The CL Series is the first Yamaha console family to run natively using the Dante digital audio network protocol, with connections between Dante elements via Cat5e cable further reducing overall weight and truck space requirements.

Other Dante devices can be connected to the network keeping the domain completely digital, and in this context, CPL’s CL5 was supplied with two Yamaha RIO-3224-D stage racks giving 64 inputs and 32 outputs which connect via Cat5e in a simple daisy-chain or redundant star-network. Up to eight RIO stage-boxes can operate on the same network, including the smaller RIO1608-D, providing excellent expansion and audio distribution potential.

James Lawford, sales manager at LMC Audio Systems, states, “The availability of the RIO-1608-D stage box, and the move towards Dante by other third party manufacturers, will allow CPL to harness the power of a fully integrated digital audio network, working across standard Cat5e cabling and Ethernet switches, future-proofed with the move towards AVB, and with the reliability, ease of use and wide industry acceptance of a Yamaha console.”

The CL Series also includes a Premium Rack with a Rupert Neve Designs five band Portico 5033 equalizer and a Portico 5043 compressor/limiter. Up to eight Premium Rack devices can be assigned, including VCM technology emulations of other classic analog devices.

Via Dante, the console is also optimized for very straightforward live multitrack recording, utilizing Steinberg Nuendo Live that is supplied with the system and runs on Mac or PC, or any other digital audio workstation.

Nigel Griffiths of CPL notes, “You can set the system up to minimize the need to run analog multicores and the multitrack record and playback facility, without needing additional sound cards or having to stray back into an analog domain, is brilliant. I can’t wait to get my hands on it.”

CPL is already anticipating hot demand for the console and it becoming a valuable cross rental item. CPL has arranged with LMC to provide training days for both its full-time and regular free-lance engineers and technicians, enabling them to maximize their experience with the console.

In addition to further CL5 purchases, Boyse also has his sights set on Yamaha’s new CL1 and CL3 variations, which will become available late summer, the idea being that the company will stock the full CL range.

Central Presentations Ltd. (CPL)

Posted by Keith Clark on 08/09 at 05:51 AM
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Entertainment Legend Donny Osmond Receives Audio-Technica 50th Anniversary Microphone

As Donny and Marie Osmond continue their hugely successful run at Las Vegas’ Flamingo Hotel, Audio-Technica is proud to be the microphone of choice for the superstar siblings’ act.

As Donny and Marie Osmond continue their hugely successful run at Las Vegas’ Flamingo Hotel, Audio-Technica is proud to be the microphone of choice for the superstar siblings’ act.

Between recent performances, Donny Osmond was presented with a limited edition AT4050URUSHI Multi-pattern Condenser Microphone.

The AT4050URUSHI, a visually striking version of Audio-Technica’s acclaimed AT4050, was created to commemorate A-T’s 50th anniversary and sports a stunning traditional urushi lacquer finish with hand-painted Japanese maple leaves.

The microphone was presented to Osmond by Philip Cajka, Audio-Technica U.S. President & CEO, and Michael Edwards, Audio-Technica U.S. V.P. Professional Markets.


Posted by Keith Clark on 08/09 at 05:29 AM
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Wednesday, August 08, 2012

Line 6 Ships StageScape M20d Smart Mixing System For Live Sound

StageScape M20d streamlines and accelerates the process

Line 6 has announced the availability of StageScape M20d, a smart mixing system for live sound with a unique touchscreen visual mixing environment.

The visual mixing system replaces the traditional mixer channel strip with intuitive touchscreen control. In Perform Mode, a graphic display of the stage setup uses icons to represent each performer or input.

Further, a color-coded encoders provide immediate access to level control. A single touch on a performer’s icon gives access to all parameters relating to that channel, from basic tweaks to deep effects editing.

The audio signal chain can be controlled via an X-Y tweak pad. Drag a finger toward common sound descriptors like “bright” or “dark” and multiple parameters adjust simultaneously to achieve that sound.

