Analog

Tuesday, May 27, 2014

XILS-lab Breathes Life Into Prototyped EMS VCS4 Analog Matrix Modular Synthesizer

Music software company XILS-lab is proud to announce availability of XILS 4.

Music software company XILS-lab is proud to announce availability of XILS 4 — an authentic software emulation of the legendary VCS4, a ‘dual VCS3’ analogue matrix modular synthesizer prototyped by EMS back in 1969, but never commercially released.

EMS (Electronic Music Studios), a British company founded by the pioneering Peter Zinovieff, made musical history in 1969 with its introduction of the VCS3, the first portable synthesizer commercially available anywhere in the world.

Its innovative modular matrix-based patch-board dispensed with the telephone exchange-like cabling of other (much larger) modular systems in favor of making connections with (removable) colored pins, so it could be comfortably housed in a small wooden (solid afrormosia) cabinet.

The history of the EMS VCS3 is well documented with XILS-lab later playing its part in resurrecting its still-sought-after sound with its cost-conscious and award-winning XILS 3 software emulation for Mac (OS X 10.4 and above) and PC (Windows 7, Vista, and XP) proving popular with both first-time buyers and also seasoned synth explorers including Richard Devine, Tim Blake (Hawkwind), and even Peter Zinovieff.

The 1969-vintage VCS4 was EMS designer David Cockerell’s so-called ‘Live Performance Module’, comprising two VCS3s side by side, together with a five-octave keyboard, a mixer, and a signal-processing unit, all housed in a single wooden cabinet. Only one prototype was ever produced.

By being based on two intricate and interacting VCS3 (‘Synthi’) cores — XILS-lab’s XILS 4 favourably emulates EMS’ VCS4. Indeed, those two cores can be set to work side by side or operate in serial (with one feeding the other).

Each and every module on one side can be used to modulate or feed anything on the other side with stunning sound possibilities plus weird and wonderful effects readily available in abundance as a direct result.

That said, XILS 4 shows its true 21st Century colors by also allowing amount settings to be individually applied to each patch ‘pin’. Providing patch-board power par excellence to an already special soft synth shows that there is clearly so much more to XILS 4 than solely emulating vintage hardware — rare as the vintage hardware in question clearly is.

Little wonder, then, that XILS 4 is billed by its creator as being the ultimate analog matrix modular synthesizer.

XILS-lab has sought to take things several steps further still by coupling those cores with the SEQUENCER 256 module, inspired by EMS’ trailblazing Synthi Sequencer 256 namesake.

Needless to say, this three-layer sequencer with analog-to-digital and digital-to-analog converters to enable digital processing of control voltages to drive multiple analogue synthesizers or multiple parameters with storage of up to 256 ‘events’ was well ahead of its time when released in 1971.

Here in the virtual world, XILS-lab has well and truly transported it to the present day with three independent layers, slew rates, and recording modes, together with added abilities like sequencer layers acting as modulation sources in a dedicated SEQ MATRIX — matchless, even by today’s most sophisticated DAW standards!

The addition of a second ‘pin matrix’, two additional envelopes, an LFO, comprehensive SAMPLE AND HOLD module, and VOLTAGE PROCESSOR, plus several new input modules — including GATE, ENVELOPE FOLLOWER, and PITCH TRACKER — means that there are hundreds of additional connections available to the discerning synthesist set on exploring XILS 4 to the full.

The fact that there are over 1,140 possible connections per patch makes for a literal lifetime of programming possibilities that will surely far outlast the host computer concerned, though those in need of a helping hand have easy access to almost 700 professionally-programmed presets from world-renowned sound designers, including the complete XILS 3 factory library and over 350 presets specifically designed for XILS 4.

A selection of tutorial-style patches are also available to help users seeking to take their first tentative steps towards scaling the heady heights of this mountainous modular monster of a soft synth.

Simply put, with a whole host of modules and associated far-reaching functionality — for starters, 12 aliasing-free oscillators, grouped in six pairs with wave-shaping and hard sync — yesteryear’s impossibilities have become today’s possibilities with XILS 4… a great step forward for modular synthesis software.

XILS 4 is available to purchase as an eLicenser or iLok copy-protected virtual instrument and effects plug-in for an introductory discounted price of €149.00 EUR (rising to €179.00 EUR on June 17, 2014) from the XILS-lab web store. Note that this time-limited offer also includes the XILISTICS sound bank with 160-plus presets worth €25.00 EUR, plus a free USB-eLicenser copy-protection dongle.

XILS 3 owners can upgrade to XILS 4 for an introductory price of €29.00 EUR (rising to €49.00 EUR on June 17, 2014).

XILS 4 can be directly downloaded as a 32- and 64-bit-compatible virtual instrument and effects plug-in for Mac (AAX, AU, RTAS, VST) and Windows (AAX, RTAS, VST) from here: http://www.xils-lab.com/pages/XILS4_Download.html

Check out several informative XILS 4 tutorial videos at http://www.xils-lab.com/pages/XILS4_Videos.html.
XILS Lab

{extended}
Posted by Julie Clark on 05/27 at 01:07 PM
ProductionNewsAnalogManufacturerSoftwareStudioPermalink

Audient’s New Distributor Helps Boost Brand In BeNeLux

British manufacturer, Audient appoints M Works as Benelux distributor for Audient.

British manufacturer, Audient appoints M Works as Benelux distributor for Audient.

With immediate effect, dealers in the Benelux territory can get the full range of Audient products from M Works, from the flagship ASP8024 right through to the latest additions to the range: USB interface - iD22 and 8-channel mic pre and ADC - ASP880.

Sales Manager Okke van Dijk is very pleased at the agreement. “Audient is a great brand with passion for designing excellent sounding products. With their expertise and deep understanding of demands in the pro audio market, the people at Audient know how to create the right quality and product mix,” he says.

“High quality comes as standard,” confirms Luke Baldry, Sales and Marketing Director at Audient, highlighting the fact that “…all products feature the renowned console mic pres designed by the company’s co-founder David Dearden.” He adds, “I’m delighted that we have Okke and the M Works team on board to ensure Audient quality reaches this important market.”

Celebrating its 10 year anniversary this year, M Works is one of the leading independent distribution companies in the Benelux, making it easier than ever for engineers, producers and musicians to get hold of Audient products, focus on the music and enhance their creativity. “We at M Works are very excited to welcome Audient to our portfolio, as it is complementary to our current line-up of products and we are looking forward to a long-term business adventure with Audient,” comments van Dijk.

Audient

{extended}
Posted by Julie Clark on 05/27 at 12:58 PM
RecordingNewsAnalogBusinessConsolesManufacturerSound ReinforcementPermalink

Friday, May 23, 2014

Clear Path: Keyboards In The Electronic Realm

Electronic keyboards, the start of it all. Right from the beginning of modern concert sound, DI boxes have played an essential role in getting the sound from the stage to the PA system.

Probably the most iconic “direct” instrument of all was the Fender Rhodes. Harold Rhodes started developing the idea as far back as the 1950s, but it was in 1970 that the Rhodes Stage piano took the concert stage bringing the first “portable” keyboard to market.

The original Rhodes piano tone was created by a piano-like hammer striking a “tine” that would vibrate up and down in front of a magnet to create the tone—very much the same way an electric guitar string vibrates atop a magnetic pickup. One would adjust the tone by changing the “tine-to-magnet” relationship.

And like an electric guitar, the output from the suitcase was not amplified (or buffered) in any way. So the output from the piano was generally sent to a guitar amp where it was mic’d.

Some years later, the first active DI boxes came round. They didn’t load the Rhodes pickups, which made it practical to send the “direct” sound to the PA system and monitors.

But something happened. That something was Keith Emerson and Rick Wakeman, and the Moog synthesizer, which found its way out of the electronic music department to the stage. These guys no longer had one or two keyboards—they had racks of them! 

An early Fender Rhodes, the one that started it all.

The Arp 2600 and String Ensemble, Oberheim, Korg, the venerable Sequential Circuits Prophet 5 - it was an analog explosion. Everyone had a Rhodes (or Wurlitzer) and a bunch of synths. 

Fast forward to 1981, and Yamaha introduced the DX7, which would go on to become one of the most successful keyboards ever. It brought along something totally new: frequency modulated digital technology. Now you could get a bell-like Rhodes sound without the weight.

The world then changed again with the E-MU and the Akai S900 digital sampler. All of a sudden, we had complete orchestration, real sounding piano samples, and digitally sampled drum tracks were everywhere. There was no going back.

Relatively Powerful
Today, pretty much all keyboards and drum machines are digital, and can basically be thought of as keyboard controlled CD players. And like a CD player, the output from a digital synthesizer is relatively powerful when compared to an electric guitar or an old Rhodes piano.

