Thursday, August 09, 2012
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)
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.
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
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 Sound •
Sound Reinforcement •
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.”
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.
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.
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
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
Lake LM26 Enhances System Performance At Tennessee Church
Chosen based on the wide range of control it offered over equalization, limiting and delay for the mains
Featuring a full electric praise band regularly augmented by strings and other acoustic instruments, services at the new permanent facility of Living Hope Church in Piperton, TN are extremely contemporary.
Correspondingly, when Memphis-based designer/integrator Elite Multimedia set out to provide a highly flexible system to meet the needs of Living Hope’s growing congregation, they chose a Lake LM 26 signal processor based on the wide range of control it offered over equalization, limiting and delay for the mains.
“There’s a substantial difference in the quality of this system compared to the one they used when they operated as a portable church,” explains Wade Russell, lead systems integrator at Elite. “One of the main reasons we selected the Lake LM 26 was the fact that it did 96k AES.
“We’re running AES from a Midas PRO2C console to the LM 26 and the amps,” he continues, “and integrating the Midas and Lake processing was key to creating a very dynamic system that was powerful and had the depth and stereo imaging they were looking for. The variety of customizable filters within the Lake LM 26 definitely helped us achieve that.”
The response from both congregants and other engineers has been overwhelmingly positive. “When we made the decision to move into a building of our own, we approached everything with our sound system by saying ‘let’s go ahead and put the best gear in and make a big impact’ and Lake was simply the best,” says Living Hope audio technician Tim Johnson. “Not only does it provide a great signal path, but with the built in filtering we were able to get the sound image we were looking for so wherever people sit they have the same experience.
“People who don’t know a lot about audio know that they love what they hear, and people who do can’t get over the how crystal clear the sound is. You’re hearing true, clean signal at 96k and the detail of every instrument. Literally, you feel like you’re up on stage with the musicians.”
Living Hope Church Both Johnson and Living Hope’s worship pastor, David Lewis, come from mega church backgrounds, and both wanted to bring the same level of sound quality to this smaller venue.
“But in actuality,” Johnson adds, “with the Lake LM 26, this far surpasses the quality we had in those larger churches.” There is one potential problem, however. “After we listened to the system, David said, ‘I don’t know if this is a good thing or a bad thing. My band is really going to have to practice hard because if we don’t sound good, we certainly can’t blame it on the system.’”
Monday, August 06, 2012
Extron Now Shipping DisplayPort Switcher And Distribution Amplifier
Extron Electronics is pleased to announce the immediate availability of the SW2 DP two input DisplayPort switcher, and the DP DA2 two output DisplayPort distribution amplifier. Both units are HDCP compliant and support data rates up to 10.8 Gbps and computer resolutions up to 2560x1600 @ 60 Hz, including HDTV 1080p/60.
Extron Elextronics is pleased to announce the immediate availability of the SW2 DP two input DisplayPort switcher, and the DP DA2 two output DisplayPort distribution amplifier.
They are HDCP compliant, and support data rates up to 10.8 Gbps and computer resolutions up to 2560x1600 @ 60 Hz, including HDTV 1080p/60.
The SW2 DP and DP DA2 feature EDID Minder, an Extron exclusive technology, which maintains continuous EDID communication between connected devices for reliable video content display.
The DP DA2 also features Key Minder, another Extron exclusive technology that continuously authenticates HDCP encryption between all devices, ensuring the simultaneous distribution of source content to both displays.
Additionally, dual-mode support on the DP DA2 allows source signals to be distributed to multiple HDMI, DVI, or VGA display devices with appropriate adapters.
“The SW2 DP and DP DA2 are among the first DisplayPort switcher and distribution amplifier products for the pro AV industry,” says Casey Hall, Vice President of Sales and Marketing for Extron. “These products enable AV system designers to take advantage of DisplayPort technology, delivering exceptional performance in a wide range of AV environments.”
The addition of the SW2 DP and DP DA2 to Extron’s expanding line of DisplayPort products allows the integration of this technology into a wide variety of AV applications, including conference areas, houses of worship, and rental and staging environments.
The SW2 DP and DP DA2 are ideal for AV systems that require reliable switching and distribution of DisplayPort signals.