Deep Edit mode gives more experienced operators access to every effect parameter via a familiar plug-in style interface.

StageScape M20d streamlines setup with auto-sensing mic and line inputs and outputs that can detect when a connection is made and automatically configure the channel gain, EQ, effects and routing.

A host of recording options are also available. StageScape M20d provides multi-channel recording in high-resolution, 24-bit WAV files to SD card, USB drive or direct to computer, enabling easy capture of every rehearsal and performance.

StageScape M20d provides professional-grade effects on every channel, including fully parametric EQs, multi-band compressors, feedback suppression and more.

In addition, four master stereo effects engines are available, comprising reverbs, delays and a vocal doubler. Users can configure channel effects quickly with a wide range of channel presets that cover everything from individual drum settings to lead vocals.

The mixer can be controlled by using one or more devices with StageScape M20d. This makes it possible to set individual monitor mixes from the stage, or adjust the front-of-house mix from any location inside the venue.

Equipped with the L6 LINK digital networking protocol, StageScape M20d allows easy configuration and control of PA systems of any scale. Connect StageScape M20d to L6 LINK-enabled StageSource speakers and the system automatically configures stereo signals and effects, sets individual component levels and adjusts individual loudspeaker performance.

“For musicians who take care of their own live sound, getting great results consistently can be a time-consuming and frustrating experience—particularly when they want to be focused on performing,” says Simon Jones, vice president of new market development at Line 6. “StageScape M20d answers those challenges by reinventing mixer workflow to better empower musicians to dial in a great mix with speed and ease.

“With a smart design to dramatically reduce setup time,” he continues, “intuitive touchscreen control that simplifies complex mixing tasks, and premium mic pres and effects to ensure the best sound quality, StageScape M20d will deliver great sound in a range of situations.”

• 12 high-performance, digitally controlled, auto-sensing mic/line inputs
• 4 additional auto-sensing line inputs
• 2 digital streaming inputs direct from computer, USB drive or SD card
• Stereo line inputs for integrating MP3 players or other sound sources
• 4 auto-sensing monitor outputs on balanced XLR connectors
• 2 auto-sensing main outputs on balanced XLR connectors
• L6 LINK multi-channel digital networking for integrating L6 LINK-enabled speaker systems

• 7-inch, full-color touchscreen visual mixing environment
• Remote control capability via one or more iPad® devices via an optional USB WiFi adapter

• Multi-channel recording to computer, USB drive or SD card
• Quick-capture recording to internal memory for sound check

• Internal 32-bit floating point audio processing
• Parametric EQs, dynamic EQs, compressors, multi-band compressors, gates, delays, limiters and more
• Multi-band feedback suppression on every mic input
• 4 stereo master effects engines including reverbs, delays and vocal doubler
• Virtually unlimited I/O setups, scenes and channel processing presets

Line 6

Posted by Keith Clark on 08/08 at 07:00 AM
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Innovason Mixes Jekyll & Hyde During Szeged Open-Air Festival

Dóm Square in Szeged, Hungary, (literally, Cathedral Square) is one of the largest squares in Hungary. Every year its 12,000 square metres of open space are transformed into a 4,000-seat auditorium to host the Szeged Open-Air festival. This year's event was mixed on an Innovason Eclipse GT digital console.

Dóm Square in Szeged, Hungary, (literally, Cathedral Square) is one of the largest squares in Hungary. Every year its 12,000 square meters of open space are transformed into a 4,000-seat auditorium to host the Szeged Open-Air festival.

With a tradition stretching back over 75 years, the festival is Hungary’s largest open-air theatre and music event and the most visited summer cultural event of the region.

The unique nature of the festival with its stunning backdrop of the Szeged cathedral is enhanced by a flamboyant program of world premieres and internationally ranked stage shows and concerts, and every year, the stakes are higher in the quest for technological perfection and flawless sound.