Because they’re so “loud,” they needed headroom to operate, meaning that the old active DI box that may have been a boon to the low-output Rhodes piano can no longer keep up.

The headroom is limited by the internal battery or limited by the low current afforded by phantom power. To make matters worse, unlike a CD that is processed and compressed before it is mass produced, digital samplers are raw. They can generate huge transients that will overload most active DI boxes, and end up distorting horribly.

The problem is further exacerbated with digital pianos. These full-range devices are not only very dynamic, they have a frequency range that starts way down low and goes up forever.

To handle modern keyboards, there are two choices:

1) Send the keyboards into a mixing console where the internal rail voltage is sufficiently ample that it is able to handle the range.

2) Send the signal to a passive direct box where the headroom is not limited by the current afforded to them. Passive DI boxes are different- they use transformers.

Phantom Of The Power
Replace the diesel engine inside a dump truck with a 4-cylinder car engine and fill the truck with gravel. What will happen? Nothing. The engine will be unable to handle the load.

The same applies to phantom power. Folks tend to “believe” that if it’s active, it must be good. But the truth is, phantom power was never designed to power direct boxes.

As noted here, phantom was invented by Mr. Neumann as a means to charge the capsules on his microphones. He needed a lot of voltage (48 volts) and very little current (5 milliamps).

A quality preamp requires +/- 16 volts (32-volt swing) and about 50 milliamps of current. With 1/10th the current, it’s like trying to run a dump truck with a motorcycle engine.

Passive direct boxes are not power limited. They’re old-fashioned devices that basically combine a couple rolls of wire (coils) with a chunk of metal (the core).

A DI has the task of converting a high-impedance signal to a low-impedance balanced signal where it can be managed by the mic splitter and mixing console’s preamp.

With keyboards, current enters the transformer where the conversion occurs. But instead of overloading. like an active circuit, transformers distort gradually. More precisely, they don’t so much distort as saturate.

We often say that transformers sound “vintage” or have a limiting quality about them. This is because good quality transformers generate warm sounding even order harmonics, or what is commonly known as a warm Bessel Curve.

Thus the reason highly dynamic buffered signals like digital keyboards sound great when they are used with a passive direct box.

Multiple Choices
Mono, stereo or multichannel - which DI is best? It depends upon your point of view.

If audience members are sitting right in front of the left PA loudspeaker, they will be unable to hear what is coming out of the right PA loudspeaker. Will they benefit from stereo? Probably not.

Stereo may sound great in the practice room or be invigorating on stage, but in most live venues, it is rarely enjoyed by all. The advantage that a stereo DI brings is capturing the stereo sample without having to reprogram the synth.

And if you do decide to have a stereo rig on stage, a stereo DI allows the house engineer to mix both channels and pan them stereo (if beneficial).

A stereo DI is often used at the output of a keyboard mixer. Here’s why: on a stage, all of the microphones go to a mic splitter before the signal is sent to the house mix position. Mic splitters are designed to handle mic levels, typically around -50 dB. A keyboard produces a -10 dB signal while a mixer can produce +4 dB or more.

A look at one reason why a passive DI can be a better choice for electronic keyboards. Courtesy of Radial Engineering.

This excessive level will cause the mic splitter to overload. The pad on a direct box lowers the output so that it matches that of a microphone and protects the mic splitter from being overloaded by the mixer. Multichannel direct boxes bring forth the added advantage of independent control over each instrument.

Here’s the deal: when you mix the sound so that it is comfortable on stage, it may in fact not be ideal for front of house. In other words, you may find that you need extra jam to hear your piano on stage, and have the string synth pushed back in the mix.

But in the arena, the piano-to-string volume ratio may not sit well with the rest of the band. What happens? The keyboard mix gets pushed back.

When you have the luxury of sending independent keyboard signals to front of house, the mix engineer is allowed to orchestrate. With more control, the engineer can decide how much piano fits and if the strings are too loud, can simply back them off.

Who would have ever thought that a DI would have so many twists and turns, especially after being around for 40 years!

Peter Janis is president of Radial Engineering. In 1982, he was hired by CBS Fender as new product director for what would eventually become Fender Canada, and spent time at the ARP factory in Massachusetts learning to program the advanced Chroma polyphonic synthesizers. He met Harold Rhodes and spent several years servicing Rhodes pianos before they were eventually discontinued, and during that time, also added the Akai product range to the Fender sales portfolio and developed many of the Akai’s early samples.

 

{extended}
Posted by Keith Clark on 05/23 at 05:42 AM
Live SoundFeatureBlogStudy HallAnalogDigitalInterconnectSignalSound ReinforcementStagePermalink

Tuesday, May 20, 2014

Multi-Plantinum Rock Group Live Chooses SSL Duality

Think Loud studios installs SSL Duality analog console.

Founding members of the multi-platinum band LIVE recently established Think Loud Studios in their hometown of York, Pa.

When it came time to install a console in the world-class facility’s expansive Studio A, they opted for a 48-channel Solid State Logic Duality analog console/controller.

The extensive experience that the group and their engineers have with classic SSL analog consoles led them directly to Duality.

Known for classic albums including Throwing Copper, Secret Samadhi and The Distance to Here, the members of LIVE have returned to the community in which they grew up and launched the studio.

In addition to the band’s own projects, Think Loud Studios serves the artists on its record label, Think Loud Entertainment, and friends of the group, including the band Everclear.

The vaulted ceilings and abundant natural light of the 53,000-square-foot building’s fourth floor made it a natural setting for Think Loud, which was designed by Horacio Malvicino, says bassist Patrick Dahlheimer.

“This is inspirational,” he remembers thinking. “This is going to be the studio that we always wanted to build and is driven to be songwriter and musician friendly.”

“Part of the LIVE ‘signature sound’ is the sound of SSL,” says guitarist Chad Taylor, noting that Tom Lord-Alge, who has mixed the majority of the band’s recordings, works exclusively on an SSL 4000 G Series console. After Taylor and Lord-Alge spent a day evaluating Duality, the decision was easy.

“There’s a convenience factor and a history of the SSL that exists through Duality. In the studio, I’m predominantly focused on the performance of the musicians and the arrangement of the song, and less on the technical aspect of the engineering.

“I found that those worlds got married very conveniently through the Duality.”

“My immediate reaction was that there’s a dimension and a spatial factor to the Duality,” Dahlheimer adds of SSL’s SuperAnalogue sound. “Perceptively, it was really very clear.

“There’s definitely a punch and a clarity, especially to the drum tracks. One of the other qualities is the bus compressor. Once you are in it, there’s cohesiveness to the songs that jumps out.”

Duality’s unparalleled sonic characteristics, analogue/DAW hybrid approach and historic lineage made it the only choice for Think Loud Studios, say its principals.

“It has a front-end signature, in particular with the mic pres and the EQs, that really plays into that soundscape,” says Dahlheimer. “That definitely helps our creative process.”

The console’s hybrid approach, combining a traditional analogue path and processing with DAW control in a single hardware surface, allows the creative process to flourish as it did in the pre-DAW era, Taylor says. 

“One thing we took into consideration was that we still like to work in analogue,” he adds “With the convenient flexibility of Duality, we are able to switch very fast back-and-forth between our analogue and the digital workstations.

“Working with Duality pulls my brain from looking at music on a computer screen to actually interfacing with the console, like we did 20 years ago. The concentration is on ‘listening’ again, and not ‘seeing’ the music so much. That’s an important characteristic to the creative flow.”

The choice of Duality has paid immediate dividends as LIVE went to work recording its forthcoming album.

“We had a bunch of tracks in a rehearsal state that had been recorded through various preamps,” Dahlheimer says. “We got a very polished sound through the Duality very fast. I was thrilled with that. Duality helps outboard equipment shine, but we found ourselves using a lot of the onboard preamps in tracking.

“In fact, we recently re-recorded existing drum tracks to take advantage of Duality’s sonic signature. There was definitely clarity, presence and dimension to the new tracks.”

Taylor also recalls having previously owned an SSL G+ console.

“The format and feel of the Duality remind me of the G,” he says. “While it simultaneously moves into a new-world environment of, essentially, Pro Tools control and interface, but one that still has the markings of a traditional analog console. I’m glad we stayed in the SSL family.”

Solid State Logic

{extended}
Posted by Julie Clark on 05/20 at 02:12 PM
RecordingNewsAnalogBusinessConsolesStudioPermalink

Friday, May 16, 2014

Focusrite Announces Saffire PRO 26 FireWire/Thunderbolt Compatible Audio Interface (Includes Video)

Record at 24-bit/96kHz with four Focusrite preamps, 18 in/8 out, using FireWire or Thunderbolt

Focusrite announces Saffire PRO 26, the latest addition to the Saffire PRO range of FireWire/Thunderbolt compatible audio interfaces for expanding recording and live system capabilities, housed in a portable, desktop-sized chassis.