Posted by Keith Clark on 08/06 at 10:06 AM
D.A.S. Teams Up With Marti Audio For Roger Hodgson Concert
Roger Hodgson, one of the founding members and voice of the legendary English band Supertramp, offered a total of 16 concerts aross South America and Argentina where Hodgson played to a full-house at the city’s Salón Metropolitano, D.A.S.
Aero Series 2 sound systems provided top quality sound for the event.
Roger Hodgson, one of the founding members and voice of the legendary English band Supertramp, recently completed the first leg of his “Breakfast in America Tour 2012” in South America.
A total of 16 concerts were offered across the continent, and in Rosario, Argentina, where Hodgson played to a full-house at the city’s Salón Metropolitano, D.A.S. Aero Series 2 sound systems provided top quality sound for the event.
Local company Marti Audio, with Ariel Marti at the helm, rolled out the big guns for the concert. A comprehensive D.A.S. sound system was set up, which included Aero Series 2 line arrays, specifically 12 units of the large D.A.S. Aero 50 for the PA, set up in two arrays of six units each on either side of the stage in a traditional left-right configuration.
Another 12 units of the powerful D.A.S. LX-218R subwoofers were brought in to reinforce the low frequencies at the concert. The units were ground stacked on opposite ends in front of the stage in two groups of six units each. To guarantee proper sound coverage for the first rows of the audience, a front-fill comprised of eight powered D.A.S. Aero 8A systems were used.
The flawless sound quality gave concert goers the chance to appreciate the vocal nuances of Hodgson and the talent of his band. Fans were delighted to hear Hodgson’s new songs along with Supertramp’s biggest hits, including “The Logical Song”, “Give a Little Bit”, and “Dreamer”. Marti Audio’s experience in setting up sound systems at this same venue proved invaluable for the concert´s success.
Church Sound: Clearing The Stage Quickly For Smooth Transitions
Plan ahead; it really does make life easier!
VBS. Those three letters can strike fear into the heart of the most seasoned tech director.
In our case for 2012 Vacation Bible School, we had a lot to deal with.
Each day consisted of two main sessions—an early morning and late morning—and each session had both live music (in some cases several sets)and a skit. In between, we rehearsed music and drama.
Since the drama and band needed to occupy the same piece of stage real estate, we needed a way to quickly strike the band (or at least part of it). Here’s how we did it.
First, we used Steeldeck stage platforms on wheels for the entire band. Steeldeck is a little pricey, but is rock-solid and the wheels roll like butter on a non-stick skillet.
The band consisted of an 8- by 8-foot deck for drums (raised up to 2 feet high), and three 4- by 8-foot platforms (1-foot high) for keys/guitar, vocals, and bass/acoustic. All of the platforms were on wheels so we could move them quickly.
Taping the cables down meant they didn’t tangle up the band members, or get caught under the wheels. (click to enlarge)
But that’s only half the problem. The other half is cable management. And that’s where we got really creative.
Everything was hard-wired, so we cleanly wired all the mics and DIs to the deck. All cables were run neatly and gaffed down so they wouldn’t go anywhere.
We set the lengths so the male ends made it just over the rear corner of each deck so they could mate with a snake, and coordinated our snakes to cables so we didn’t have a lot of extra channels floating up there.
We also labeled both sides of the connection with gaff tape and silver Sharpie using a simple letter code: A connects to A, B to B and so on. Simple is better.
On stage right, we used a 4-channel snake to mate with the three lines (stereo keys, electric guitar, which used an SGI). The center deck had four lines—three mics and an SGI for the other electric. Stage left held the bass and acoustic DIs.
Even if you don’t know the alphabet, you can still match the shapes… (click to enlarge)
We ran a 6-channel snake up the center on the left side to connect both the center and stage left decks, and ran short XLR cables over from the bass/acoustic deck (we also stacked the DIs on the right side of the deck to keep the runs short).
Our drums are normally connected via a Whirlwind 12-channel snake with a MASS connector, so that was easy to disconnect if we needed to. We also ran our 4-channel snake back to four unused channels of our drum snake so we only had two main lines running out to all four platforms.
