Analog

Thursday, May 12, 2016

Sonic Nirvana? Thinking Outside Of The “Technically-Oriented” Box

Whenever I’m at the local Guitar Hut, I like to listen to the people who come in and talk with the pro audio sales guy about gear. These conversations are often filled with nebulous audiophilic adjectives like “warm”, “sweet” and “punchy”.

The sales guy has little motivation to be a source of truthful or accurate information. He just wants to make a sale. Meanwhile many of his customers already have their minds made up as to what piece of gear they need, and why. 

It’s fairly easy to pick out those who will make a purchase and install it in their system - and then, in time, become disillusioned enough to again pick up the quest for the next piece of gear that promises sonic nirvana.

After more than 30 years of work with professional live and recorded sound, I find it unfortunate that so many are trapped in this scenario. Collectively, we have yet to reach a uniform level of conceptual awareness about sound systems and ways of attaining excellent results because of a fixation with gear.

For many years, I was bound, seeing just individual trees. Fortunately, Bob Brooks helped me to see the rest of the forest.

Bob came up back in the heyday of 1950s broadcasting, has been extensively involved with both live and studio production, and for 10 years owned one of the most successful studios in western Canada, Little Mountain Sound. 

I met Bob eight years ago, and wish that I’d met him much earlier in my professional development. A true mentor, Bob has pushed me to hear and think outside the “technically-oriented” box that traps so many of us. We easily fool ourselves into believing that because the technical issues are “technically” correct, the sonic issues are “sonically” correct. 

Even when we’re absolutely sure our ears are telling us that something is amiss, we still deny and defend, even to our own demise. 

I like gear, but now recognize that if I release my inner Tim Taylor, I’ll end up sitting on the couch in my underwear surrounded by boxes of Class A this, digital that, and tubes galore, giggling like Beavis & Butthead. 

Sorry, it’s best not to go there.

Since Bob helped enlighten me, my personal “key” to achieving consistent, reliable and (pardon the lack of modesty) excellent results boils down to this: it’s not what you’ve got, but how you use it. 

I’ve learned to be careful in judging the provenance or status of the tools at my disposal., and have discovered that my preconceived ideas have an influence on my own success or failure. That’s not the fault of gear. 

So I’ve adopted the view that I can successfully use any piece of equipment as long as it has a sufficiently low noise floor, appropriate headroom and an absence of sonic “funkiness”.

Anything beyond these factors is lagniappe (lan yap), a Cajun word meaning “something extra”. 

The problem with lagniappe is that it tends to make us fat, or more specifically, bloats our thinking. Lagniappe promotes the welfare mentality.  It leads us to believe that we can’t just make do with the bare necessities, and lagniappe belies the simpler truth: when it comes to producing quality sound, less is usually more, less is usually better. The more we add, the more chance we have of screwing it up.

In early recording and broadcasting, consoles only had one way to control volume on each channel, and that was the gain adjustment. What? Mixing via the gain knob?? Yup. Simple and effective. Either it was right or it wasn’t, and there was only one place to make it so.

Contrast that with modern consoles often providing four or five gain stages that have a direct influence over the level of the output. Sweet. In the right hands. (And conversely scary to the wise.)

The problem is that along the way, yesterday’s techniques for excellence have been lost on so many of us. We don’t come to this field equipped with solid production technique, and then we’re presented with so many choices.

Again: the more we add, the more chance we have of screwing it up.

There’s hope, however. We just need to embrace the dark side. In other words, look at our habits and admit that what we’re doing might not be producing the results we desire. Accepting this fact is the first step to moving on to a much better direction.

The most basic key to building excellence is to learn good technique in simplicity, and then evolve it as things get more complex, and as understanding increases.

I’m betting that at least a few of you are ready to embrace some “revolutionary” thinking and methods. The fun part is that the foundation of this revolution is largely based upon proven and reliable, not new and improved.

Since his start more than 30 years ago on a Shure Vocalmaster system, James Cadwallader remains in love with live sound. Based in the western U.S., he’s held a wide range of professional audio positions, including live mixing, recording, and technician duties.

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Posted by Keith Clark on 05/12 at 06:02 AM
Live SoundFeatureBlogOpinionAnalogBusinessEngineerMixerSignalSound ReinforcementTechnicianPermalink

Wednesday, May 11, 2016

Streaming & Casting

As audio professionals, we’re usually not concerned with how the content of podcasts and webcasts is delivered. Our focus is getting quality audio to the recorder or computer and making things sound their best.

I categorize casting and streaming into two basic groups: speech gigs and musical performances. Typically, the web conferences and corporate podcasts that I work consist mainly of speech with some pre-recorded music thrown in. Music performances are a bit more complex and I’ll address how to approach those a little later.

Important Issues
Like everything else we do, getting it right starts with pre-planning. I begin by asking dozens of questions, starting with the number of presenters and where they’ll be located, followed by queries about things like whether there will be Q&A (question and answer) microphones in the room, will there be remote audio to interface (such as from a telephone or voice over IP), will there be computer audio payback or music, and so on.

Another key question is what type of interface my audio signal will be feeding and where the interface will be located so I’ll know what type and how much cable is needed for the feed. This is also the time to address issues like additional power requirements for the recording and computer gear, who is operating the computer equipment, who is providing internet connections, etc. Only after I have a firm grasp on the situation do I devise an audio plan for both the house sound and web feeds.

For smaller input events, it’s usually a good idea to use a second mic on each person and at the lectern/podium, in order to provide a totally separate audio feed for the webcasts. Using a separate feed has the advantage of isolating the web audio feeds from any hum or noise issues that may arise when interfacing recording and computer gear with the PA system.

It also allows locating the webcasting components away from front of house or monitor areas to a more quiet location and/or closer to the computer and internet connection. For events with a large number of inputs, the choice is one mic per input and then splitting the signal down the line for the web.

If presenters are wearing lavalier mics for the house mix, I may place second lavs on them, on double clips, for the web feed. I also usually outfit them with a lav if they’re wearing a headset or ear worn mic. For recording and web feeds, omnidirectional mics are usually the best choice because they pick up sound a little more evenly when people turn their heads.

At the lectern/podium, I go with one of three options. The first is adding a second podium mic for the web feed, the second is to “gaff” a small mic under the podium mic’s element, and the third is placing a lavs on the presenters.

If they’re seated at a table and the house system microphones are on desk stands, a solid approach is placing ear worn mics on them. Many of them tend to lean into the house mics to talk, and a lav clipped to their chests will either get bumped into the table or pick up a lot of reflections from the table surface.

But if the input count gets too large, I simply split the signal from the house mics for a web feed. It can be set up as a separate mix on the house or monitor console, or the inputs can be split out of the console into a dedicated console for the web feed. 

Further Ideas
For podcasts or on-demand webcasts that won’t be going out live, there’s also the option of multi-track recording. In fact, even if an event is live streaming, I multi-track the show and make a safety recording “just in case.”