Saffire PRO 26 provides an extensive selection of professional analog and digital I/O options —a total of 18 inputs and eight outputs includes four preamps, two instrument inputs, two headphone outputs, six line outputs, and ADAT and S/PDIF connectivity.

Saffire PRO 26 connects to a Thunderbolt port via a FireWire to Thunderbolt adaptor (not included) or directly to a FireWire 800 port with the cable provided. Its dual-protocol compatibility (Firewire and Thunderbolt) means it will work seamlessly for years to come with the next generation of computers.

The four Focusrite preamps provide a great deal of recording flexibility while also ensuring low noise and distortion with plenty of headroom to capture the full dynamic range of even a loud drum kit or guitar amp. Precision 24-bit/96-kHz digital conversion and JetPLL jitter-elimination technology maintain pristine audio quality in both analog and digital domains.

Saffire Mix Control, a control software application that runs on the host computer is particularly useful for live situations. The low-latency 26 x 8 DSP mixer/router provides flexible output routing and monitoring for custom monitor mixes as well as intuitive one-click presets to help in setting up sessions as quickly as possible, whether tracking, mixing or monitoring.

Although Thunderbolt provides some advantages, it’s only available on the very latest computers. An audio interface fitted with FireWire can be used on both older computers and Thunderbolt-equipped computers via an inexpensive adaptor.

ADAT optical input allows extensive input expansion of up to eight additional preamps or line inputs. Pair Saffire PRO 26 with Focusrite’s OctoPre MkII eight-channel preamp and instantly transform it into a 12 preamp unit.

Included is a free DAW in the form of Ableton Live Lite, Focusrite’s professional Midnight and Scarlett plug-in suites, Novation BassStation virtual synthesizer and 1 GB of Loopmasters sample content.

 

 

image

 

Focusrite

{extended}
Posted by Keith Clark on 05/16 at 04:15 AM
AVLive SoundRecordingChurch SoundNewsProductAnalogAVDigitalEthernetInterconnectNetworkingSignalSoftwarePermalink

Monday, April 28, 2014

Czech Violinist, Composer and Singer Chooses Audient

A compact analog mixing console was what Czech violinist, composer and singer Karel Holas was after for his Prague studio -- which he accomplished with the purchase of an Audient ASP4816.

A compact analog mixing console was what Czech violinist, composer and singer Karel Holas was after for his Prague studio.

“From the wide variety of quality studio consoles available, I found only one complying with all my requests and wishes: Audient ASP4816,” he explains, having done his research.

“Excellent microphone preamps, great and very easy routing options, precise EQ sections together with optimal ergonomics for studio usage,” he says, outlining some of the features on his wish list.

Holas opened his studio earlier this year, and describes his impression when the Audient desk first arrived.

“I must say I was really amazed when the console was plugged in. The sound quality and the whole potential of the console was confirmation that I had made the right choice with the ASP4816.”

A popular choice for small music production studios looking for all the features of a large console in a compact, ergonomic format, at an affordable price, the ASP4816 is an analogue desk with fully-featured inline architecture. This particular desk was supplied and delivered by Czech Audient distributor, MusicData.

Currently a member of Cechomor, a Czech traditional music band playing songs in rock arrangements, Holas has collaborated with many international artists such as Suzanne Vega, Tony Levin, Celtic harp musician and singer Alan Stivell and many others.

Audient

{extended}
Posted by Julie Clark on 04/28 at 12:37 PM
RecordingNewsAnalogConsolesStudioPermalink

Friday, April 25, 2014

Multiple Consoles For Live? Top Engineers Weigh In

The where, when, why and how, with problems and solutions differing.

A few years ago, my company developed a prototype of a console switcher that would enable an engineer to quickly switch to a backup should the main desk go down, or quickly switch between multiple consoles at events such as festivals.

But when we showed it to various engineers, the response was all over the place. Some thought it was a great idea, others felt that with modern processors, the need was no longer there, and some suggested fixes such as increasing the size to accommodate larger systems.

We decided to table the idea, but I thought it would be interesting to reach out to some of the same engineers to get their take on using multiple consoles and the concerns they encounter.

And just as above, the problems and solutions differ. The cast includes James Warren (Radiohead), Sean Quakenbush (Robert Randolph), Dave Natale (Rolling Stones), Brad Madix (Rush) and David Morgan (James Taylor).

The most common situation where multiple consoles are used together, of course, is connecting a support band to the main system. Other uses include festivals where multiple bands share the same PA, corporate shows, TV shows, and performances where large orchestras increase the channel count.

Brad Madix

According to Natale: “When subbing one mixer into the other, the main console will usually act as the master. We also see many situations where a matrix switcher is used to feed the PA.”

Sub-Mixing Consoles
Subbing consoles together is done using the sub-group inputs, channel strips, or sometimes even using the mic inputs. Madix: “If bringing one mixer sub or mains out into sub ins, there’s usually not any problem. If bringing into line-ins, there might be issues with matching gains as line ins tend to have less adjustment range level-wise.

“Coming into mic inputs can present challenges from impedance matching to level mismatches where mic preamps might not have the range to handle the levels, even with pads inserted. It’s something to steer away from, for sure, but sometimes the only option.” 

Dave Natale

Quankenbush: “We sometimes encounter noise from different power systems such as generators and there are often gain issues between some analog consoles and the digital boards. For instance, one console’s 0 dB may not be the same on another desk. I’ve found that some digital consoles do not ‘play well together’ due to gain stage issues where one may be so hot that it overloads the other.”

Natale: “Hum, buzz and level discrepancies can pose problems. I usually have transformers in hand to solve noise problems.”

Matrix, Processor
There are other ways to switch and combine consoles such as using a matrix switcher or an audio processor. And with today’s digital desks, even more options come into play.

Sean Quackenbush

Warren: “When combining consoles, since most bands are now using digital desks, we usually connect the sub support console into ours via an AES connection. We give a festival either analog or AES from our system processor.  In both cases, we will often be giving a separate sub feed.” 

Quankenbush: “Most festivals have switching systems for the left, right and sub fills, but you do still see some festivals where they want you to drive in to the main console with stereo. The big problem is you will load in early, EQ and sound check for your band with their EQ bypassed or flat.

“Eight hours later, the house system engineer or other mixers will have hacked the EQ to all hell and all of the sudden your show sounds way different than your sound check earlier in the day. My preference is to bypass all of that by connecting directly to the audio processor and then save my own page.”

Noise Problems
As noted earlier, noise problems do arise, and the most common problem solver is inserting an isolation transformer into the signal path.

A transformer is a magnetic bridge that converts the audio signal into a magnetic field at the primary winding, employs a core made from laminated nickel, steel or a combination as a conduit for the magnetic field and then this excites a secondary winding which in turn generates current.

The beauty of a transformer is that the input and output are completely separate. This stops tray DC current from traveling between the input and output which helps eliminate the hum and buzz caused by so-called ground loops.

Morgan: “For years, Yamaha and Midas consoles did not like to be combined. One often needed to lift the AC ground on one of the desks and rely on audio ground only.

“As long as the consoles share the same AC and audio ground, transformer isolation is not usually necessary. If I’m unsure of the system AC ground, or if there is too much going on electronically at FOH, I do prefer inserting transformers.”

Madix: “I’ve had to use transformers occasionally when feeding to lawn delays and the system for the hearing impaired. For this, we use a box with two transformers, plus ground lifts.”

David Morgan

Both Warren and Natale note that they always carry transformers in their kits. Quakenbush adds: “Back in the day, I was the lawn guy for a large amphitheatre and always had pockets full of isolation transformers. I still have tons of in-line transformers in my workbox. They don’t come out a much as they use to, but I still use them for delay towers or sometimes when I sub another desk into mine.” 

Averting Disaster
One of the most common concerns that folks have with digital technology is the stability of the console’s internal computer. Thus, adding a second console would seem to be a natural solution.

James Warren

Interestingly enough, this no longer seems to be as prevalent as it once was. I was recently at a Bob Dylan concert and front of house engineer Jim Homan was working with a new digital console that was having some software conflicts. I asked him if he had a backup, and he said that he didn’t, but if he had to, he could quickly patch in the support band’s mixer and be up and running fairly quickly.

Warren echoes this approach: “In a touring situation, I would refuse to use a console that I felt needed a permanent instant backup option. On Radiohead at the moment, I have the opening act’s console loaded with my show and plug-ins in case of catastrophe, but it’s not online or standing by during the show.”