The process of moving decks was simple and took only two people; my trusty assistant TD Jon, and a junior high volunteer. One of them unlocked the wheels while the other unplugged the cables.
The bass/acoustic platform connected to the center snake with two lines and a power cord. (click to enlarge)
After that, it rolled right off stage. It took less than a minute to get a deck off stage. Because of the way the drama was set up, we never had to move more than one deck at a time, which made transitions really fast. Even re-setting from band to drama rehearsal several times a day was no big deal.
We probably spent the better part of an hour working out cable runs, patches and how we were going to implement this, but once it was done, it saved us a ton of time.
In fact, we had the whole stage set so that we could clear all the band platforms in under three minutes if need be.
Plan ahead; it really does make life easier!
Mike Sessler is the Technical Director at Coast Hills Community Church in Aliso Viejo, CA. He has been involved in live production for over 20 years and is the author of the blog, Church Tech Arts . He also hosts a weekly podcast called Church Tech Weekly on the TechArtsNetwork.
Friday, August 03, 2012
Church Sound: Using Consoles For Double Duty On House & Monitors
How can a church engineer provide monitoring without substantial financial outlay?
For any church that offers a contemporary service involving a praise ensemble, and even those presenting more traditional services that involve a substantial choir, it may be necessary to provide monitoring for the musicians and singers so that they may hear themselves and their colleagues.
Assuming that the sound system supports more than simple voice reinforcement and includes sufficient microphones, amplification, loudspeakers and a mixing console, how can a church engineer provide monitoring without substantial financial outlay?
For those unable to spend thousands of dollars on a dedicated monitor desk, or one of the dual-purpose consoles available from several manufacturers, the good news is that it is possible for the majority of front of house consoles to do double-duty.
There are a number of different ways that monitors may be provided from the house console, including using the auxiliary sends or using the matrix outputs.
But while the matrix outputs typically source the subgroups and therefore individual inputs that have been grouped together, using the aux sends allows the level of each input source to be adjusted individually for each separate monitor mix.
Aux send controls have several functions. They can be used to control the levels of the signal going to effects units, to drive subwoofers, as feeds to an external recording device, and to create separate onstage monitor mixes.
The simplest and best method for creating monitor mixes from the house console is to use aux sends selected to be pre-fader and preferably pre-EQ. This means that any changes in level or EQ in the main sound system will not affect the monitor feeds since they are sourced, and their levels controlled, before each input channel EQ section and fader.
Allowing those changes to also affect the monitors would be undesirable and could be disastrous. Rolling off a little of the high frequencies on the drum kit in the main system will not necessarily improve the sound in the monitor loudspeakers. And raising the level of a vocal in the house system will increase the level in the monitors and could lead to feedback.
Whether or not, and exactly how, a console can switch the aux sends between pre- and post-fader and pre- and post-EQ depends entirely on the desk model and manufacturer. Some consoles switch pre/post-fader in pairs of aux sends, or banks of four, or even globally—and some not at all.
Pre/post-EQ switching may be similar, and on some consoles is achieved via individual switches on the input circuit boards, not the console faceplate. Consult the console operation manual or the manufacturer if in doubt. If you intend to purchase a new console for the purpose of mixing both front of house and monitors then this is something to keep in mind.
The total number of aux sends available and that may be switched pre-fade/pre-EQ will dictate how many individual monitor feeds may be set up. Keep in mind that you may wish to retain some of the auxes for effects sends in the main system, to drive subwoofers or to feed a recorder.
If the console design dictates that all the aux sends must be set to pre-fader then the effects sends will be, too. This means that unless you run your input faders around ‘0’ then your input channels will be very effects-heavy. The more adventurous and technically proficient may wish to modify their FOH console to meet the challenge of also mixing monitors.
Sending a monitor mix to the stage without EQ is not perfect, but is preferable to a post-EQ mix that changes during the service as the FOH engineer makes adjustments to the main sound system. In order to EQ each mix it is possible to insert an outboard equalizer into each monitor feed.
A minimum of two monitor feeds is probably desirable. How the monitor speakers are physically positioned depends on your circumstances—perhaps front and back or both along the front of the stage.