Since my digital consoles have Dante networking, I just interface a computer loaded with a Dante Virtual Soundcard and use a DAW (my favorite is Reaper). This can be up and running in minutes, connected via a single Cat cable.

For speech-only gigs, it’s a good idea to utilize mics that have a low-cut filter that will roll off the bottom end. (Or use the “voice” setting.)

On the console, rolling off the lows at and below the 80 to 100 Hz range on each mic input helps reduce any proximity effect and low-frequency stage noise. Also consider windscreens on every mic to help tame any pops and plosive noises, and make sure isolation feet on mic stands are in place.

I also employ isolation clips to lessen the chance that the mic picks up noise transmitted from the stand. One of my “tricks” is to place rubber mouse pads under the stands of table mics to further isolate them from noise. I put the smooth side down so the stands can easily slide out of the way of the presenters if they prefer.

Keeping It Smooth
Musical acts are a bit more challenging. One of the biggest noise issues is stage amplifiers that hiss, buzz and/or hum. These noises might not be too bad live but on a recorded webcast they can be very apparent.

The buzz and hum are indicators of a grounding issue. My first move is to make sure the amplifier has a ground pin still on the plug, or that it’s not plugged into an extension cord that’s missing the ground.

Also check any ground switches on the unit. If these measures don’t solve the problem, I plug the amp into a different outlet (even if it’s offstage).

Another thing that can help is changing the location of the amp’s mic, moving it farther away from the loudspeaker (the audible source of the noise). 

Musical acts can also have a wide variety of dynamic levels. Compression and limiting can help keep instrument and vocal volume levels in check for the webcast mix.

For the main outputs I deploy a leveler – while compressors and limiters work great at keeping things from getting too loud, they can’t compensate for things that are too soft. A leveler allows setting a target window for the audio, compressing loud sounds and raising the gain of soft signals to keep the overall signal in the target range.

Some consoles offer leveling or accept leveling plugins, and there are also hardware models available, including the “old” Symetrix 422 stereo AGC leveler that I use. Inconsistent volume levels can frustrate the listener, causing them to constantly turn up or down the volume and resulting in a mediocre listening experience. Most if not all broadcast stations employ leveling, and that professional sound is what we strive to achieve.

As their name implies, levelers can help keep things consistent, whether in software (Waves Vocal Rider, above) or older hardware like a Symetrix 422 stereo AGC leveler that the author uses.

Checking It Out
Unless there’s a separate audio mix system for the web feeds, it’s a good idea to place isolation transformers between the sound system and the computer(s). While there may be no noise issues at sound check, they have a bad habit of suddenly appearing before show time, so the isolation will help keep the signal clean without demanding your attention when other issues are more pressing.

The next stop in the signal chain is the front end of the computer. There are a lot of FireWire and USB audio interfaces on the market today that sound great. I have several Focusrite Scarlett interfaces that work well and allow some basic metering and monitoring capability.

Monitoring the mix is very important because you want to hear how it sounds to the average listener, who will probably be using a laptop or desktop with small loudspeakers or a portable device like a phone or tablet with earbuds or headphones. While I may mix using good headphones or quality monitor loudspeakers, I monitor my feeds with a set of cheap earbuds as well as a laptop – using the built-in loudspeakers – so I can get an idea of what things will sound like on the user’s end.

With a little extra attention to details and an understanding of what the audio sounds like over small, low-fidelity loudspeakers, it’s a pretty straightforward matter to deliver professional sounding podcast and webcast sound.

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

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Posted by Keith Clark on 05/11 at 11:33 AM
Live SoundFeatureBlogStudy HallAnalogConsolesDigitalInterconnectNetworkingProcessorPermalink

Audient Selected For Santa Fe’s Kitchen Sink Recording Studio

Jono Manson and Tim Schmoyer renovate studio and add ASP8024 analog console with Dual Layer Control to vintage gear collection.

Opened at the end of last year, Kitchen Sink Recording Studio is the product of audio engineers Jono Manson and Tim Schmoyer’s pooled talent and resources. In a brand new location in downtown Santa Fe, the ‘new’ place is actually an old studio, where they were lucky enough to keep and renovate some of the equipment left behind by previous owners.

Adding an Audient ASP8024 with Dual Layer Control (DLC) into the mix has enabled them to benefit from the best of both worlds, combining all the advantages of DAW control and automation with classic analog gear.

“My old studio already had a nice complement of gear, all of which migrated to the new location,” says Manson , who had owned the original Kitchen Sink Recording Studio just north of the city, for the previous 10 years. “I did purchase some additional gear, mostly in the form of dynamics, and we also inherited some gizmos from the old studio which existed in the space before we took over. Among these items are fairly impressive complement of vintage microphones which, added to what I have in my closet already, comprises a formidable array.

“Two analog tape machines (both 24 and 2 track) came with the new place. These have required some love and attention, but they’re back up and running and humming right along. We also replaced every inch of cabling in the entire facility, and every solder point on every jack on every panel has been redone.”

Yet how did they come across the board from British audio company, Audient?

“Time after time I heard the familiar refrain that these were extremely reliable, great sounding desks, that the EQ was very musical, and that the preamps offered an incredible amount of bang for the buck. So, Tim and I flew to New York so that we could get our hands on one, in the flesh. I ran the desk through its paces and pushed it hard (maybe harder than I ever will in my own studio, but don’t tell anyone), and from that moment on that we were sold.

“The desk is extremely well designed, and very intuitive,” continues Manson. “The routing is extensive and very flexible. With the exception of a couple questions regarding the functions of the DLC, I have not had to crack the manual once since the desk has been up and running. I’ve been spending 12 to 15 hours a day in front of the ASP8024 and I can now honestly attest to the fact that, whether in tracking or mixing, all of the glowing endorsements were 100% true. So, kindly add mine to the list.”

“We contracted Pro Audio Design in Boston to conceive and execute our new patch bay. Apart from all of our outboard gear and our DAW and tape machines, every input, output, group, insert on the Audient comes up in the bay and we can easily assign tape sends (everything’s normalled when it needs to be) to either ProTools or the 2-inch machine - or both, simultaneously.

“In short, it looks, feels and sounds great.”

Together Schmoyer & Manson have built one of the highest spec’d studios in the region, so it’s no wonder that’s where Amanda Palmer chose to record vocals for her recent Bowie Tribute. Perhaps this place really does have everything - including the kitchen sink.

Audient

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Posted by House Editor on 05/11 at 10:36 AM
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Monday, May 09, 2016

Radial Engineering Now Shipping mPress And Exo-Pod Press Distribution System

Analog system allows distribution of the signal to as many as 112 users with phantom power and balanced connections.

Radial Engineering announces the mPress and Exo-Pod press distribution system in now shipping.

The mPress is a new press audio distribution system that offers near unlimited signal expansion without degradation or noise.