This year, ES Audio’s Sandor Elek, who has been managing audio for the festival for the last 12 years via his rental company, Votec, decided to raise the bar even higher and opted for a combination of his own loudspeaker system with Innovason digital mixing consoles and digital microphones from Neumann.

The season opened this year on July 6th with a 3-day run of Wildhorn’s acclaimed Broadway musical of Jekyll & Hyde, controlled and mixed by an Innovason Eclipse GT digital console accompanied by an Innovason Sy80 for the radio mics.

“My goal is to improve the sound year on year,” explained Sandor. “We already made a big difference last year in working with internationally renowned classical engineers such as Carsten Kümmel and Thomas Mundorf whose approach to classical music and in particular their knowledge about the use of condenser microphones measurably improved the sound quality.”

Sandor also deployed his own loudspeaker system that he developed specifically for music and theatre use.

“I couldn’t find anything on the market that fulfilled my requirements for transparency and flexibility for classical music applications, so I built my own!” he said with a grin.

This year, following a number of conversations at ProLight +Sound and, importantly, an introduction to PANDORA, the new panning algorithm on the Eclipse GT console, Sandor decided to take things to the next level and deploy an Eclipse GT for the festival.

He got in touch with the local Innovason partner, Microsound in Budapest to supply the Eclipse GT, as well as an Sy80 for Jekyll and Hyde.

“The PANDORA function is amazing,” stated Sandor. “I didn’t think it was possible to make such a difference to the definition of the stereo image for the audience just by turning a pan pot, but PANDORA makes it possible.

“You can hear a true stereo image that goes from full left to full right and back without any loss of signal for those sat at the extremes. It’s incredible.”

He also decided that given Eclipse’s capacity to control all the parameters of Neumann digital microphones from the control surface, this would be the perfect occasion to try them out and improve sound quality still further by dramatically reducing any noise coming from analog circuitry and cabling.

Sandor therefore arranged for a 40-channel set of digital microphones and four Neumann DMI stageboxes to be run by the Eclipse GT at FoH in the capable hands of tonmeister Carsten Kümmel.

The Sy80 was installed to handle the feeds from the radio microphones of the singers and the choir. This mix was then sent to the Eclipse GT where it was mixed with the orchestra (all using Neumann digital microphones) to provide the main mix which was then diffused by the ES Audio PA system.

“It was a real pleasure to mix this event with the technical set-up we had here,” confirmed Carsten Kümmel.

“What more can you wish for with classical music than a wide open, transparent image that highlights the natural colorations and blending of the different instruments,” he remarked. “The MARS system also proved its worth as I used it constantly to fine-tune the system using the tracks I had recorded during rehearsals.”

Sandor Elek was equally pleased with the results. “I was impressed by the quality of the orchestra sound,” he said. “Carsten’s mix worked really well throughout the 4000- seat auditorium, no matter where you were placed. You could hear everything from everywhere. It’s going to be a real challenge to improve on this next year!”

At the same time Innovason used the opportunity to show this setup to interested clients and business partners.

On Friday over 30 guests from all over Hungary enjoyed a half day of seminars and presentations by Marcel Babazadeh (Innovason), Eric Veres (ES Audio), Imre Selmeci (AudioPartner) and Carsten Kümmel (Tonmeister) in the nearby REÖK palace. The second part of the day took place at the Dom Ter with a hands-on training session and demonstration of the full system using the original tracks taken from the rehearsals recorded by the Eclipse’s onboard MARS multitrack recording system.

Posted by Keith Clark on 08/08 at 06:30 AM
Live SoundNewsVideoConsolesDigitalSound ReinforcementAudioPermalink

Tuesday, August 07, 2012

Macy’s Brasil Campaign Kicks Off With RCF Loudspeakers

Philadelphia based Mitta Sound was tasked with providing sound systems for each of the events during campaign kick-off. A long-time RCF user, owner Terrence McDuffy utilized a variety of RCF loudspeakers to handle the diverse sound requirements of the day.