Morgan: “We carry a backup computer for the console, but I haven’t needed it in over six years.” Madix replied with the same sentiment. However, Natale had a different take: “I generally go analog for just that reason. I’m not prone to nearly as much instance of console failure as my much more daring counterparts that use digital consoles. When I do TV, I have to use a digital console and the only fail-safe (ha, ha) device is a UPS on the console.”

All of this to say…there are many ways to connect consoles together or to share the PA system. Most engineers carry line level isolation boxes in case noise is encountered, and today’s digital desks seem to be less problematic than they were just a few years ago.

Peter Janis is president of Radial Engineering, and has worked in professional audio for more than 30 years.

{extended}
Posted by Keith Clark on 04/25 at 02:13 PM
Live SoundFeatureBlogStudy HallAnalogConsolesDigitalEngineerInterconnectMixerSound ReinforcementPermalink

Wednesday, April 23, 2014

SoundTools Debuts New CAT Snake At Coachella

Units create a cost-effective 4-channel analog snake that can alternatively carry up to four AES/EBU digital feeds utilizing common Cat-5 cable.

SoundTools debuted its latest product, the new CAT Snake, in prototype form at the recent 2014 Coachella Valley Music and Arts Festival in Indio, CA.

CAT Snake units create a quick and cost-effective 4-channel analog snake that can alternatively carry up to four AES/EBU digital feeds utilizing common Cat-5 cable.

“The CAT Snake is a wonderful solution for the analog lines we run to delay clusters at Coachella,” says Dave Rat, president of Rat Sound Systems, the primary audio vendor for the event. “We literally entrench thousands of feet of analog and digital audio cable that invariably gets destroyed during load out. By switching to the CAT Snake for analog backup lines, we can just leave the shielded Cat-5 in the ground and the cost per foot is so affordable, it’s easy to replace.”

The application of the CAT Snake at Coachella showcased it’s functionality for live sound as well as its potential use in recording, commercial, and home projects.

“It’s simplicity at its best,” says Bryant Poole, lead engineer on the project. “You can run four lines of audio down a simple shielded Cat-5, Cat-5e, or Cat-6 cable. It’s a passive unit that provides the added versatility of having an EtherCon input or feed through connection on either side.”


image


SoundTools
Rat Sound Systems

{extended}
Posted by Keith Clark on 04/23 at 01:06 PM
AVLive SoundChurch SoundNewsProductAnalogAVDigitalEthernetInterconnectSignalSound ReinforcementPermalink

Monday, April 21, 2014

Subjective Versus Objective: If It Sounds Good, Is It?

Does science (objective) or art (subjective) play the more important role?

As with the ever-ongoing debates about “tubes versus transistors,” “analog versus digital” and “Mac versus PC,” there’s not likely to be agreement any time soon about “objective versus subjective” when it comes to sound quality.

Extremists in the “Objectivist” camp argue that, “if it can’t be measured, it doesn’t exist” while on the other hand, the “Subjectivist” side firmly backs the idea that “human beings can hear things that can’t be measured.”

How often has it been suggested, “use your ears as the final determinant” in making a decision about sound? At the same time, most would agree that a fundamental understanding of audio systems, including the basics of how each component works, how to set gain structure, and so on, logically can lead to “better” sound quality.

Does science (objective) or art (subjective) play the more important role?

ABX Or Death
Since its development as a scientific testing method, ABX has gained ground as a clear way to determine the threshold of perceptibility in a group of test subjects.

The basics of ABX: two different sources are compared - source “A” and source “B” - and the subject must make the decision as to whether choice “X” represents either A or B. If the subject can reliably (i.e. in a statistically significant manner) identify the sources, then it is concluded that there is a perceptible difference between the sources. Otherwise, the differences are deemed insignificant.

There are some good things to be learned with ABX, and it’s proven to confound many the “golden ears” in tests involving things like 44.1 kHz versus 96 kHz sampling rates, 16-bit versus 24-bit quantization, and others. And it turns out that it’s not common for subjects to be able to reliably identify these sources.

However, I contend that there’s a vast difference between a short-term test like ABX and a longer-term experience with a product, system and the subject itself. Humans have demonstrated a truly amazing ability to learn just about anything.

Take a person who’s never spoken anything but the English language, and stick him/her in Japan for a couple of years. This person will most likely learn to speak Japanese, engaging a new part of the brain.

Or take a person who’s only tasted wine costing less than $10 a bottle. A few months after being introduced to $150 bottles of wine (let alone $3,500 bottles!) and learning about the different varietals, harvest timing, and other specifics, he/she will balk at the cheap stuff.

Even more importantly, this fledgling student of wine will have picked up the ability to discern much finer differences between all types of wines.

In both cases, what changed these people? Exposure, mostly. We all have what some call “paradigms,” meaning that we each filter outside stimuli through our own various levels of experiences and beliefs.

Fixed Level Of Bandwidth
I call these changes through exposure successive thresholds of awareness, and contend that part of this is that human perception is scalable in terms of resolution. With computers and test equipment, there is a fixed level of bandwidth and resolution available.

Not so with people - the longer someone spends being exposed to an experience, the more resolution that person is able to impart to that experience. An analogy closer to home for us audio geeks: the person that has only used a cheap dynamic microphone for years will likely find that even the lowest-grade condenser mic sounds amazing. He will hear tons more resolution, less distortion, and better transient response.

This same person will also wonder how a Neumann mic costs much more, and whether or not it would be possible to sound that much better. And in fact, upon hearing the Neumann in comparison to the cheap condenser, he will conclude that indeed, there is not really that much difference between the two.

Now take that same person five years later, after he’s made several records and used a plethora of top mics of various makes. Now he should clearly be able to identify the differences between the cheap imitation and the real thing, having reached a much higher threshold of awareness between the different mics.

Only One Problem
A few years ago, I read an interesting article about how Dunkin’ Donuts intended to update its marketing plan to target Starbucks customers, based on a very simple idea: offer the same quality of coffee, but more quickly and at a lower price. There was only one problem. These weren’t the reasons that Starbucks customers were buying coffee from Starbucks. They didn’t want it cheaper or more quickly.

What they did want was the Starbucks experience—the club chairs, the subdued lighting, the fancy woodwork, the ridiculously overpriced accessory products, and whatever else they’re seeking. For this, they’re willing to wait (part of the experience) and pay more (another part of the experience).

Although it could be argued that they would appreciate the coffee being less expensive, it’s been proven over and over that there is usually a “right price” associated with a brand experience, and if the price is either too high or too low, the brand will lose credibility.

So what does all of this mean in terms of audio and the Subjectivists versus Objectivists? For one thing, different people perceive things differently, period. What’s important to some is not important to others, and visa versa. 

For some, a slightly lower noise floor in a mic is not worth either the extra cost or the resulting lack of perceived resolution, while for others, it might be just the ticket for their application. Thus there can be no consensus on whether or not a lower noise floor is always “better.”

One thing I firmly believe is that both approaches are important for the improvement of audio (or anything else that is part of someone’s experience).

The Accidental Designer
Sure, there are stories where accidental discoveries made improvements in design. For instance, the story of the German broadcast engineer in the late 1930s that inadvertently left a high-frequency oscillator “on” while recording an orchestra.

The result? For the first time, there was playback fidelity beyond 10 kHz. This accidental discovery lead to the implementation of an AC bias for analog tape recorders, and it also pushed the envelope of what was possible with this type of system.

However, despite the muddled beginnings of AC bias, a scientific approach was required to produce repeatable, reliable and predictable results. The required circuitry had to be thoroughly understood by analog design engineers, and the right frequency and right amplitude had to be identified.

Then the right combination of these factors for each different tape formulation had to be developed in order to realize the full potential of the bias signal. It took until the 1950s before this was well understood, resulting in improvement of both subjective and objective experiences for the listeners of tape recordings.

One real problem with measuring various changes in audio quality and attempting to both attribute them to specific causes and simultaneously predict how they will be perceived is that – in the first place - we often don’t know exactly what to measure. Of course, we know the basics such as amplitude response versus frequency, phase response, distortion in its various forms and the like.

But it’s exceedingly difficult to get detailed measurements with real source material in place of standard testing signals. (Meyer) SIM and (Rational Acoustics) Smaart are measurement tools in this direction, and they’ve greatly benefited sound reinforcement.

At the same time, there is no solid standard for transient response measurements and the resulting perceived effects. Several manufacturers claim that by extending frequency response of a system well past the “audible” limit (say, to 50 kHz) and maintaining phase accuracy through that range, that transient response and distortion will be improved in the audible band.

But even so, is this necessarily the way to predict that the system will sound good? Perhaps it could be argued that all other things being equal between two systems, the one with the lower distortion will “sound better.”

But then again, an interesting experiment done long ago by Bell Labs resulted in the conclusion that for a limited-bandwidth system, the one with more distortion was perceived as sounding “better.”