According to Radial’s senior engineer Dan Fraser: “The traditional press box has always been presented as a suitcase with a mic input and a host of mic outputs. In the past, transformers performed the task of splitting the signal and providing isolation against hum and buzz caused by ground loops. In recent years, the use of active signal buffers has replaced the transformer as a means to cut manufacturing costs. Unfortunately, this has resulted in noise creeping back into the audio system, deteriorating the audio quality. The mPress solves the problem by combining a high-octane active drive circuit with a host of transformer-isolated floor boxes called Exo-Pods. This modular approach allows the system technician to distribute Exo-Pods throughout the press gallery while assuring each member of the press receives a clean, hum-free signal. To ensure no digital trickery is at hand, the mPress is 100% analog.”

The mPress is configured in two parts with a master host - the mPress - and a series of external slaves called Exo-Pods. The mPress is housed in a standard 1RU 19-inch rack enclosure and begins with two mic inputs, both of which are equipped with a variable level control high-pass filter to eliminate excessive resonance and tame the proximity effect.  For podium mics, 48V phantom power is available and activated using a front panel switch. This is recessed to prevent accidental use. Selecting between the mic channels is done using front panel switches. In order to control ballistics from overly excited orators, the mPress has been outfitted with two easy to use knob limiters for threshold and release.

To accommodate ‘walk-in’ music, the mPress is equipped with ¼-inch, RCA and 3.5mm stereo inputs along with a separate level control. This enables the mPress to be connected to a couple of powered speakers via the main stereo XLR outputs to provide background entertainment while the gallery waits for the ‘talent’ to arrive. A headphone output on the front panel is available for local monitoring and trouble shooting.

There are eight (8) specially designed balanced outputs, with two on the front panel, and six on the rear. Each of these may be configured using a recessed switch for mic level or high output drive to feed an Exo-Pod. This allows the mPress to be used as a 2x8 distro for smaller events or expanded by adding Exo-Pods on the outputs. The Exo-Pod is a passive floor box that features an XLR input, a throughput for expansion, a local level control plus ten (10) XLR mic outputs and four (4) 3.5mm mini TRS outs for those who are equipped with a mini recorder. A test tone may be activated to set local and master levels. With the use of eight Exo-Pods, one can distribute the signal to as many as 112 users. This can be further expanded using the throughputs by simply adding more Exo-Pods. Power is only needed at the main mPress box and is supplied via an external lock supply that will accept any input from 100 to 240 volts.

MAP: $1,099.99 USD for the main unit and $279.99 USD for each Exo-Pod module.

Radial Engineering

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Posted by House Editor on 05/09 at 07:41 AM
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Friday, May 06, 2016

PSW Top 5 Articles For April 2016

ProSoundWeb presents at least two feature articles every day of the working week, meaning that there are 40-plus long-form articles highlighted each and every month.

That’s a lot. In fact, so much so that we got to thinking that it would be handy to present a round-up of the most-read articles for those who might have missed at least some of them the first time around.

What follows is the top 5 most-read articles on PSW for the month of April 2016. Note that since the articles aren’t all posted at the same time, we apply the same timeframe (length of time) for each when measuring total readership.

Also note that immediately following the top 5, PSW editor Keith Clark offers some additional suggestions of recently published articles worth checking out. These articles also scored quite well in terms of readership but were just outside the head of the list.

Without further adieu, here are the top 5 articles on PSW in April.

1. What Is Dither?
How can intentionally adding noise to our audio signal ever be a good thing? (Includes Video) By Nigel Redmon

2. PA Design For Coverage & Intelligibility
Four design principles to consider when looking at a main sound system loudspeaker design. By Mike Sessler

3. Gig Savers
An in-depth primer covering key interconnect and test tools (and more!) that make it all work. By Craig Leerman

4. Thickening Up Tracks With Doubling
Techniques for creating a doubling effect to mimic a “stacked” sound without multiple takes. By Scotty O’Toole

5. Loudspeaker Advancement
The evolution of large-scale sound system optimization, in the first of a multi-part series. By Bob McCarthy

Editor Recommendations

Something In The Air
Accounting for environmental and other changes between sound check and show time. By Dave Rat

Foolproof Festival Patch
“Why not put the mics where the people are going to end up?” A a straightforward approach… By Ike Zimbel

The Power Of The Unseen
Excelling at the invisible side is one of the biggest ways we can build a quality visible side. By Andrew Stone

Room Treatment Vs Soundproofing
Differentiating between the two types of improvements that can be made to an acoustic environment. By John Calder

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Posted by Keith Clark on 05/06 at 03:29 PM
Live SoundRecordingChurch SoundFeatureAnalogDigitalEngineerSound ReinforcementStudioTechnicianPermalink

MULANN Launches Recording The Masters Analog Tape Brand

New brand of professional recording, mixing, mastering and archiving media expands market in the United States, Europe and other territories.

MULANN Group announces “Recording The Masters” a new brand and identity for its AUDIO professional activities. Creating a new visual identity for the audio products with a new logo reflects MULANN’s commitment to professional audio recording and strengthens its presence on the worldwide music, archive and instrumentation markets.

“MULANN owns the original formulas of analog recording, some of which date back to 1950, created by AGFA and BASF. These magnetic formulas deliver a very high sensitivity and dynamic sound quality. They also offer the capabilities to store data for several decades, far beyond what digital and optical media offer today,“says Jean-Luc Renou, MULANN CEO.

Oriented to professional and semi-professional recording, mixing, mastering and archiving, MULANN is expanding its position in this market in the United States, Europe and other territories.

With the introduction of this new brand, MULANN products speak directly to both the professional sector and to audiophiles. The analog recording tapes created by AGFA,  BASF then EMTEC, are now manufactured by MULANN group under the brand “Recording The Masters”. This new visual identity energizes the market searching for sound authenticity, sustainability and the original technical qualities of audio recording developed in the mid-twentieth century and never equaled over the years.

“The visual brand identity “Recording The Masters” is radically new,” explains Renou, “This new brand identity is dynamic, modern and it indicates that analog recording is active more than ever. Its technology values offer a future and an important role to capture, transmit and store audio tracks and sounds for the long term. The analog recording flame shines even brighter for an everlasting light.”

MULANN Group

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Posted by House Editor on 05/06 at 10:46 AM
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Wednesday, May 04, 2016

Leona Lewis Tours UK With Solid State Logic

Front of house engineer Dave Wooster selects the SSL L500 console for 14-date tour.

Soulful, multi-talented vocalist and performer Leona Lewis has been on a 14-date tour of the UK with a Solid State Logic L500 console and K-array Slim Array technology, chosen by front of house engineer Dave Wooster.

Lewis’ I Am show provides Wooster with around 65 inputs from the stage. “It’s a full-on set-up,” he says. “The drummer alone has two mics on the bass drum, three snares - each with at least two mics on, four toms, all the cymbals, electronic kick and snare… So he was up to about 20-odd channels on his own.”