Macy’s recently kicked off their tribute to Brasil with in-store celebrations at key locations across the county.  The purpose of the “Brasil: A Magical Journey” campaign is to celebrate and bring awareness to the culture and history of Brasil and its people through clothing lines, home furnishings, jewelry and more.

The Center City, Philadelphia store was one of the day-long kick-off locations. Throughout the day shoppers were provided with an assortment of live music and dance performances in various locations throughout the store.

Philadelphia based Mitta Sound was tasked with providing sound systems for each of the events. A long-time RCF user, owner Terrence McDuffy utilized a variety of RCF loudspeakers to handle the diverse sound requirements of the day.

The Macy’s store in Philadelphia features a large, open atrium, which was the focal point of the in-store campaign kick-off.  Alo Brasil, a full, 11-piece band – drum, bass, percussion, guitar, trombone, trumpet, 2 dancers, 2 back-up vocalists and the lead singer – performed throughout the day playing an eclectic mix of Brazilian music.

“Because the stage was located in an open atrium – which was five stories high – it was important to be able to steer the beam to cover the first floor and eliminate potential reverberation in the atrium,” explains McDuffy. “We set up the system to throw 75-100 feet to cover the first floor of the store and it worked extremely well.”

McDuffy specified a left-right RCF TTL11A active column array system for the main stage.

Each stack consisted of a TTL11A-H HF module loaded with a 2.5-inch neodymium compression driver with 1.5-inch exit throat and a TTL11A-B bass frequency module equipped with four 8-inch neodymium woofers with 2.5-inch voice coil. The TTS26-A subwoofer, featuring two high power 15” neodymium woofers, added the extra bass to drive the low end of the system. 

The overall effect was a high definition live sound system ideal for Brazilian music. McDuffy also provided eight stage monitors – four RCF TT25-SMA and four dBTechnologies M12-4 – for the onstage performers.

Although the main action was occurring in the atrium, there was additional entertainment taking place throughout the store. Mitta Sound provided small sound systems for these events as well.

The children’s department had a 20 person dance troupe performing, while the women’s and men’s departments offered their shoppers DJ music.

Once again McDuffy turned to his RCF inventory to provide solutions.

“I put a pair of the TT08 on sticks in both the children’s and women’s departments,” he explains. “Those boxes are our workhorses – we use them for everything and they never disappoint. The DJs were very impressed.”

The compact TT08 is loaded with a 90 x 60 horn, 8-inch neodymium driver and 2.5-inch copper voice coil. With a maximum SPL of 125 dB, the pairs were more than sufficient for the DJ music and the dance troupe in both locations.

Although similar to the women’s department, the DJ in the men’s department was required to cover a much larger area, which led McDuffy to use a pair of RCF TT25A loudspeakers – again on sticks – to disperse the sound to a larger area. 

“We placed the TT25As in the balcony that overlooked the men’s department to cover the entire area,” McDuffy continues. “It sounded fantastic and the folks at Macy’s were extremely pleased.”

TT25A is a full range, versatile two-way active loudspeaker system. The high frequency section is a constant directivity CMD horn loaded to a 1.5” neodymium compression driver with a 3” diaphragm assembly for smooth, controlled dispersion. The low frequency transducer is a 15” neodymium woofer with a 4” voice coil.

“It was a very exciting day at Macy’s – being one of the anchor stores for the launch of this campaign was a very big deal,” McDuffy concludes. “The RCF gear never disappoints. All of the performances throughout the day were extremely well received and sounded terrific.”


Posted by Keith Clark on 08/07 at 02:16 PM
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Drummer Taku Hirano Relies On AKG Microphones For European Percussion Clinics

Tours with AKG mics and relies on them for teaching

Trained in multiple aspects of rhythm, ranging from classical, Afro-Cuban and Brazilian to West African, Middle Eastern, Japanese and Indian, percussionist Taku Hirano, while touring with the Cirque Du Soleil – Michael Jackson: The Immortal world tour, is utilizing his down time to host drum clinics throughout Europe. 

