Mixer

Wednesday, July 31, 2013

Meyer Sound D-Mitri Drives An Immersive Undersea Experience At Denmark’s Blue Planet

Digital audio platform mixes, times and processes more than 400 custom-recorded tracks

The new Blue Planet in Kastrup, Denmark, is Europe’s second largest aquarium, and it has won accolades not only for its variety of marine life, but also for its striking whirlpool-inspired architecture and imaginative application of audio technology.

To support an immersive undersea experience, more than 400 tracks of custom-recorded audio are mixed, timed, and processed through a Meyer Sound D-Mitri digital audio platform. The audio is then distributed to 57 self-powered Meyer Sound loudspeakers installed throughout the exhibit areas.

“In most aquariums the underwater experience remains apart and behind glass,” says Arne Kvorning of exhibit designers Kvorning Design and Kommunikation. “At The Blue Planet, the underwater experience comes out and surrounds the audience, due in large part to the sound and lighting design.”

Anders Jørgensenof Kastrup-based AV consulting and integration firm Stouenborg served as both sound designer and project manager for the installation. Working in close collaboration with Kvorning Design and Kommunikation, Jørgensen seamlessly integrated a variety of unique soundscapes with corresponding lighting and visual effects in each exhibit area. 

The D-Mitri system was at the heart of the concept, managing all audio functions and show control commands for Coolux Pandoras Box Server systems and a MA Lighting GrandMA2 light desk.

“With D-Mitri, I could work with an almost unlimited number of audio tracks, instead of being confined to stereo or 5.1 surround,” says Jørgensen. “I could load all of the tracks I created in the studio into D-Mitri and do the final mix in the aquarium, fine-tuning the intricate mixes in the rooms where they would be heard. This capability, when combined with Meyer Sound’s SpaceMap® multichannel panning software, gave me the perfect tool.”

The audio system includes the UPJ-1P VariO loudspeakers and USW-1P subwoofers. It also features various Meyer Sound low-voltage loudspeaker models that allow DC power and audio signal to be delivered over a single cable, reducing installation costs. They include the MM-4XP self-powered loudspeakers, UP-4XP 48 V loudspeakers, and MM-10 subwoofers.

“The sound design in the exhibit areas works in harmony with the subdued lighting inside the tank, and in the cathedral-like space in front of the window,” says Jesper Horsted, COO of The Blue Planet. “Stouenborg has created an intimate underwater experience—this in spite of a large room often filled with visitors.”

Located five miles outside Copenhagen, The Blue Planet (Den Blå Planet) houses 53 aquarium tanks that contain more than 20,000 fish and other aquatic life. The Danish architectural firm 3XN designed the building, with acoustical design provided by Gade & Mortensen Akustik A/S.

Meyer Sound

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Posted by Keith Clark on 07/31 at 05:03 PM
AVRecordingNewsAVDigitalLoudspeakerMixerNetworkingProcessorSystemPermalink

Moya Tour Benefits From Soundcraft ViSi iPad Monitor Mix

Compact, budget-friendly system topped by innovative app deployment

Whether by accident or design, the partnership of sound engineer Bryony October and tour manager Tim Boardman has made a speciality of starting a number of emerging acts on the road to success—supporting them with their expertise if not the luxury of full production.

On many occasions over the past six months, Soundcraft Si Performer 3 and Si Expression consoles have been chosen by the pair for lower budget tours around Europe.

And when October suffered a freak accident in Dortmund recently, leaving her at the A&E department of the local hospital, Boardman found himself piloting the Performer 3 for the first time and to his relief found it an intuitive “walk-up” experience like so many sound engineers before him. On many other occasions he has simply taken October’s show files and performed the mix for other artists on an Si Expression.

“It’s amazing what can be achieved, mixing FOH and monitors together and delivering a big sounding show from a desk that sits in a 19in rack,” he said.

Since finishing her long stint with the Noisettes, October has worked with artistes such as Billy Ocean, Laura Marling, Delilah, Marika Hackman, Bo Bruce and now Moya—a 4-piece band fronted by lead singer Emily Andrews, who have been supporting Rod Stewart on his current arena tour.

Earlier this year Moya played equally large venues in an acoustic setting supporting Mick Hucknall, but this time the band wanted to plug in—with Soundcraft’s ViSi Remote app for iPad enabling the band to control their monitor mixes from the FOH desk.

With their stripped down, splitter van-friendly production consisting of little other than a router, Cat-5 multicore and the compact Performer 3 (a convenient one-person lift), they persuaded Stewart’s Major Tom production team that rather than perform again in an acoustic context, they could carry a line system for the 30-minute support stint without compromising the main event.

With 29 of the 32 recallable inputs working the board to near capacity (along with 13 outputs) October is blessed that she that she can see all activity on the top Layer of the Si Performer with no page changes necessary. At the same time the desk’s FaderGlow and color backlit name displays on every channel provide her with a definitive color reference of what monitor mixes are being sent.

“It means I never make a mistake when moving from FOH to monitor mixes and I can clearly see the dynamics present,” October says.