According to Wooster, the show absolutely benefited from the L500’s signal path - from the SuperAnalogue mic inputs, through the flexible channel path, and comprehensive internal FX Rack: “What really separates the L500 from the competition is the sound.”

“The effect on Leona’s vocal was very noticeable in the system,” he continues. “I think the 96kHz operation makes a difference, but the pre-amps make a huge difference as well, and whatever it is SSL has done on the EQ is stunning…. You really hear the HF.

“With Leona I have to deal with a massive dynamic range within every song… The mix has to be able to go right down to almost nothing and then build to everything. The way the SSL input section handles that is fantastic.

“Of course, it’s natural that when she whispers I get a load of low end from the microphone that I don’t need, and when she’s screaming down it there’s too much high end and not enough lows. I use a dynamic EQ from the internal FX rack to sort that out. The standard EQ helps calm down some resonances, though there were only two cuts with low and high pass filters that I needed to make with that.

“The channel compressor is the first layer of dynamics control, just to help take out any real big peaks; then across her stem- which includes her reverb and delay returns, as well the main vocal path - I put an SSL Bus Compressor; it’s very good.”

Wooster’s approach to the console surface configuration takes full advantage of the L500’s layer & bank approach to layout, as well as the Super-Query (‘Q’) function - a forward and reverse interrogation feature with fast-assignment feature.

“I have all my input channels as a sub-layer,” says Wooster. “That’s where all the programming is. Then I use Stems on the top layer. I completely isolate them from any recall and end up with kick, snare, hat, toms, overheads, a bass channel, guitar channel, keyboard channel, lead vocal and BV stem faders that are always below my fingers… That’s my mix.

“All the automation and scene recalls are still going on underneath, so if I then hit the Q button on any Stem all of the underlying contributions pop up from below. I can make a quick adjustment in that scene, save it, then go straight back to the Stem layer and carry on mixing.”

As well as the console, the tour rig included the innovative K-array ‘Firenze’ Slim-Array PA system with acoustic steering. Wooster is convinced that the combination of the two was unbeatable: “In the 33 years I’ve been doing this,” he says, “I’ve never mixed on a system this good…”

image
Front of house engineer Dave Wooster at the SSL L500


Solid State Logic

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Posted by House Editor on 05/04 at 10:27 AM
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Kingston College Outfits Creative Industries Centre With Audient

UK school selects ASP4816 and ASP8024 analog consoles to compliment additional Audient gear in new facility.

Boasting two brand new studios, fully kitted out with two Audient analog mixing consoles, two iD22 audio interfaces and three 8-channel mic pre ASP880s, Kingston College’s Creative Industries Centre was opened at the beginning of this academic year.

As they advance into their second term, Music Technology instructor and studio technician, Chris Winter has noticed how the students have been enjoying the recent studio upgrade.

“They love using both the ASP4816 and progressing to the ASP8024 with Dual Layer Control,” he says, pointing out that they find them “easy to understand and clearly laid out. They love that analog sound, too.” Whilst the desks are located in Studios 1 & 2 - the commercial and the primary teaching studio respectively - the outboard gear is predominantly used in the four production rooms that are linked to the rehearsal and live rooms.

“The students love the way iD22 allows you to assign a mic pre as talkback mic. The ASP880s are also really useful when you want to DI an instrument quickly.” He continues with his own thoughts on the new Audient outboard gear: “The iD22s are so easy to use and have great, big console features. The ASP880 are amazing too, and really work well with the iD22s. The idea was that we have the same mic preamps across the facilities so there is continuity in the sound. And we love them.”

Winter describes the new building that houses the studios as “…an amazing space. We are very lucky to have the opportunity to build industry standard studios to such a high specification, and work with great companies like The Studio People on getting wonderful sounding rooms that look fantastic too.” Comprising a 3D workshop, TV studio, as well as a mixture of studios and classrooms designed to enhance the learning experience, the Creative Industries Centre is designed to encourage cross collaboration on projects with other courses taught at Kingston College. “The students love the studios’ design with all that natural light; it has a big impact on their learning and creative flair,” says Winter.

“We teach from Level 2 up to level 5 courses in Music Technology and Performance,” he explains. “The great thing about these Audient desks is that for beginners right up to advanced users, they are easy to understand with no gimmicky features.” Winter is very clear that teaching signal path to students is “…fundamental to their training. So many people have tried to teach ‘in the box’, but have reverted to analogue board as a physical piece of equipment that you can touch and see where the signal starts and ends up.”

Indeed all the new technology in the Creative Industries School helps with that. “Audient has high quality products that are easy to use and are at a professional level. We wanted our students and users to learn on industry standard equipment, so they can transfer their skills learnt at Kingston College straight through to employment.”

Audient wishes staff and students all the best with the new facility.

Audient
Kingston College

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Posted by House Editor on 05/04 at 09:48 AM
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Meter Madness: What Your Level Meters Tell You And What They Don’t

This article is provided by PreSonus.

 
In May 2014, the VU meter celebrated its 75th birthday. It has served the industry well, and when properly interpreted, it’s still useful.

However, today’s digital recording processes have caused us to take a hard look at the usefulness and inadequacy of both the traditional VU meter and its modern replacement, the LED level meter, as tools for signal-level management.

The classic VU meter, though relatively rare today (primarily because of cost), has long been the most common audio-level indicator. VU stands for Volume Unit, and the VU meter indicates how loud something sounds.

The traditional VU meter is mechanical, analog, and has a standardized (even the color scheme is standard), logarithmic (decibel) scale that runs from –20 to +3, with usable resolution over about a 15 dB range. The zero mark is about two-thirds of the way up the scale. This 0 VU mark is what we mean when we say the meter “reads zero,” not where the pointer rests when the equipment is turned off.

Resolution is very good near the high end of the scale: generally 1 division per dB between 0 and +3. The scale resolution becomes progressively lower (more dBs per scale division) below zero VU.

These days, the mechanical meter’s replacement is usually a column of LEDs, with individual lights serving as the meter scale ticks. On a computer-based digital audio workstation, a meter is often simply drawn on the monitor screen. The Cool Look Factor is lower, but it’s a lot cheaper than using mechanical meters. However, some analog audio devices—notably “boutique-style” preamps and channel strips such as the PreSonus ADL 600, ADL 700, and RC 500—still sport traditional VU meters.

In this article, we’ll review the characteristics of the traditional analog VU meter and also address audio-level measurement and management in the digital domain, the concept of headroom, and how loudness and audio level are related—as well as how they’re not related.

What’s a VU?
The Volume Unit meter was originally designed to help broadcast engineers keep the overall program level consistent between speech and music. There’s a well-defined standard for the mechanical rise-and-fall response characteristics of the pointer (to which few of today’s VU-like meters actually comply).