Traveling with a complete AKG microphone set, Hirano also relies on the functionality and clarity of AKG while teaching the fundamentals of his art.

In addition to his current world touring schedule with the MJ Immortal tour, the award-winning Hirano has played with musical legends on tour and in the studio, including Fleetwood Mac, John Mayer, Lindsey Buckingham, Lionel Richie, Michael Bublé, Josh Groban, Stevie Wonder, Whitney Houston, Usher, Jay-Z, LeAnn Rimes and Shakira, among multiple other world-renowned acts.

During his clinics, Hirano’s percussion set is mic’d with two C214’s for overhead capture, a P2 for the Cajon, one P17 for the artist’s signature “Handbale” – the timbale played only by hand, one P3 on a stand for speaking during the clinics, a P4 for the Bongos, and C518M’s clipped on his congas.

During live performances on tour and in the studio, Hirano also utilizes the classic C414 for overheads.

“I’ve been part of clinics in the past where a lot of time, the artist or performer is subject to use whatever equipment the backline companies provide,” Hirano states. “My AKG mics came straight out of the box and provide a pristine sound. 

“With AKG, my sound actually gets better as I’m able to hear every nuance I play.  The gear is working out great as I continue to delve further into playing for the students and prospective percussionists.”

Hirano’s clinics will take place in every major European city the Immortal Tour lands, including London, Copenhagen, Stockholm, Frankfurt, Munich, Vienna, Hamburg, Berlin, Madrid, Moscow, Prague and Barcelona.


Posted by Keith Clark on 08/07 at 01:15 PM
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Peavey Opening West Coast Showroom & Multimedia Dealer Education Center In Hollywood

A high-end retail showcase for Peavey and its affiliated brands

Peavey Electronics will open a West Coast factory showroom, artist relations headquarters and multimedia dealer education center on Sunset Boulevard in Hollywood, CA, on September 1.

A high-end retail showcase for Peavey and its affiliated brands—Composite Acoustics, Trace Elliot, Budda Amplification, Crest Audio, Architectural Acoustics and MediaMatrix—Peavey Hollywood is designed to create awareness for the company’s products and history of innovation.

Peavey Hollywood will present regularly scheduled high-definition webcasts featuring artist interviews and performances, product reviews, and music industry-related tips for musicians. Peavey will also broadcast sales training to its dealers, with live-chat capability to enhance the sessions.

In addition, the location will serve as the West Coast home for the company’s growing roster of artist endorsers.

“Peavey has a rich and exciting legacy in music and audio, and proactive outreach has been key to our success since Hartley Peavey founded the company in 1965,” says Courtland Gray, chief operating officer of Peavey Electronics. “With so many artists living in L.A. or passing through on a daily basis, Peavey Hollywood’s location at the heart of ‘guitar row’ on the Sunset Strip gives us access to artists and marketing opportunities that will serve Peavey retailers and musicians around the world.”

Peavey, which established the first industry-wide dealer training series in 1975, also plans to use Peavey Hollywood as a hands-on educational resource for its dedicated product dealers.

Through an extended-stay intern program, dealers and other affiliates who qualify will be able to send employees to learn about products, sales, marketing, display and promotions first-hand by working with experts. The staff will then be able to take those skills back to their hometown stores.

Interactive displays for Peavey’s amplifier modeling software and iOS recording interfaces will engage musicians on multiple technology platforms, while a Peavey MediaMatrix system will control audio, lighting and video demonstrations simultaneously from an iPad. The space will also allow Peavey to test products, displays and services.

A section of the showroom is dedicated to the history and philosophy of Hartley Peavey, the industry’s longest-running founder, sole owner and CEO, who built his first amplifier as a teenager in 1957 and has earned countless accolades for his achievements in music product innovation and international business, including the Presidential “E Star” Award for excellence in exporting.