She has been using the Si Performer’s output processing to send post-fade FX to the stage so the reference sound heard in the band’s in-ears matches the audience experience.

“I’m using ambient mics, reverb and panning to give them a feel that they are right there in the mix—providing an open sound so that they feel what the audience feels. I really like the ease in which I can change the panning on each mix using the same encoder as panning for the main mix. It means I can space things out in each band member’s mix depending on their position on stage.”

As to how she builds the soundscape, October uses vocal reverb, drum reverb, slapback delay and tap delay (on Moya’s voice). “Although she has an incredible voice I also want the dynamics to be present in everything else, producing a good full range sound.”

She admits to having had concerns about mixing five in-ears from FOH in arenas and even a stadium at Dublin’s RDS, until the iPad idea struck her. “I upgraded the desk at production rehearsals with the new firmware and latest version iPad operating system—and the band was all over it.”

Nevertheless, on the first tour date they were tight for time. “We didn’t know we had to provide our own complete line system until night day before the first show so I had to build my looms and wire up the patch during the first day’s sound check,” she says. “Thankfully Rod Stewart’s team allowed us to set up on stage before Rod’s sound check, which gave us a lot more time. He was happy to work around our gear which is very unusual.”

The drummer and keyboard player used small mixers to facilitate hardwired in ears and a separate click channel which was split from the playback system. “If the drummer loses click, there’s no show,” October notes, “so he has to have total control over that. We also gave him one of the two stage iPads as he is sat down and can access it most easily.”

This also provides reassurance for Moya herself (since she has no direct contact with her sound engineer) although she quipped, “The idea of trusting my life to my drummer was scary at first. But the setup has been quick and easy—and provides both confidence and consistency.

“The Si Performer has made this tour an absolute joy. It’s the first time I’ve used Soundcraft’s iPad app—and having the band control their own levels has completely freed me up to concentrate on the FOH sound.”

Soundcraft
Harman Professional

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Posted by Keith Clark on 07/31 at 04:43 PM
AVLive SoundNewsAVConcertConsolesMixerSoftwareSound ReinforcementPermalink

American Music And Sound Appoints JMS Marketing As New Allen & Heath Rep Firm

Representing A&H in Texas, Arkansas, Oklahoma and Louisiana

American Music And Sound has announced the appointment of JMS Marketing as the new sales rep firm for Allen & Heath in Texas, Arkansas, Oklahoma and Louisiana.

Based in Austin, TX, JMS Marketing is an aggressive service-oriented rep firm that covers the pro audio/video, sound/lighting contractor, and musical instrument markets.

JMS offers extensive territorial coverage with full-time reps on the road and staff in the main office, including two telemarketers. This mix of personnel provides the company with considerable service capabilities.

“We’re excited in the new opportunities that JMS brings to the table. The appointment of JMS brings in a new era of representation for Allen & Heath, which has experienced massive growth in the marketplace over the last few years,” states Michael Palmer, Allen & Heath USA vice president. “We feel that addition of the multiple personnel in the field will bring further development of an already strong and established brand in the territory.”

American Music And Sound
Allen & Heath
JMS Marketing

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Posted by Keith Clark on 07/31 at 03:17 PM
AVLive SoundRecordingNewsAVBusinessConsolesDigitalManufacturerMixerPermalink

In The Studio: An Interview With Noted Mix Engineer Andrew Scheps

Not still living in the analog past
This article is provided by Bobby Owsinski.

 
Here’s an excerpt of an interview that I did with top mix engineer Andrew Scheps that’s featured in the recently released 3rd edition of The Mixing Engineer’s Handbook.

—————————————-

Andrew Scheps has worked mega-hit albums for a who’s-who of superstar artists like The Red Hot Chili Peppers, Metallica, U2, Justin Timberlake, Jay Z, The Rolling Stones, Linkin Park, Jewel, Neil Diamond and Adele.

Even though he’s working out of his pretty outstanding home studio built around dual Neve 8068s, a massive wall of outboard gear, and dual Studer A800 24 track tape machines, amazingly Andrew is not one still living in the analog past, as the DAW is an integral part of his workflow.

Can you hear the final product in your head before you mix?

Andrew Scheps: If I know the song then I already have a pretty clear picture of what I’d like it to be. If not, I’ll usually get that the first time I listen through a track. It’s not so much for the sonics, but more in terms of size, like figuring out how big the chorus will be. Sometimes I’ll get really specific ideas about effects that I’ll try as well.

In terms of starting a mix, I think the main thing, especially if it’s a song I haven’t recorded, is that I go through instrument by instrument to see how it sounds, but what I’m really doing is learning every single part so that I when I come to build my balance, I know where everything is going to be.

Do you have a template for your effects before you start to mix?

Andrew: Kind of, although I don’t use a lot of effects. I use a lot of parallel compression so that’s more of what I have set up. In terms of what gets sent to those compressors, some of it is consistent and some of it changes with every mix, but they’re ready for me at the push of a button, which on an analog console is great because I just leave that part of the patchbay alone.