Perhaps this is one way to explain why low-power, all-tube, all-Class-A amplifiers are often perceived to sound more “musical” than huge, solid-state, “mega-kilowatt,” machined-aluminum monsters that are competing for the same piles of money.

Or maybe it’s other, psychological factors, such as the idea that tube amplifiers replaced the hearth in the home as a centerpiece around which to congregate…

Or perhaps it’s a result of something that is more easily quantified.

Class-A amplifiers distort differently from other designs. Not only this, but by running “wide open” in some cases, there’s more power available for short-term small-scale dynamic changes such as transient information.

It can be easily shown that although two systems may have the same signal-to-noise ratio and the same distortion figures on an analyzer, they sound radically different. The spectra of the noise, and the character of the distortion, play huge roles in perceived sound quality.

So again, the challenging question about quantifying performance in audio systems is what to measure in the first place, and how to measure it.

Getting Along
The bottom line is that both camps have something very important to offer. Without a scientific approach, we’d be stabbing in the dark trying to find solutions to problems about which we know very little.

But without a reliance on the subjective experience, even our most clever inventions would perhaps never reach the level of “art.”  What good can come of setting fire to a silk-screened portrait of Andy Warhol in the middle of the woods if there’s no one present to snicker?

Designers and sound system users make decisions every day based on whatever they have at their disposal, including theory, available equipment, testing and measurement, intuition, and finally, critical listening. If there is not a balance among these resources, the results are likely to be unbalanced.

How would you like some power amps with “DC to light” response but producing crappy sound? Care for some loudspeakers that sound amazing but look like a “Dogs Playing Poker” on black velvet? How about mics that can pick up a gnat burping but make a Stradivarius sound like a banjo bowed with rosined fishing line?

Let’s leave it to the great Duke Ellington: “If it sounds good, it is good.”

Karl Winkler is director of business development for Lectrosonics and has worked in professional audio for more than 20 years. {extended}

Posted by Keith Clark on 04/21 at 09:46 AM
Live SoundFeatureBlogOpinionStudy HallAmplifierAnalogDigitalEngineerMeasurementSignalSound ReinforcementPermalink

Thursday, April 10, 2014

Virginia Arts Recording Installs API 1608 Analog Console

Recently replaced large-format digital console with a 16-channel 1608 analog console with P-Mix fader automation.

As part of a complete architectural and electronic renovation, Virginia Arts Recording recently replaced its large-format digital console with a 16-channel API 1608 analog console with P-Mix fader automation.

The north-central Virginia-based studio has been serving local musicians, labels, and radio stations for over 30 years. The current owners, Chris Doermann and Sean Dart, are embracing that history and the industry’s pivot to analog with the new API console and a 24-track, 2-inch tape machine.

Virginia Arts Recording also retains all the professional digital platforms with top-end converters to allow projects to effectively hybridize between the two technologies. The facility resides in a historic house in southeast Charlottesville, just miles from the University of Virginia campus. A little over a year ago, the coupling that merged the city water supply and the house’s water heater on the second floor failed just as everyone was closing up shop for a holiday weekend.

“When we returned, the control room, and much of the equipment was totally wrecked,” recalls Dart. “The digital console was one of the casualties, but we decided to make the most of it. We wanted to put the studio on solid footing for the next twenty-five years.”

Analog consoles, tape machines, ADATs, and a steady progression of DAWs all had a place in Virginia Arts Recording’s history. Doermann and Dart decided to build a hybrid analog/digital studio with a workflow that made negotiating the two technologies transparent. 

“We definitely wanted an analog console, and we pride ourselves on capturing big drum sounds,” says Dart. “That’s API’s signature talent, so naturally we chose the 1608.”

Doermann and Dart took an API factory tour as a part of their research. “Interacting with API is a different experience,” notes Dart. “Mark Seman of API invited us to the factory, and we packed a few mixes that we know well. API let us see everything, and gave us a few hours behind the 1608.

“It sounded amazing, and the feel of real faders has been a welcome relief from menus and double clicks. I just get in there with my hands, and thank API for giving us the recording feel we were missing.”

API Audio

{extended}
Posted by Julie Clark on 04/10 at 02:18 PM
RecordingNewsAnalogConsolesInstallationStudioPermalink

Audient ASP4816 Console No Compromise For Hawaii Studio

The first ever Audient ASP4816 in the South Pacific has been installed at The Analogue Café in Hawaii.

The ever Audient ASP4816 in the South Pacific has been installed at The Analogue Café, a self-professed “old fashioned recording studio.”

Studio owner Eric Malamud was keen to ensure that The Analogue Café lives up to its name, and his research into analog gear led him straight to Audient. Discovering that console designer David Dearden is Audient’s co-founder piqued his interest. “I already own a vintage Midas XL200, so I’m a Dearden fan by default,” says Malamud.

General manager and in-house technician Robert Unger is happy that Malamud didn’t compromise on the desk, and lists a few of the many benefits it offers the studio. “The routing flexibility is excellent, especially the buss assignments to subgroups and outputs,” he notes. “The integrated internal power supply is silent and energy efficient, and the console has the best fold-back and monitoring system integration ever.”

The in-line console packs the key features of a larger console into a compact, ergonomic form, also including the same pre-amps found in Audient’s Dearden-designed flagship console, the ASP8024.

“The Audient pre-amps are time tested, and very clean,” confirms Unger. “In the old days EQ was for surgery, we never use EQ when tracking,” he continues. “When we do need it the result is excellent – especially the filters.”

Malamud adds, “The studio maintains the feel of a 1950s—1970s analog studio using 48 tracks of iZ RADAR and classic outboard,” adds Malamud. “As the studio motto goes, No computers are harmed in the making of our records.”

Audient

{extended}
Posted by Julie Clark on 04/10 at 01:30 PM
RecordingNewsAnalogConsolesStudioPermalink

Monday, April 07, 2014

Ahead Of The Game: Console Strategies For Festivals

The goal is to be as prepared as possible. Spring is nigh...

Mixing at festivals – good times! Or is it?

Anyone who has worked as either a guest mixer or system tech in a festival environment probably has stories about the inherent ups and downs and, certainly, the hyper pace and stress that are part of the gig. And we’ve all heard a few horror stories of artists hitting the stage patched incorrectly or without a sound check.

But there’s also the unique thrill of mixing in a hyped environment with tens of thousands of fans on hand, and sometimes in really cool outdoor settings. The goal of the mix engineer is to be as prepared as possible, particularly when it comes to working with the console. Spring is nigh…

Preferences & Strategies
It’s been common for years to see multiple consoles “leap-frogged” between acts, allowing one or more offline consoles to be dialed in while another is live. They may be switched over by the system engineer or sub-mixed to a master console, and in the latter case, gain structure or ground loop hum/noise issues can pop up between consoles.  Carrying in-line pads and audio isolation transformers is always a good idea.

Digital consoles have obviously changed the workflow at festivals by allowing preset show files to be prepped and uploaded, which helps in terms of establishing baselines and promoting efficiency. Premium analog boards may still be carried by certain headliner acts, but they’re usually not shared.

Whatever the console(s) in use, advancing the date is still the most important step in a successful gig. Even the best system techs can’t prepare properly if they don’t have enough information in advance. Further, even when this info is available and shared ahead of time, it’s still wise to arrive at the gig with a copy of the stage plot, patch list, input list, and whatever else is important to the production.

Having mixed at plenty of festivals and other multi-act events, I’ve developed a number of personal preferences and strategies. And I’ve observed that the balance of science versus art that we know as “live mixing” tends to weigh heavily toward the science side when the “run-and-gun” mode common to festivals kicks in. After all, things just have just work, first.

A Yamaha CL5 provided by Gand Concert Sound to serve as the house console at the annual Pitchfork Music Festival in Chicago.

But as veteran freelance mix engineer Chris McMillan (John Mark McMillan/Promenade Media) told me, “Mixing is much better when the art takes priority over the science, and that means ergonomics can determine how nuanced your mix becomes. I like channels grouped the way I’m used to so that I see what I need and never know anything else exists.”

This is where festivals are so different than tours. Touring engineers get very used to their daily setup being consistent, and can take advantage of that repeatability to achieve highly detailed mixes. System techs that aren’t mixers should try to keep in mind that mix engineers aren’t always crazy or unrealistic when they want their console laid out a certain way.

It’s about familiarity. It really does matter if the lead vocal gets patched to the rack tom channel. Things like this can be dealt with in a pinch, and maybe quickly, but they can impact the end result by either causing a failure or a compromised (weaker) mix.