“As a technology company that is dedicated to innovation, we have to be able to move quickly and adapt to the needs of consumers,” states Peavey. “Peavey Hollywood enables us to create marketing opportunities that will bring immediate benefits to all of our brands and distribution channels.”

Hartley Peavey will be on hand during the September 1 festivities, which will include artist and celebrity appearances, musical performances and more. Peavey Hollywood is located at 7422 Sunset Boulevard in Los Angeles.

Peavey Electronics

Posted by Keith Clark on 08/07 at 08:38 AM
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In The Studio: Digital Audio 101—The Basics

The key to understanding digital audio is to remember that what’s in the computer isn’t sound – it’s math
This article is provided by the Pro Audio Files.


Digital audio at it’s most fundamental level is a mathematical representation of a continuous sound.

The digital world can get complicated very quickly, so it’s no surprise that a great deal of confusion exists.

The point of this article is to clarify how digital audio works without delving fully into the mathematics, but without skirting any information.

The key to understanding digital audio is to remember that what’s in the computer isn’t sound – it’s math.

What Is Sound?

Sound is the vibration of molecules. Mathematically, sound can accurately be described as a “wave” – meaning it has a peak part (a pushing stage) and a trough part (a pulling stage).

If you have ever seen a graph of a sound wave it’s always represented as a curve of some sort above a 0 axis, followed by a curve below the 0 axis.

What this means is that sound is “periodic.” All sound waves have at least one push and one pull – a positive curve and negative curve. That’s called a cycle. So – fundamental concept – all sound waves contain at least one cycle.

The next important idea is that any periodic function can be mathematically represented by a series of sine waves. In other words, the most complicated sound is really just a large mesh of sinusoidal sound (or pure tones). A voice may be constantly changing in volume and pitch, but at any given moment the sound you are hearing is a part of some collection of pure sine tones.

Lastly, and this part has been debated to a certain extent – people do not hear higher pitches than 22 kHz. So, any tones above 22 kHz are not necessary to record..

So, our main ideas so far are:

—Sound waves are periodic and can therefore be described as a bunch of sine waves,

—Any waves over 22 kHz are not necessary because we can’t hear them.

How To Get From Analog To Digital

Let’s say I’m talking into a microphone. The microphone turns my acoustic voice into a continuous electric current. That electric current travels down a wire into some kind of amplifier then keeps going until it hits an analog to digital converter.

Remember that computers don’t store sound, they store math, so we need something that can turn our analog signal into a series of 1s and 0s. That’s what the converter does. Basically it’s taking very fast snapshots, called samples, and giving each sample a value of amplitude.

This gives us two basic values to plot our points – one is time, and the other is amplitude.

Resolution & Bit Depth

(click to enlarge)

Nothing is continuous inside the digital world – everything is assigned specific mathematical values.

In an analog signal a sound wave will reach it’s peak amplitude – and all values of sound level from 0db to peak db will exist.

In a digital signal, only a designated number of amplitude points exist.

Think of an analog signal as someone going up an escalator – touching all points along the way, while digital is like going up a ladder – you are either on one rung or the next.

Dynamic range versus bit depth (resolution). (click to enlarge)

Let’s say you have a rung at 50, and a rung at 51. Your analog signal might have a value of 50.46 – but it has to be on one rung or the other – so it gets rounded off to rung 50. T

hat means the actual shape of the sound is getting distorted. Since the analog signal is continuous, that means this is constantly happening during the conversion process. It’s called quantization error, and it sounds like weird noise. 

But, let’s add more rungs to the ladder. Let’s say you have a rung at 50, one at 50.2, one at 50.4, one at 50.6, and so on. Your signal coming in at 50.46 is now going to get rounded off to 50.4. This is a notable improvement. It doesn’t get rid of the quantization error, but it reduces it’s impact.

Increasing the bit-depth is essentially like increasing the number of rungs on the ladder. By reducing the quantization error, you push your noise floor down.