A standard VU meter responds a little too fast to accurately represent musical loudness but fast enough to show movement between spoken syllables. Unscientific as it may seem, the dynamic response of the VU meter was tailored so that the pointer motion looks good when indicating speech level. It was easy to tell at a glance whether speech or music was going out over the air, whether it was about at the right level, and when something wasn’t working. Engineers learned that brief excursions up to the +3 dB top end of the scale rarely caused distortion in the analog equipment of the day, nor did they sound too loud.

How is Audio Level Measured?
Once sound is converted to electricity, we can represent the audio level by measuring the alternating electrical voltage. A symmetrical signal, such as a sine wave—a single-pitched note with no overtones or distortion—spends equal time on the positive and negative sides of 0 volts.

The numerical average of the positive and negative voltages over many cycles is zero—not a very useful measurement when we want to know how loud a sound that voltage represents.

The Classic VU Meter
Today, a VU (or pseudo-VU) meter is most often used for little more than to indicate an impending overload. We don’t watch a recorder’s VU meter to tell how loud our recording is, but rather, to ensure that we don’t exceed the available headroom—the 10 dB or so between 0 VU and the point where THD reaches 1%.

However, the VU meter is only a good headroom indicator if you’re working with program material that’s fairly consistent in level and doesn’t require a generous allowance for surprise peaks.

Take a close look at the VU meter scale. While the meter scale has a total range of 23 dB, fully half (the top half) of the scale represents only 5 dB.

This is good resolution for measuring steady tones when calibrating the recorder or setting levels within a system but pretty wasteful when working with a recorder that’s capable of handling a dynamic range between 65 dB (analog tape) and better than 90 dB (garden-variety digital [QQQ 16/44.1?]).

There’s no usable resolution below -10 VU. If we assume that we have at least 10 dB of headroom above 0 VU, a VU meter is really only informative within about a 13 dB range.

The classic VU meter scale

Why such a compressed scale? Practicality. Perceived loudness is a logarithmic function. It takes more than twice the signal voltage for something to sound twice as loud. A linear scale that represents loudness just wouldn’t look right. Remember that one of the design criteria for the VU meter was that the pointer response looked good for speech.

It’s no surprise that modern, highly compressed music will shoot the meter pointer well up scale, and it’ll stay right there until the fadeout. On uncompressed material with a wide dynamic range, there’s plenty of audible material down below the -20 mark on the VU meter, but an inexperienced engineer who trusts the meter rather than his ears can be misled into thinking that anything that barely moves the meter is too soft.

Today, many meters, both mechanical and LED or LCD ladder-style, have scales that look like a VU meter but don’t meet the VU standards. These are useful for establishing steady-state calibration levels when setting up a system, but they don’t accurately represent loudness or headroom.

LED ladder meters are often found on digital equipment but until you dig into the inner workings, you usually don’t know whether the meter indicates an analog voltage or the amplitude of a digital sample. In either case, the garden-variety meter doesn’t provide the same dynamic response of a real VU meter. It can show you average and sometimes peak level but it won’t tell you much about apparent loudness.

Zero VU is an arbitrary voltage level. It’s whatever is “normal” at the point in the circuit where it’s measuring. A meter that indicates input or output level is generally calibrated according to one of several industry “sort-of” standards. The most common is that 0 VU represents a voltage level of +4 dBu, about 1.23 volts RMS.

Most modern mixers are calibrated to this convention: When the output meter reads 0 VU, the device is putting out +4 dBu RMS. But this isn’t always the case. For many years, Mackie believed this was too confusing so they calibrated their meters so that 0 VU = 0 dBu. The “semi-pro” recording gear popular throughout the 1980s was generally calibrated for 0 VU = –10 dBV (10 dB below 1 volt, about 0.32 volts). Professional recorders are usually calibrated so that their meters read 0 for an input level of +4 dBu. In the broadcast world, “line level” is often +8 dBu, so that’s where their VU meters are calibrated.

Are you beginning to see what the “madness” is in the title of this article? Wait! There’s more!

Digital Metering
A digital meter—which is not a VU meter—has a scale with 0 dB all the way at the top. (Shown below is the Selected Channel level meter from a PreSonus StudioLive 32.4.2AI digital mixer.)

This doesn’t represent a specific analog voltage level but rather represents maximum digital level – a sample represented by a binary number with all the significant bits turned on.

Digital meters (regardless of what the scale says) measure dB relative to the maximum value (“full scale”) rather than a nominal value, as with an analog or standard VU meter. We call this dBFS, where the all-bits-on value is represented by 0 dBFS.

Unlike the VU meter, with a digital meter there is no headroom above 0. You can’t turn on more bits than the system’s word length. It’s up to you, the engineer, to decide where to set the analog-to-digital converter’s input gain to allow as much headroom as you’d like.

The object is to leave enough analog room so that only the loudest peaks approach a digital level of 0 dBFS. While today’s digital systems typically accommodate internal processing that yields a word length greater than what goes in or comes out, a 24-bit analog-to-digital converter is flat out when that 24th bit is turned on.

The digital LED meter scale

Perceived loudness is the same whether the source is analog or digital but we view headroom and operating range differently in the two worlds.

But How Loud is It?
Loudness is a function of both the recorded level and how far the listener has the volume turned up. In the past few years, commercial (and following in their footsteps, independently produced) recordings have been in a race to make each recording sound louder, when played at the same volume setting, than the previous one. A whole segment of our industry has sprung up as a result.

K-System Metering
In a presentation at the October 1999 Audio Engineering Society convention, mastering engineer Bob Katz of Digital Domain described the concept of monitoring at a calibrated sound-pressure level. He proposed a new type of meter scale that essentially displays headroom relative to the calibrated monitoring SPL rather than the digital level. His meter scale looks like a VU meter in that it’s calibrated both above and below 0 VU, but unlike a VU meter, it has a linear scale.

The K-System, then, is an integrated metering system tied to monitoring gain, and it is intended to standardize the levels at which sound is mixed and mastered.

Those Pesky Reference Levels
The real meter madness associated with digital level measurement is when you have devices with different reference levels (volts or dBu vs. 0 dBFS) in the same system.

To make this even more befuddling, you don’t always know a device’s reference level unless you measure it. A lot of gear is specified only in terms of a nominal analog reference level, without revealing the corresponding digital level.

Further, many digital I/O devices have no input or output level controls, so you can’t easily calibrate the reference level to match other gear in your system – you have to accept whatever calibration the manufacturer gives you. It’s bad enough when there’s a standard and not everyone follows it, but in this case there’s no standard for the analog level equivalent to 0 dBFS.

This leads to the common complaint of “my mixes aren’t hot enough” or conversely, “it plays much too loud.”

A dirty little secret is that budget-priced A/D converters (whether a stand-alone converter or integrated into another device) tend to be a bit on the less-sensitive side. It’s more likely that +4 dBu going in will give a digital level in the -20 dBFS ballpark than -12 dBFS. The reason is that with less gain on the front end, they’re digitizing a lower quiescent noise level, so the manufacturer can advertise a lower noise floor on the digital side. You need to hit the input pretty hard in order to get to 0 dBFS.