In talking with Chris and a couple of other festival mixing veterans, and thinking about my own experiences, certain themes are clear. Mix engineers desire a “perfect” console setup and the ultimate processing tweaks to satisfy their mix plans. But when working festivals, they do realize that it’s a daunting task to support many acts a day as opposed to one artist on multiple tour dates. As a result, they just hope for a reasonably well-tuned PA, a thoughtful system approach, solid gain structure, and an intelligent output bus layout.

Patching Adventures
Input patching is critical – particularly at festivals. What’s the best way to handle it? If the sound company has the qualified hands and there is enough change-over time, it’s great when stage inputs can be updated for each artist on the bill.

Whether the consoles are digital or analog, this extra effort goes a long way in helping keep things familiar for visiting engineers.

And if troubleshooting becomes necessary, engineer(s) are likely to have the stage patches for their artists memorized and know things like “hats are on line 5” and the like. All of this said, it’s simply not all that realistic in most festival situations…

Festival stages are typically patched in a logical order with plenty of lines, and the patches don’t change between acts. If one drummer needs 10 lines and another needs only six, then the latter has four open lines during his set – the overall count remains the same. 

“Soft patching” on digital consoles allows laying out input and output channels in any order without making physical patch changes. This is extremely powerful. No longer does snake line 1 have to appear on input channel 1. Each engineer’s preferred console layout can be implemented without impacting the physical patches. But this requires sharing console show files in advance (pun definitely intended) or doing it on site while another act is playing.

It’s common to use matrixes to drive PA outputs such as main left and right, down fills, front fills, delay zones, subwoofers, etc. Many engineers simply distribute their stereo mix across these various zones (either L/R or L/R+sub), while some actually mix to each zone, which requires building specific mixes into each matrix. The exact PA zones and distribution varies per event, per stage, and not all companies do it the same.

But whatever the configuration, it’s imperative that the console’s output patches match the PA. With digital consoles this means soft patching the output patches, and for this reason, system techs need to be careful when loading each act’s show file, as output patching errors or surprises can create a perfect storm and wreck a system real quick.

A Soundcraft Vi6 as the front of house console provided by Premier Production & Sound Services for the main stage at Louisiana State University’s Groovin’ on the Grounds multi-act concert in Baton Rouge.

A couple of times I’ve worked as a guest mix engineer at a festival and then stayed on as a pre-booked system tech. While this isn’t my forte or preference, I found it very interesting to work from the other (host) side of things. Many visiting engineers arrive with an expectation of certain doom, and it was fun to “make their day” with exceptional support and PA organization.

In one case, the long-time mix engineer for a well-known classic rock band clearly wasn’t happy about the digital console at FOH. He just wanted to “get by and get out of there.” I knew this desk inside and out and did everything possible to make it painless for him. He sought to keep it simple, with input faders and EQs accessible, in order, but with no other processing – not even DCA groups.

Further, he actually broke out his console tape and Sharpie and proceeded to label the input channels analog style, in spite of the nice programmable LCD labels! When I pointed out that the tape was only applicable on “Bank A” and would be inaccurate as soon as he banked the faders, he simply replied, “I don’t bank.” The band fit on the 24 input faders without any banking (layering), and by the end of the first song, it sounded absolutely amazing. Simple setup, talented musicians, and great ears.

In considering this topic, I did some Q&A with long-time mix engineers Daniel Ellis (David Crowder Band, Jesus Culture) and the aforementioned Chris McMillan.

Here’s what they had to say.

What do you appreciate most from the host system tech in terms of console prep and work flow?

Chris McMillan: I love it when signal flow and busing are simple. That’s really the most important thing. I want to know I’m just responsible for a stereo mix and maybe a send for subs, and everything else is going to be fine. If that’s right, and there’s a solid talkback situation, then we’re golden. It’s also much appreciated when the system tech has thought through the input list and our specific goals and considered what that means in terms of the system configuration. There’s nothing as useless as taking the time to advance a show only to have nothing prepared and no feedback.

Daniel Ellis: I want to see a production console for videos, emcees, and things that I do not need/want in my show file. This also means that I can load and prep my show file in between acts without waiting for the perfect 30-second gap where nothing is happening on stage.

What’s your take on “festival patch”?

CM: In an ideal world, I stay away from festival patch, although this is pretty much only accomplished with a show file. I like channels grouped the way I’m used to so that I see what I need and never know anything else exists. You know, the typical spoiled brat method of engineering.

DE: As a headliner I want my show to be patched per my input list. The only problem with this is that many festival patch guys for some reason can’t get it right the first time so half of the sound check ends up being “fixing the patch.” At least this is how it works at Christian festivals. Sometimes it seems like a random guy has been hired off the street to patch when in essence, patching is one of the most important jobs.

A DiGiCo SD5 that’s one of numerous SD models supplied by Clearwing Productions for the annual Summerfest in Milwaukee.

Do you carry a show file if it’s a compatible digital console or do you send it in advance? Or neither?

CM: I carry a show file if it seems like it will make a difference. Sometimes the process of conforming a show file or the time it takes to be convinced it’s correct isn’t worth the effort, because patching and busing can become compromised. Anyway, the acts I work with aren’t doing anything so weird that a default festival scene can’t work as a great starting point.

DE: I always try to know ahead of time what console I’ll be using and have a show file ready. Even if it’s a blank show file built on my laptop, I find that it helps because at least I know where all of my inputs are. If you try to run a 48-input show from a festival console file, you spend the entire time switching between banks trying to remember where everything is. It helps me tremendously to have the same workflow every time even if I’m starting with flat EQ and no processing on anything.

Do you find that “artist EQ” or “output bus processing” is usually enough to get your sound or do you often wish (ask?) for access to the PA processing?

CM: Limited bus-style processing is usually acceptable, if not from a creative standpoint, then from the understanding that everyone else is working off of that same tuning.

DE: Lately I often find myself at an Avid desk at festivals, so I just slap a Waves Q10 (10-band paragraphic EQ plug-in) across the stereo bus. Luckily I haven’t had to do much to the systems themselves. Just two or three small cuts on the Q10 in problem areas and I’m usually happy. If it’s a console that doesn’t work with Waves, I simply use the parametric on the master out.

What makes for a good system tech?

CM:
I don’t hesitate to communicate with the system tech about expectations and any changes I feel the PA needs. Most good techs can balance the reality of the promoter and their employer’s expressed interests and still meet your creative and technical needs. A good tech wants a good sounding show in reality and not just on paper.

DE: Good attitude and good ears! And please don’t set up a measurement mic in one spot and put in 15 EQ adjustments.

Kent Margraves began with a B.S. in Music Business and soon migrated to the other end of the spectrum with a serious passion for audio engineering. Over the past 25 years he has spent time as a staff audio director at two mega churches, worked as worship applications specialist at Sennheiser and Avid, and toured as a concert front of house engineer. He currently works with WAVE in North Carolina and can be contacted at .(JavaScript must be enabled to view this email address).

{extended}
Posted by Keith Clark on 04/07 at 12:57 PM
Live SoundFeatureBlogStudy HallAnalogConsolesDigitalEngineerMixerSound ReinforcementTechnicianPermalink

Friday, April 04, 2014

It’s All Interconnected: Analog & Digital Cabling For Performance Audio

A few weeks ago one of the neighbors in the industrial complex where I keep my shop came over to say hello while I was in the middle of doing some PM (preventative maintenance) on cables.

As I sat at a bench surrounded by piles of microphone and loudspeaker lines, he asked why I was spending so much time on “stupid cords.” I replied, simply, that without the stupid cords, the rest of my equipment is worthless. A system is only as good as its cables, interconnects, snakes, and networks—period. Fortunately, we have a wide variety of analog and digital options to choose from, and it’s getting better on a constant basis.

Digital audio transport technology (a.k.a., digital snakes and networks) have taken pro audio by storm in the last few years, pushed at least in part by the proliferation of digital consoles, with virtually every manufacturer offering some way to move audio over Cat-5/6, coax, and fiber optic cabling. While digital networking certainly offers a lot of advantages and flexibility, it hasn’t pushed analog completely out of the picture—and in my opinion, at least, I don’t think it will, at least in the foreseeable future.

One reason is personal preference, another is the sheer amount of cabling that will have to be replaced, and yet another big one is that digital systems need A-D and D-A conversion at each end of the cable (or fiber), which increases cost, and this is particularly dramatic for smaller systems that only run a few channels of audio. As technologies improve and prices come down, I’m sure we’ll see even more digital, even on the smallest of shows, but there’s still the issue of preference.

That said, let’s take a look at the various cables, connectors, and audio transport used in production audio systems.

Here To There
The first cable in the signal chain is usually the humble XLR cable sporting 3-conductor connectors at each end. These cables connect low-impedance microphones and direct boxes to consoles, as well as send line level signals around to various gear.