Which aux sends are used to create these mixes is also a matter of choice. Those farthest from the operator perhaps make most sense as the engineer will not be constantly reaching for the controls once the monitors are set correctly, so the controls do not need to be easily accessible.
Taking the instance of two monitor mixes, on each input channel Aux Send #1 (as an example) will control the amount of that channel’s signal that is sent to monitor mix one and Aux Send #2 likewise to the second mix. Aux Send Masters 1 and 2 will control the overall output level of each monitor mix.
So, if the overall balance is good but the performer just needs the level to the loudspeaker adjusting up or down, modify the level using the aux master level control. But if the performer needs more level of his or her vocal, for example, adjust that vocal input channel’s relevant aux send to their specific monitor mix.
In truth, the engineer does not “mix” monitors, whether from a dedicated monitor desk or the house console. Prior to the service, each monitor channel is tuned and equalized, console input and output levels are set, and everything is then best left as it is, unless a performer requests a change.
Any musician will tell you that it can be unsettling for the monitor mix to change during a performance just because the monitor engineer has decided that he or she can somehow “improve” it. However, anyone controlling monitor sends needs to be ever vigilant toward performers and quickly make any changes that are requested.
There are a number of basic rules to keep in mind. Above all, when making adjustments, keep in mind that feedback is to be avoided at all costs. And never take for granted that the level at which a musician plays during setup, rehearsal or sound check will be the same level at which they will play during the service.
Assuming that there are sufficient aux sends available, performers within the same group will each prefer very different levels and mixes to be sent to their individual monitor speakers or in-ear setups.
Those having little previous exposure to monitors often ask for a mix of everything, which is to be avoided. The goal, after all, is to keep the stage volume as low as possible.
Think like a performer and you will learn to anticipate their needs. What they need to hear to perform well is very different to the audience mix. Vocalists need to hear themselves and those with whom they are harmonizing, and everyone will probably want a mix of the vocals.
Rhythm sections will also need to hear kick drum and snare. Instruments with their own amplification don’t need to be in the monitors as they are most likely loud enough onstage already.
Monitor volumes can be a problem in establishing a good mix in the sanctuary if the main sound system cannot overcome the volume from the monitor loudspeakers. High monitor levels can also find their way into the house system via the microphones.
Monitor speaker levels and positioning must therefore be set optimally to ensure a good house sound. Of course, in-ear monitor systems reduce these problems.
Once levels are set, be wary of making any further adjustments or adding more of anything to the mix. Once a satisfactory onstage level is achieved that does not interfere with your ability to mix the house sound you can maintain that level by making any further adjustments subtractive. The result is the same, but onstage levels will not get out of control.
For example, if a singer needs to hear more of his or her vocal, decreasing the level of the other elements in that mix will raise the relative level of the vocal in that monitor mix, satisfying their requirement without increasing the overall level.
The unwanted alternative as each performer asks for more of this or that in their mixes is a steadily increasing onstage volume.
PreSonuSphere 2012 Second Annual User’s Conference Coming Up In September
Expanded to deliver more seminars, speakers, information, music, networking opportunities, and fun
PreSonus will be hosting the PreSonuSphere 2012 Second Annual User’s Conference in Baton Rouge, LA this coming September 28 and 29.
Following up on the success of PreSonuSphere 2011, this two-day user’s conference has been expanded to deliver more seminars, speakers, information, music, networking opportunities, and fun.
Focusing on tips, techniques, and applications for PreSonus Studio One 2 DAW software and StudioLive Series digital mixers, PreSonuSphere 2012 will include guest lectures and demos by noted professionals such as EM/Harmony Central editorial director Craig Anderton, front-of-house engineer Ace Baker, worship-team trainer Doug Gould, Grammy-nominated producer and bassist Brent Milligan, studio designer John Storyk, front of house lead engineer (Kenny Chesney tour) and audio consultant John Mills, live sound “wizard” and forum favorite Jon Taylor, microphone expert Steve Savanyu, and veteran journalist and engineer Mike Rivers.