Sometimes you can crank up the output level of the source—for example a mixer—and sometimes, as with a microphone, you can’t. If there’s something you can adjust, you must take care that you don’t push the source feeding the A/D converter into clipping before the converter reaches maximum digital level. This can occur if you have a mixer with little headroom (a greater risk with older semi-pro mixers than with modern ones) or if you have a mismatch of nominal analog operating levels between the mixer and recorder. If you’re using a recorder with +4 dBu input sensitivity together with a mixer with a nominal operating level of -10 dBV, the only way you’ll be able to get the recorder’s meters to approach zero on peaks is to run the mixer’s output level meter well above zero, which will seriously compromise, and may exceed, the headroom in your mixer.

Loudness in the 21st Century
In the past few years, the audio industry, primarily in response to complaints that television commercials are too loud, has developed a new set of standards for loudness. This is primarily a broadcast thing but there’s now a variety of loudness meters that measure compliance with such loudness standards as ITU-R BS.1770. This is pretty complicated stuff and beyond the scope of this article but don’t be surprised if your DAW gets a plugin for it eventually.

Keep Your Eye on the Ball (Not on the Meter)
Once you’ve properly calibrated the gear in your studio, meters can tell you a lot about what’s happening, but don’t become a slave to them. Use your meters as tools, and you’ll spend more time recording and less time worrying about whether your levels are set properly. Don’t forget that you have a playback volume control. Leave getting the hot levels to the end of the process.

Mike Rivers has an electronic engineering degree and over 35 years as a design, system, and government engineer, while also operating a part-time recording studio and remote-recording truck. He is retired, but he continues to take on occasional projects, such as writing for PreSonus Audio Electronics. The original article, along with plenty of additional links to reference material, can be read here.

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Posted by House Editor on 05/04 at 05:48 AM
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Monday, April 25, 2016

The High Pass Filter, Your Best Friend

What Is It?

A high pass filter, or HPF, is exactly as it sounds.  It is a filter we can use on our soundboards that ONLY allows the higher frequencies pass. It is sometimes referred to as a Low Cut filter for a similar reason. It is also the most overlooked tool in the sound engineer’s arsenal.

Where It’s Found.

Some soundboards only have a High Pass Switch which is fixed at a certain frequency, often 80Hz or 100Hz.  This includes most Mackie, Behringer, Allen & Heath and similarly priced consoles.  Usually, only on higher priced consoles do you find the most amazingly useful type… the coveted golden ticket… the end-all-be-all… the “Variable High Pass Filter.”

The variable high pass filter is more useful because it allows you to change the frequency where the cut off begins, or more importantly where the lows no longer muddy up the bottom of our mix.  But rest assured, I have a little trick for you folks not yet blessed with a variable HPF.

Why We Need It.

Well, simply put, the more low frequencies allowed into a mix, the more muddy or unintelligible a mix usually is.

Let’s take a violin for example. For the most part the violin is made up of mostly mids and highs. So if we have 4 mics on our violin section, we are probably picking up a good deal of low frequency content from the timpani, bass guitar, kick drum, and so on. The problem is that the leakage from the other instruments, into our violin mics, is out of time with any of the close mics on the low frequency instruments.

Let’s take a short trip back to physics class. Sound is made up of waves, waves take time to move through air, and low frequency waves are longer than high frequency waves. Son if one mic hears two sound sources arriving at the mic at different time, we can say they waves are out of sync.  When waves are out of “sync” with each other we have cancellations and/or additions.

It is best to not have multiple mics picking up multiple instruments, especially if they have the same frequency content, but are different distances from the source.

Like I mentioned above. If the violin mics were picking up the bass guitar, it would be safe to say that the low frequency leakage of the bass into the violin mics is not “in time” with the actual bass input. Which would result in some of the bass guitar sound being compromised because of the out of time (or out of phase) leakage into the violin mics.

What Do I Do With It?

If you are lucky enough to have a Variable High Pass Filter the trick is to engage it and while listening to the violins play, sweep their HPFs up until you hear their lower notes change. At that point, back it off just a little bit, and know that the bass guitar leakage has been eliminated from the violin channels.

Did you follow that? By making the HPF higher, but not so high it altered the low notes of the violin, we have effectively eliminated any lower frequencies from leaking into those inputs and ultimately into our mix.

What If I Don’t Have A Variable HPF Or I Have A Fixed Frequency One?

So you more moderately priced Mackie, Behringer, and other folks are feeling a bit left out at this point. I wish we all had unlimited budgets to buy the consoles that had this feature, I realize they are expensive consoles, and sadly I know how that one goes.

Here is the trick for you guys.  Almost all soundboards have at least a HPF switch that can be engaged.  First trick… engage it on all channels except things like Kick, Bass, CD, Video, and anything else that has the potential to make really low notes.

Now since you do not have a variable HPF what else can you do… well you can use your low EQ to do a similar trick.

The low eq knob on most of the consoles at this price point are what is called a shelving filter. Which means everything below that frequency is attenuated similarly. So even though you still can not sweep it up to hear the low notes cut off, you can still clean up a little more low frequency leakage by turning this eq knob down.

What you do here is similar to the variable folks. Listen to your instrument, and have them play some of their lower notes. Turn down your low eq until you hear a substantial change in the sound of the low notes. Then turn it back up just a notch. You have now cleaned up any leakage from those mics similarly to how the folks with the variable HPFs were able to.

Why It Cleans Up The Sound.

It’s actually not just about the leakage. It is also about finding the holes in the mix for instruments. If an acoustic guitar is to be placed in a contemporary mix with electric guitar, bass, and keys, then the low frequencies of the acoustic are really not necessary. That is not to say you should make it sound like a swarm of bees, but the bass and electric guitars are certainly more capable of providing low frequencies. So if the acoustic is also taking up that range in the mix, it is very likely that section of the frequency spectrum will easily get clogged up.

How Does It Help The Amplifier And Loudspeakers?

Eliminating any unnecessary low frequency content also helps our amplifiers and speakers. Amps and speakers pretty much do what we tell them. So if we have a sloppy low and low mid section of our mix, they will reproduce it just as we mix it, but if we clean up our mix by eliminating conflicting and extraneous low and even low mid frequency content, we not only get a cleaner mix, we actually allow our speakers and amplifiers to run more efficiently since we are not asking them to reproduce content that is not necessary.

A 20-year veteran of working sound on the road, John Mills is the Education & Development manager for Morris Integration.

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Posted by House Editor on 04/25 at 12:45 PM
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Friday, April 15, 2016

Barry Weir Jr. Streamlines Workflow With Blue Sky’s Audio Management Controller

New AMC monitor controller supports up to 7.1 channel configuration and handles monitor measurement and calibration

Barry S. Weir Jr., a re-recording mixer and sound designer with Hollywood-based post production company Levels Audio, works on a wide variety of projects in any given week, requiring him to frequently switch between nearfield monitor configurations and calibration modes.