Left to right: XLR (female end), TRS 1/4-inch, signal 1/4-inch, loudspeaker 1/4-inch, loudspeaker 1/4-inch with larger barrel, 4-pin Speakon, 19-in Soco male, and 19-pin Soco female. Note the use of colored heat shrink to quickly ID signal (blue and red) and loudspeaker cables.

They operate on the balanced principle and contain two insulated conductors that are twisted together inside a shield under the outer jacket. The audio signal is applied to the pair of conductors differentially, that is to say that one wire has the polarity of the signal reversed but the levels are the same. Any noise or outside interference that gets into the signal lines will mostly be defeated because one conductor transmits the noise with a positive polarity and the other is at a negative polarity.

When signals with opposite polarity (in this case, the noise) are combined, they will cancel each other out. The reason the inner conductors are twisted is that it allows external noise to be introduced to both signal conductors equally (or as equally as possible) and improves the common-mode rejection ratio. Some cables use four inner conductors (two pairs of two) that offer better rejection from outside electromagnetic interference like transformers and fluorescent lighting ballasts.

The conducting shield that wraps around the inner wires is used for the signal common and can be a spiral winding or a braided winding. Braided shields provide more surface area coverage and better rejection of radio frequency interference (RFI) than spiral wound shields.

Similar in construction to the XLR is a cable that instead has 3-conductor 1/4-inch phone plugs at each end, usually called a TRS cable. The TRS refers to Tip, Ring, and Sleeve, the three conductor positions on the connector.

These are commonly used as interconnection cables between rack gear and are a popular option for manufacturers who want to use balanced connections but have limited real estate on the product in which to squeeze in XLR connections.

Many consoles have insert jacks that allow patching of external processing into a channel or group. They normally use a TRS 1/4-inch jack and a special Y cable called an “insert cable” that is outfitted with a TRS plug on one end and a 2-conductor 1/4-inch plug at each end of the Y that is used to route to the inputs and outputs of the external processor. The TRS end is usually wired so the tip is the send to the external unit, the ring is the return and the sleeve is the shield or common.

While similar in looks to a TRS cable, a regular 1/4-inch signal cable is quite different. It has only one inner conductor surrounded by a spiral or braided shield. They are used with high-impedance signals from a guitar or keyboard to connect them to a stage amplifier or DI. The outer braid acts as both a conductor and a barrier to help keep RFI and other noises from reaching the center “hot” conductor.

When used with a guitar or other high-impedance input, the cable’s capacitance couples with the high impedance to create a low-pass filter that varies depending on cable length. The longer the cable, the more highs it rolls off, so 1/4-inch cable runs are usually kept under 25 feet in length unless they’re serving electronic keyboards, which output a hot line-level signal that can drive longer runs.

Another cable that may look identical to these first two is the 1/4-inch loudspeaker cable. While these may have a 1/4-inch plug on each end, the loudspeaker cable is a different animal altogether, designed to move large amounts of output current from an amplifier to a loudspeaker, not the mere milliamps that signal cables handle. Constructed of two heavy-gauge inner-insulated conductors housed in an outer jacket, these cables are commonly used to connect a stage amplifier head to its loudspeaker cabinet, or a small PA loudspeaker to a powered mixer.

Just a reminder—signal cables should never be used for loudspeaker lines, and vice versa. Signal cable isn’t designed to handle high current, and loudspeaker cable is not shielded from outside interference.

Scaling Up
The most popular loudspeaker connector in pro audio is the Speakon (stylized as speakON) from Neutrik. They come in 2-, 4- and 8-pin varieties, allowing a multitude of connections options. The wire size (gauge) of loudspeaker cable depends on a few factors, chiefly the load impedance and the length of the cable. Simply put, the longer the cable, the larger the conductors should be. Common sizes for audio production include 12- and 14-gauge, with a few manufacturers also offering multi-conductor cable in 13-gauge.

Two 50-foot, 6-channel boxes to fan stage snakes.

Some sound companies deploy an 18-conductor cable with a 19-pin connector called a Soco, borrowed from the lighting world. The term Soco comes from the trade name of the most common 19-pin connector manufactured by Socapex, but companies like Veam and Kupo also make compatible connectors. Lighting folks use the cable for six circuits of power, while audio folks wire up their systems differently and can get up to nine speaker circuits in one cable. A “Soco to fan out” distributes signal to the various loudspeaker cabinets.

Speaking of multi-circuit cables, snakes are the answer for running multiple channels of audio from one place to another. These cables could have a breakout fan on one or both ends to individual channel lines, or could use a box at one end (usually at the stage end) that individual XLR cables can be plugged into. Snakes can also integrate multi-pin connectors that make it faster and easier to hook up a system. To save weight and size in the cable many snakes use a foil shield around each pair of channel conductors instead of a braided or spiral wrapped shield.

Aside from the obvious stage to FOH mixer application, smaller “stage” snakes are a popular way to help manage cable runs on stage and keep things neat and organized. Another use of snakes is for “crosslink cables” running the signal to the PA system from one side of the stage to the other.

Many snakes have the capability to run signals from and to the stage. The “sends” are for the mic inputs to the mixer and the “returns” get the output of the mixer to the amp rack or powered loudspeakers. Larger systems may use a separate return snake for the line-level outputs to keep any crosstalk (interference from adjacent snake channels) to a minimum.

Splitter snakes provide more than one output off the send side of a snake, so the same inputs can be sent to multiple consoles (i.e., when using a separate monitor or broadcast console along with the house console). Some splitters are passive and simply hardwire a “Y” off each channel.

A better practice is to use isolation transformers to isolate each console from potential noises and hums and buzzes caused when plugging them into different power sources. In a split snake system, usually one split is hard-wired to the inputs so that the console can pass phantom power to mics and DIs.

Another version, called a “power snake,” combines a few loudspeaker lines along with the signal channels. These can work well for a small system on short runs but their use is usually limited to about 100 feet.  Yet another multi-circuit version that has become popular recently is cable systems that include signal and power in one jacket. These are perfect for getting audio and AC power to a powered loudspeaker or floor wedge.

Networked World
While analog cables still fill the road trunks, digital systems are starting to take over many of the audio transport duties. They offer a host of signal routing benefits that analog simply can’t match, including using a small thin cable to route multiple channels of audio. Smaller cable equals less stagehands required to lay out a digital network as opposed to large, heavy multi-core snakes.

Digital cables are also less prone to RFI and crosstalk. Networks, as we now call our digital transport systems, can offer audio almost anywhere along the line, and can easily interface with multi-track recording systems, personal monitoring rigs and broadcast trucks.

Transport networks use one of three types of cable: fiber optic, coax or Ethernet Cat-5/6. Coax cables offer up a rugged solution and are used by a few manufacturers to transport signals between stage boxes, consoles and recorders. Fiber optics offer the ability to send signals over very long distances, and because the signals travel as light, are immune to all outside electromagnetic disturbances and RFI. Ethernet Cat-5/6 cables are the most popular, found in many different systems to transport audio at distances of up to about 330 feet (100 meters). Some of these have accessories that can extend this distance.

A typical molded RJ45 connector (left) with an Ethercon connector.

Ethernet cables have RJ45 8-pin connectors that are stout enough for home computer use but not rugged enough for most gig uses, so they’re best replaced with rugged Ethercon connectors that surround the plastic crimp-on with a metal barrel that provides added protection in addition to better locking.

Ethernet cables come in a variety of styles. Some have solid wire conductors that offer the best performance, while others have stranded conductors that provide greater flexibility. They can be unshielded but it’s better to go with shielded in noisy environments.

Ethernet cables can also be wired in different ways. The “standard” wiring scheme runs pin 1 to pin 1, pin 2 to pin 2, etc. “Crossed over” cables wire pin 1 to pin 3, pin 2 to pin 6, pin 3 to pin 1 and pin 6 to pin 2. Before choosing an Ethernet cable, check manufacturer recommendations on which cable is recommended for interconnection of specific gear.

As noted earlier, a downside to digital transport is that there is the need for analog to digital conversion, and further, manufacturers utilize a variety of variety of different protocols that are not compatible. However, that’s been changing rapidly, as more and more devices support multiple protocols, and the Audinate Dante protocol in particular has really caught on the past few years. And, AVB (Audio/Video/Bridging) is a standard that manufacturers are also starting to embrace. The bottom line is that our job of interconnecting various gear from various manufacturers is getting easier.

Senior contributing editor Craig Leerman is the owner of Tech Works, a production company based in Las Vegas.

{extended}
Posted by Keith Clark on 04/04 at 04:16 PM
Live SoundFeatureBlogStudy HallAnalogDigitalEthernetInterconnectNetworkingSignalSound ReinforcementPermalink

Wednesday, April 02, 2014

Church Sound: Setting Input Gain Structure

The keys to an often underrated aspect of proper operation of a system...
This article is provided by ChurchTechArts.