(click to enlarge)

Who cares? Well, in modern music we use a LOT of compression. It’s not uncommon to peak limit a sound, compress it, sometimes even a third hit of compression, and then compress and limit the master buss before final print.

Remember that one of the major artifacts of compression is bringing the noise floor up! Suddenly, the very quiet quantization error noise is a bit more audible. This becomes particularly noticeable at the quietest sections of the sound recording – (i.e. fades, reverb tails, and pianissimo playing.)

A higher bit depth recording will allow you to hit your converter with more headroom to spare and without compression to stay well above the noise floor.

Frequency Bandwidth & Sampling Rate

Sampling rate is probably the area of greatest confusion in digital recording. The sample rate is how fast the computer is taking those “snapshots” of sound.

Most people feel that if you take faster snapshots (actually, they’re more like pulses than snapshots, but whatever), you will be capturing an image of the sound that is closer to “continuous.” And therefore more analog. And therefore more better. But this is in fact incorrect.

Remember, the digital world is capturing math, not sound. This gets a little tricky, but bear with me.

Sound is fundamentally a bunch of sine waves. All you need is at least three point values to determine a sine wave function that crosses all three. Two will still leave some ambiguity – but three – there’s only one curve that will work. As long as your sample rate is catching points fast enough you will grab enough data to recreate the sine waves during playback.

In other words, the sample rate has to be more than twice as fast as the speed of the sine wave in order to catch it. If we don’t hear more than 22 kHz, or sine waves that cycle 22,000 times a second, we only need to capture snapshots more than 44,000 times a second. Hence the common sample rate: 44.1 kHz.

But wait, you say! What if the function between those three points is not a sine wave. What if the function is some crazy looking shape and it just so happens that your A/D only caught three that made it look like a sine wave?

Well, remember that if it is some crazy function, it’s really just a further combination of sine waves. If those sine waves are within the audible realm they will be caught because the samples are being grabbed fast enough. If they are too fast for the our sample rate it’s OK, because we can’t hear them.

Remember, it’s not sound, it’s math. Once the data is in, the computer will recreate a smooth continuous curve for playback, not a really fast series of samples. It doesn’t matter if you have three points or 300 along the sine curve – it’ll still come out sounding exactly the same.

So what’s up with 88.2, 96, and 192 samples/second rates?

Well, first, it’s still somewhat shaky ground as to whether or not we truly don’t perceive sound waves that are over 22 kHz.

Secondly, our A/D uses a band-limiter at the edge of 1/2 our sampling rate. At 44.1, the A/D cuts off frequencies higher than 22 kHz. If not handled properly, this can cause a distortion called “aliasing” that effects lower frequencies.

In addition, certain software plug-ins, particularly equalizers suffer from inter-modular phase distortion (yikes) in the upper frequencies. The reason being, phase distortion is a natural side effect of equalization – it occurs at the edges of the effected bands. If you are band-limited to 22 kHz and do a high end boost, the high end brickwall stops at 22 kHz.

Instead of the phase distortion occurring gradually over the sloping edge of your band, it occurs all at once in the same place. This is a subject for another article, but ultimately this leaves a more audible “cheapening” of the sound.

Theoretically a 16-bit recording at 44.1 smpl/sec will have the same fidelity as a 24-bit recording at 192. But in practicality, you will have clearer fades, clearer reverb tails, smoother high end, and less aliasing working at higher bit depths and sample rates.

The whole digital thing can be very complicated – and in fact this is only touching the surface. Hopefully this article helped to clarify things. Now go cut some records!

Matthew Weiss is the head engineer for Studio E, located in Philadelphia. Recent credits include Ronnie Spector, Uri Caine, Royce Da 5’9” and Philadelphia Slick.

Be sure to visit the Pro Audio Files for more great recording content. To comment or ask questions about this article go here.

Posted by Keith Clark on 08/07 at 07:29 AM
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