In addition, key PreSonus personnel will be on hand to present seminars and talk with attendees, including president/co-founder/chief strategy officer Jim Odom and chief technology officer Bob Tudor, as well as Studio One development team leaders Wolfgang Kundrus and Matthias Juwan. Get applications tips and a peek into the future from the PreSonus staff, and much more!
PreSonuSphere 2012 will be held at the Shaw Center for the Arts in historic downtown Baton Rouge. (They use the StudioLive 24.4.2.)
The event actually kicks off on Thursday evening, September 27, with a Cajun dinner party at the Shaw Center’s fifth-floor outdoor terrace. Also on Thursday night, there will be live music by Papa Grows Funk, featuring special guest Larry Braggs, lead singer of Tower of Power. Ace Baker will handle the mixing chores. Friday is blues night with Baton Rouge favorite the Chris LeBlanc Band, mixed by John Mills.
The cost to attend PreSonuSphere 2012 is one day for $30 or both days for $50. For $99, attendees get all of the above, plus Saturday evening dinner on the terrace, swag, and video DVDs of all PreSonuSphere proceedings. The swag and DVD are also available separately for those who can’t come to Baton Rouge for the conference.
Special room rates have been arranged for attendees at the Hotel Indigo Baton Rouge Downtown Riverfront, which is walking distance from great restaurants, historical sites, museums, and the Mississippi River. Don’t miss it!
Go here for more information and to sign up for the PreSonuSphere 2012.
Thursday, August 02, 2012
Understanding, Contrasting Lossless & Lossy Audio
What are they, and what's the difference
Lossless audio refers to recording compression techniques that allow all recorded sounds to be present in their original form.
Essentially, it includes all the dynamic range from super-quiet to very loud, all the frequencies from subsonic to harmonic frequencies, stored in one of several digital codec formats.
Lossless audio recording is typically used for making master or archival recordings.
Contrasting to lossless is “lossy” audio, which is remarkably prevalent in the consumer audio world.
Examples of lossy audio include MP3 files on your phone or audio player, as well as audio carried by streaming media sources from the Internet, satellite and cable radio formats, selected video DVD and broadcast formats.
The primary reason lossy audio formats such as MP3 are deployed is storage capacity; if a standard CD carries one hour of material, and a lossless audio recording of the same capacity carries two hours of material, the seven hours of capacity using MP3 in the same space affords lots more potential material to be accessed in portable playback devices.
At the advent of portable players storage costs were high and the perception was that the market for MP3 and similar formats was not as critical as lossless audio formats.
Lossy audio is predicated on psychoacoustics, specifically the idea that human hearing, when presented with a loud sound and a quieter sound, will hear the loud sound.
Using algorithms, the content of the original recording is analyzed and quiet parts are eliminated from the recording. This technique essentially provides the increased efficiency of recorded times that we discussed.
The problem with that idea, of course, is that comparing an MP3 (or other) recording to an original master recording clearly will demonstrate what happens during the compression process. Dynamic range of the music, from quiet to loud, will be diminished. Warmth of instruments and vocals is reduced. Ambience, in the case of the recording hall, is compressed.
To the keen listener, the lossy format just doesn’t sound right. The music is there but it’s not all there, so to speak, and does not have the same immediacy and impact that the original or lossless format offers.
Lossless audio formats will give users the ability to record music and maintain all the goodness present in the original. Certainly, many versions of lossless formats exist, including Apple Lossless, Microsoft’s MPEG4 ALS, and Free Lossless Audio Codec (FLAC).
In applications where the audience and the reproduction equipment can support it, lossless recording makes increasing sense, especially as the cost of recording storage space diminishes.
That being said, there are many applications where lossy codecs should be deployed as it clearly increases the speed of downloading information off the Internet. In some installations, the noise floor is such that higher-resolution playback simply is not suitable and a compressed audio signal will handle the job just fine.
I do think it’s interesting that the mid-level audio market on the consumer side appears to be moving away somewhat from highest fidelity playback of two-channel audio and emphasizing surround-sound formats.
Is it because new audiences haven’t heard real high-quality audio playback?
Fred Harding handles technical sales and design at Capitol Sales. This article is courtesy of Commercial Integrator.
Posted by Keith Clark on 08/02 at 01:05 PM