 
Gain structure is one of those very important, yet highly underrated topics in audio. It’s not nearly as glamorous as EQ, plug-ins or parallel compression, but if your gain structure is whack, no amount of EQ, plug-ins or compression will fix it.

Here I’m going to focus primarily on input channel gain structure (overall system gain structure is another article entirely, but I’ll mention it briefly). The impetus for this article came from a simple question: Is it better to hit the preamps hard then turn down at the main output, or run the mains up around unity and dial back input gain to get the SPL you want out of the system?

As a general rule (there is an exception, which I’ll detail in a minute), I would argue the former is the correct (or at least better) method, and here’s why. Most preamps sound best when you hit them pretty hard (at least up to the point of clipping, which is too hard). By running preamps hard—and by hard I mean around -6 dB full-scale on a digital board, or within 6 dB of clipping on an analog board—you are maximizing signal-to-noise ratio.

And for some reason, they just sound better. Keep in mind, that’s a general rule, your mileage may vary. Now, it’s quite possible that if you dial the input gains up so that all of the preamps are running high, the overall system level will be too high. 

That’s when you would lower your main level to compensate. This method will keep the signal to noise ratio high throughout the mixing chain, and will attenuate the signal at the last possible moment.

Before we get to setting up the gain structure, let me lay out my goals in for the process.

First, I want to maximize S/N ratio, and use up as many of the bits in the analog to digital (A/D) conversion process that I can. Keeping the input level high meets both goals.

Second, I like to mix with my faders around unity. Mixing with faders at unity is another key ingredient to good mixing. The fader resolution is highest right around unity, so it’s easy to make small adjustments.

If you try to mix with your faders at -20, a slight change in fader position might yield a 3-5 dB change rather than the 1-2 dB you actually desire.

Finally, I want to be sending a very solid signal out of the mixer to the processors for the same reasons (only in reverse) as the first point. That’s why proper system gain structure is important.

Next, how I would approach the process.

Gain Setting In A Digital World
For each input channel, I would have the musician play their at their loudest level.

I then dial up the input gain until I’m within about 8-12 dB of full scale (minus 8-12 dB on the meters). I like to leave a little room for the musician to play louder when the lights go up (they always do).

Many digital boards also have a trim (or attenuation) control in addition to the input gain. I use my trim to dial the level back to where it should be in the mix with my faders at unity.

Because I’ve gained my entire system properly, my main fader is sitting at unity as well, and all is right with the world. As I’m using VCAs to manage groups of faders (drums, guitars, keys, background vocals, etc.) and those live at unity as well, at least to start.

All of this ensures that my signal-to-noise ratio is optimized at the A/D stage (just after the mic preamp), and my starting point for my mix is faders at unity.

Now, if you don’t have a digital trim control on your board, there’s a decision to make. You won’t likely be able to run the mic preamps hard without having too much signal at some point, so it’s necessary to dial the level back somewhere.

Digital gain.

Of course, you can always turn the fader down, but then you lose fader resolution. A better alternative would be to use a VCA to keep the fader at unity, though that can get tricky.

Take a drum kit for example: If you optimize the gain on the kick, snare and hat, chances are, the hat will be way too loud in the mix. But more than likely, you’re using a single VCA for the entire drum kit. So now what?

Well, you could break the drums up into zones and use one VCA for each—kick and snare, toms, hi-hat and overheads might work. That way you can pull back the faders at the VCA level (a VCA is really an electronic remote control of the faders), and maintain fader resolution. A similar trick can be done with groups if you have them.

If VCAs are running, break my rule and set the input gain up so that the fader remains around unity for a proper mix. Audio is a lot about compromise, and in this case I’ll give up absolute input S/N to run my faders at unity. I’ve found that to be the wiser trade.

Gain Setting In An Analog World
Really, the process is much the same, though you are much less likely to have a trim control after the gain control.

In that case, the same rules apply as a digital board without a trim knob. You still want to have good input level coming into the channel (for the most part), then turn it down as needed later in the mixing stage.

You also want to keep your faders running around unity. Make the trades where you have to. In either the digital or analog world, what you don’t want to do is underdrive the mic preamps and have to add a lot of gain down the road.

Sure, you can push a fader up for a guitar solo, but you don’t want to regularly run input faders at +8, groups at +10 and main at +5 because the input gain is set too low.

Exception To The Rule
Now, all of this assumes you’re running on a professional grade mixer that has a mix structure designed with proper headroom.

If you’re using an inexpensive mixer, chances are you’ll run out of headroom in the mix bus very quickly. Setting input gains on these mixers the way you should, when all those hot signals hit the mix bus, is not going to be pretty.

Analog gain

The buses quickly saturate and lose all sense of dynamics. In that case, really keep an eye on overall output level and run input gains down accordingly.

This isn’t a dig on cheap mixers—they fulfill a need but you can only expect so much for what you pay for them—it’s just reality.

Conclusion
That’s a quick guide to setting up your gain for an input channel.

As I mentioned earlier, if you go through this whole process only to find that your overall SPL in the house system is either way too loud or way too soft, you have some work to do at the system processor or amplifier level.

But that’s another article entirely…

Mike Sessler is the technical director at Coast Hills Community Church in Aliso Viejo, CA. He has been involved in live production for over 25 years and is the author of the blog Church Tech Arts.

 

{extended}
Posted by Keith Clark on 04/02 at 01:32 PM
Church SoundFeatureBlogStudy HallAnalogConsolesDigitalEngineerMixerProcessorSound ReinforcementTechnicianPermalink

Two SSL Duality Consoles Power The Creative Flow At Tape Studios

Two cnsoles are critical to the sonic success at new Edinburgh recording/mixing facility

Two Solid State Logic Duality SE consoles power the creative flow for Tape Studios in Edinburgh, UK, a facility described by alternative record producer/mixer Stephen A. Watkins as “built to boldly go where no studio has gone before.”

The new two-story recording/mixing facility offers an equipment list and acoustics that match top facilities around the world while being able to deliver the creative experience without excessive cost. The Duality consoles are critical to the sonic success.

“We wanted to put together a studio that was completely over-the-top to match my personal recording/mixing style,” says Watkins. “The setup features an all-guns-blazing, maxed out patchbay handling a multitude of ‘who’s who’ analogue outboard gear. Duality was the perfect choice for Tape Studios because it brings all the elements we use together, while still addressing a DAW workflow. Even when I push them to the limits, which I very often do, the consoles sound incredible.”

Officially opened in February 2014, Watkins and partner, Fiona Mcnab, found an old Victorian whiskey bond building and converted it to a world-class facility with design help from Munro Acoustics. Tape Studios was conceived to provide a no-holds-barred creative environment to support a wide range of clients, including the hot new Scottish band BooHooHoo.

“We wanted to build a studio that was filled with the technology everyone dreams about, and that included the two Duality consoles,” Watkins adds. “I had this sound in my head and I knew how to make it. There is absolutely nowhere in the country that could provide the kind of facility at which I could make the records I wanted to make the way I imagined them.

“A genuine forte of mine is blending a little chaos into the banality of today’s beige multi-track recordings,” he continues. “I thrive on dynamic range manipulation by patching in multiple compressors instead of just one. I love envelope tickling and things that surge toward you from deep within the noise floor. A little Scottish swing, if you will, instead of safe boring everyman grid life. Duality provides the sonic foundation that supports my creative efforts.”

The ground floor of Tape Studios houses Studio 1 with the 48-channel Duality driving a Studer A80 24-track analog machine and a Cubase/SSL Alpha-Link rig. Studio 2 features a 96-channel Duality in a unique, custom built “L” shaped configuration. Everything in the facility is connected together so the resources of Studio 2, for example, can be used for tracking in Studio 1. Duality also offers a “wow” factor for clients coming through the doors.

“Our studio is quite new, but the reaction by the creative community thus far has been nothing short of fantastic,” Watkins explains. “Every single time anybody walks into any room at Tape and sees Duality, nothing more needs to be said. When we first started working, Duality became invisible very quickly, delivering exactly the benchmark sound we needed.

“When I began doing my thing, Duality was absolutely HQ. The line amps pushed hard into the channel dynamics post EQ is just heaven. Plus Duality plugs into the wall direct on only a couple of IEC cables. No machine room. No extra cooling. Duality is a no brainer – end of story. So we bought two.”

image

 

Solid State Logic

 

{extended}
Posted by Keith Clark on 04/02 at 09:08 AM
RecordingNewsAnalogConsolesDigitalEngineerMixerStudioPermalink
Page 2 of 40 pages  <  1 2 3 4 >  Last »