In late 2015, Weir installed Blue Sky‘s Audio Management Controller (AMC) to handle those monitor switching and calibration tasks.

“I decided to go with the AMC because it is one of the only monitor controllers on the market in the $2500 price range that provides so many features,” explains Weir. “Because the AMC is a monitor controller that supports up to 7.1 channel configuration and also handles speaker measurement and calibration, it was the best monitor controller for my workflow and studio setup.”

Weir has been with Levels Audio since August 2009, working on the Emmy Award-winning audio post team for HBO’s The 25th Anniversary Rock & Roll Hall of Fame Concert and taking a variety of roles on a wide range of projects, including feature films and television series and specials, promos, trailers and music shows. His credits include Emmy-nominated primetime TV series such as American Idol, America’s Got Talent, Cake Wars, So You Think You Can Dance, The Amazing Race and The Voice, on which he has worked variously as a re-recording mixer, sound designer, sound editor, sound effects editor, and voiceover recordist.

“Because I work on a wide variety of projects, I really need the flexibility to switch quickly between different monitoring setups, going from 5.1 to stereo, not to mention various room EQ calibration, from X-Curve to flat response, and bass management on or off,” Weir elaborates. “With the AMC Remote located directly in front of me within arm’s reach, it has really enabled me to streamline my workflow. I’m able to quickly and seamlessly switch between different monitor configurations and speaker calibration modes, bass management on/off, individual channel delays, levels and input source selections.”

Weir’s room setup is compact yet powerful, and places all of the important audio tools immediately under his hands. “My studio is only 100 square feet and I don’t have too much room for large equipment, so I require gear that has a small footprint but large firepower,” he says. Immediately in front of him is a Native Instruments Komplete Kontrol S49 keyboard, Avid Artist Mix and Control panels and RTW Primus meters, all centered around the Blue Sky AMC. His reference monitor setup comprises a 5.1-channel ADAM Audio system (three S3As for LCR plus two S3X surrounds) with a Genelec subwoofer, plus a pair of Behringer Behritone C50A mini speakers.

“I’m running all of my audio from my Pro Tools 10/11 rig out of an AVID OMNI interface via AES directly into the AMC. I’m using an ART TRS 48-point patchbay and my AMC is normalled to my ADAM Audio 5.1 monitor setup. Channels 1 to 3 are my ADAM LCRs, channel 4 is LFE, channels 5 and 6 are my surrounds and I use channels 7 and 8 for stereo playback through my mini speakers. Being able to use any speakers with the AMC is really convenient,” he comments.

The AMC ships with Blue Sky’s Speaker-Room Optimization (SRO) software, which combines precision room measurement and calibration tools with powerful corrective equalization capabilities. “I really like the SRO software,” Weir reports. “It was pretty easy to understand out of the box. Anyone with common professional studio knowledge can operate it. There are a lot of advanced features that I have been digging into more, which will enable more precise room calibration.”

Indeed, Weir has already immersed himself in the software and has had some suggestions for enhancements. “The customer support from Rich Walborn [Blue Sky’s chief technical officer] has been amazing,” he says. “Anything that I had a question about, or any bugs that I found, or any features that I suggested, he quickly addressed and implemented as new software and firmware updates. Not too many people or companies respond that quickly.”

Blue Sky

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Posted by House Editor on 04/15 at 10:14 AM
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Lightning Boy Audio Returns With New Op-2 Comp Tube Compressor Pedal

Features hand wired point-to-point construction, NOS paper in oil capacitors, and a pair of new-old-stock 12AU7 vacuum tubes.

Lightning Boy Audio (LBA) returns to the FX pedal business with the release of the new Op-2 Comp vacuum tube powered optical compressor pedal.

Designed from the ground up to deliver studio quality compression in a stomp box format. Op-2 Comp offers a wide, well balanced frequency response, which makes it suitable for both electric guitar and bass. Unlike its predecessor, Opti-Mu Prime, Op-2 Comp dishes out a very clean sound with low noise. It manages to do this while providing more gain and a wider range of compression than the former.

Op-2 Comp has the same simple feature set as its predecessor, supplying the user with a compression knob, volume knob, knee switch, and power switch. Inside the pedal is the same photo resistor found in the classic studio compressor, the Teletronix LA-2A. This device provides the auto-release characteristics of that much loved studio staple, while having a faster attack time from its LED light source. The heart of the pedal’s tone comes from its pair of new-old-stock 12AU7 vacuum tubes wired up with a healthy dose of Lightning Boy’s own secret sauce.

Op-2 Comp is 100% vacuum tube powered and runs off a standard 9V DC power supply. However, it is power hungry so make sure you have at least 1270mA available from your supply. The pedal is hand wired point-to-point and is made with NOS paper in oil capacitors (just like a quality vintage amp).  Made in the USA and ready to ship when you order. Op-2 Comp is available online direct from Lightning Boy Audio for $399.99 USD.

Lightning Boy Audio

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Posted by House Editor on 04/15 at 09:47 AM
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Thursday, April 14, 2016

Audient Console Selected For The Laundry Foley Studio

Analog ASP4816 console chosen to complete brand new studio facility in Essex.

“An incredibly quiet recording chain is achieved with an Audient ASP4816 console,” says The Laundry’s website, when detailing the equipment used in the brand new dedicated Foley studio, which opened in Essex at the beginning of the year.

Combining the compact, analog desk with a 3m x 1.7m projector screen, a NEC 4k projector, JBL 3678 screen channel speakers and two Crown DSI amps, they’re all set for high quality playback, while creating all manner of interesting sounds.

Owner Barnaby Smyth cites, “Patience, musicality and articulate direction,” as prerequisites for the job – he clearly has all of these in spades. “It’s hard work, but hugely fun,” he says. “If you want to good Foley you have to immerse yourself in the scene or character you are performing. Half-hearted Foley is all too easy to spot.

“The challenge is trying to achieve good recordings for a huge dynamic range of sounds: from face touches to car crashes. I have a lot of ‘go-to’ objects, but each project usually throws up new situations in which we have to create new sounds. This often involves sourcing new props – something that I’m now in a better position to do, having my own studio,” he explains.

“We tend to perform with the scene on screen, but we record a lot of wild tracks – not to picture – which are useful in the edit.” Audient wonders if he can describe some of the more peculiar ways of creating sounds. “When I’m rubbing my chest for lovemaking scenes, and kissing my hand and my arm,” he suggests.

The edit suite is located upstairs from the main studio, which where the Audient desk has fitted in a treat. “It’s been fantastic. Great, clean mic pres and the EQ is very versatile. It’s small but very flexible with a great number of returns from ProTools available.”

Surrounding the desk is an array of seemingly disparate items all of which help create sound effects. A glance at the company website is enough to confound visitors: row after row of shoes, a butler sink, an alarm clock, a chest of drawers and numerous floor coverings, ensuring The Laundry has the capability to produce “…a range of sounds from rickety old house, to plush stately home.”

With his impressive credit list, including The Night Manager; Tinker, Tailor, Soldier, Spy and Downton Abbey, Smyth and his colleague, Emmy nominated Foley engineer Keith Partridge are a formidable team in this new venture. Between them they have a broad professional network to draw on, especially for TV work. “Clients, editors and mixers I have worked with and have a good relationship with, bring return work. Films are slightly harder to come by, but I usually do a few a year.

“Now that I have my own place, the best thing about my work is that I can do what I like with the studio and develop and improve as I need to,” he adds, definitely happy in his work.  “We are being creative all day.” What’s not to love about that?

Audient

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Posted by House Editor on 04/14 at 01:15 PM
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Wednesday, April 13, 2016

SALZBRENNER media Introduces NIO xcel Series Dante-Based Audio Interfaces

New “plug & play” NIO xcel (NIO = Networked Input/Output) series of compact devices provide “Audio-over-IP” network for mobile or installation use.

At this year’s Prolight + Sound in Frankfurt, SALZBRENNER media presents its brand-new NIO xcel series of Dante based audio interfaces.

With their sturdy housing and professional connectors, these interfaces are perfect for any live scenario. The four models in this series are flexible and robust solutions for decentralized distribution of the required inputs and outputs exactly where they are needed.

While users in professional sound reinforcement, installation and studio settings rely increasingly on a network infrastructure, most existing solutions are both expensive and rather time-consuming to configure. SALZBRENNER counters this with a “plug & play” solution called NIO xcel (NIO = Networked Input/Output), a series of compact devices based on the Dante protocol for an inexpensive and reliable “Audio-over-IP” network.

With the NIO xcel series, SALZBRENNER media expands its product portfolio with four devices for both mobile and installed pro audio applications on stage. Two of the new devices (NIO xcel 1101 and 1102) provide inputs and outputs in the AES 3 format, another (NIO xcel 1201) offers 8 microphone inputs with 4 splits, phantom power and 4 line outputs, and the fourth (NIO xcel 1202) is equipped with 8 line inputs and 4 line outputs.

With these interfaces, pro audio users have all the relevant applications covered: feeding digital power amplifiers, inserting side-rack effects processors, providing microphone and line inputs for musicians and artists, and line outputs for monitoring purposes, and feeding analog power amps, press distribution systems, etc. The big advantage of these devices is that they can be positioned close to the sources and destinations and do away with need for separate DI boxes.

To ensure reliability and signal quality in rough and tumble environments, these format converters are fitted with professional connectors throughout. Dante and power supply redundancy are a given, and the interfaces can be stacked almost anywhere or mounted into 19-inch racks. Using conventional Cat 5 or Cat 6 Ethernet cables, they connect to an IP-based Dante audio network (100Mbps, 1Gbps, 10Gbps) that covers the entire stage, production studio, venue or installation.

While the underlying concept corresponds with that of proprietary network solutions from other manufacturers, thanks to adopting the Dante protocol this series is significantly more cost-effective, flexible, scalable, user friendly and future proof. Each device has its own web server for speedy configuration of many parameters and has 4-band EQs on all input and output channels and a freely configurable 16x16 mixer.

All NIO xcel interfaces that handle digital signals are equipped with sample rate converters that will accommodate all sample rates from 44.1 to 192kHz on every input. NIO xcel 1201’s microphone preamps provide 4-way splits, each with individual level adjustment, and a dynamic range of more than 152dB.

Other NIO xcel highlights include frame-accurate synchronization across several switches, negligible deterministic latency of the overall network, the flexible and scalable network topology with massive I/O counts, simple installation and intuitive operation, AVB support (TSN) and AES67 compatibility.

The new series will be on display at the SALZBRENNER media booth (E21/Hall 4.1) from 5th to 8th of April, 2016, where visitors of the Prolight+Sound exhibition will be able to get to know them hands-on.

SALZBRENNER media

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Posted by House Editor on 04/13 at 09:03 AM
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Monday, April 11, 2016

Solid State Logic Releases Sigma δelta Version 2 Software Package

Update includes DAW-based analog automation, new Sigma Remote Control App, and MIDI-Over-Ethernet MCU control.

Solid State Logic has released a Version 2 software package for its Sigma remote controlled analog summing mixer.

Now Sigma owners can take advantage of three-way round-trip control and DAW-based analog automation with SSL’s δelta-Control (δ-Ctrl) plug-in technology, the new Sigma Remote Control App, and MIDI-Over-Ethernet MCU control.

This upgrade includes the V2 Sigma δelta firmware and the new Remote Control App (Windows or Mac OSX), and is available via Sigma Owners’ My SSL registered user profile on the SSL website.

δelta-Control technology uses DAW plugin architecture to enable automation of the Sigma δelta analog signal path. This streamlines Sigma as a standalone mix environment and also allows existing DAW automation to be transformed into SSL SuperAnalogue automation, as well as enabling easy transfer of Sigma δelta sessions to Duality or AWS-equipped studios.

The main interface for δelta-Control is a native AAX/RTAS/VST/VST3 plug-in inserted into DAW mix or aux channels. The plugin sends and receives Sigma level and mute control data via the SSL Logictivity Network (Ethernet), which can be recorded, viewed, edited, and played back as normal plugin automation. The ‘Paste Special’ command can be used to copy existing DAW fader automation data into the δelta-Control plugin.

Sigma δelta volume and mute control data can be entered using the plug-in GUI, or by direct control of the Sigma hardware via either an SSL MCU control surface (such as Nucleus) or the new Sigma Remote Control App.

Audio on the DAW track passes through the plug-in slot unprocessed so the δelta-Control plug-in can be combined with other DAW plug-ins.

The δelta-Control plug-in is available to purchase exclusively from the SSL online store and is compatible with AAX, RTAS, VST, and VST3 plugin platforms.

The new Sigma Remote Control Application offers full control of all Sigma functions, plus storage and recall of saved settings from Mac or PC. The user interface provides a series of intuitive pages that make controlling Sigma’s powerful feature set straightforward. Three main screens provide control and setup of the Master Section, Channel Control, and Global Settings parameters. Every parameter can be saved for future recall making swapping between projects simple and efficient.

MCU control for Sigma facilitates direct control over the Sigma analog mix path and monitor switcher from SSL’s Nucleus DAW controller or any other SSL, MCU enabled control surface. Sigma’s MCU control requires no DAW host to function, only an active Remote Control App.

Combined with the Nucleus, Sigma becomes a fully integrated 32 into 4 automated line mixer, with full monitoring and talkback capabilities.

δelta-Control is not currently available for the AU plugin platform (Apple Logic).

For Pro Tools, Cubase, Nuendo, and Ableton Live, δelta-Control plugin stores its volume data using exactly the same dB law as the DAW fader volume data and will translate to Sigma channels with an accuracy of better than 0.2dB.

Solid State Logic

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Posted by House Editor on 04/11 at 06:57 AM
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