In terms of effects, sometimes I’ll have one kind of chorus-spreader kind of thing and one reverb and that’s it. I don’t tend to use many effects because a lot of the stuff I mix is straight up guitar rock and it’s more about the balance and making things explode.

Do have an approach to doing that?

Andrew: You’re never really as aware of your own process as you think you are. I’ll think that I really didn’t do much of anything and then I’ll look at a mix and find that I’m using 50 things on it.

Also, because I mix on a console there’s the whole process of laying out the outputs of Pro Tools to see where everything is going to come up on the console. There are things that always live in the same place, like channel 24 is always the vocal, so I’m usually figuring out how to lay out everything between the drums and the vocal. I do that while I’m finding out what everything is doing, so there’s a long discovery process where it doesn’t seem like I’m getting much done, but then everything happens really quickly after that.

Where do you build your mix from?

Andrew: It depends. I’d love to say that I always build it from the vocal, but usually what I’ll do is deal with the drums to get them to act like one fader’s worth of stuff instead of 20 or whatever it is. Once I’ve gone through that process that I just described, everything seems to come up at once.

I’ll have listened to vocal and the background vocals and know exactly where they are, but I’ll get the band to work without the vocals first, which I know a lot of people don’t think is a good idea.

I think it’s the same thing when you’re working on a particular instrument in solo. After 20 years, my brain sometimes unconsciously knows what an instrument will sound like soloed, so I’ll tend to get the tone on things separately, and then it’s all about the balance. I almost never have to go back and change things once I get the vocals in. My brain seems to know what that balance is going be when the vocals are inserted.

How much do you do in the box?

Andrew: I always think that I do nothing in the box, but I really do a lot of the technical things. The EQs on the Neve are very broad and very musical, they’re not good for anything surgical. If there’s a nasty frequency in the overheads or the snare is ringing too much, I take care of all of that in Pro Tools.

Usually I’ll have the background vocals coming out of one stereo output pair, so I’ll deal with them in the box. Sometimes I might split a couple of them out, but I don’t want 20 tracks of background vocals on the console; it’s just a waste. A lot of the crazier effects can come from plugins there as well.

There’s quite a bit that goes on in Pro Tools but it’s more about shaping things before they get out into the console. The console is much more of an organic balance thing while Pro Tools is more for making things sound the way I want them to sound. The console is more about putting it all back together and mixing it.

I actually mixed in the box for years in this same room. I had a [Digidesign] ProControl in here and that was great. In fact, there are some things that I mixed in the box that I listen to now and go, “Wow, that sounds really good.” I don’t have any philosophical differences with mixing one way or the other way. It’s more of once you have the console, as much of a drag as it is to document everything, it’s such a joy to mix on it.

When I’m mixing, it doesn’t matter whether it’s coming off tape or Pro Tools, it’s just faders and speakers and that’s it. I love that because sometimes mixing in the box makes you so precise that you then fix things that don’t really need fixing. I like the sloppiness of doing it on the console.

Do you find that you’re using your outboard gear less?

Andrew: No, not at all. When I document every mix, I wish that was the case because it’s a lot more to write down, but because a lot of it is parallel processing and stays patched in, it’s so much faster for me to hit a button on the console than it is for me to set the same thing up in Pro Tools. I may send the bass, the guitars and the background vocals to a stereo compressor, and in doing that in the box, it could change the balance on the board, so that doesn’t really work for me at all. It’s less of a sonic thing than a convenience thing.

Bobby Owsinski is an author, producer, music industry veteran and technical consultant who has written numerous books covering all aspects of audio recording. For more information be sure to check out his website and blog. Get the 3rd edition of The Mixing Engineer’s Handbook here.

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Posted by Keith Clark on 07/31 at 02:37 PM
RecordingFeatureBlogStudy HallConsolesDigital Audio WorkstationsEngineerMixerProcessorStudioPermalink

Church Sound: 10 Mixing Lies You Shouldn’t Tell Yourself

Removing ideas that are in the way of growing in the craft
This article is provided by Behind The Mixer.

 
There are times when you didn’t learn the right way to mix a vocal or an instrument or a band. That’s OK. You learned. You moved on. 

But then there are the lies you tell yourself over and over. You don’t learn. You can’t move on. Your work suffers. WAKE UP!*

*Any judgment on my part is because I’ve believed some myself. I don’t live in a glass house…at least not anymore.

#1: I can always boost the volume. The singer is moving the microphone from their lips to their stomach and then somewhere in between. Poor microphone usage techniques cannot be overcome by boosting the volume and accompanying massive fader riding. You must talk with the singer or go through the worship leader so they can be trained in proper microphone usage. 

And you know you can only boost a volume so much until their volume either feeds back or their mic is just too far away from their mouth. Spend the time, all of two minutes, and show them how to hold the microphone.

#2: I can cover up the problem with reverb. Reverb isn’t for masking audio problems…not entirely. Before you bring in any type of audio effect such as reverb or delay, fix the frequency problems with some EQ work. Otherwise, you are only “reverbing” those bad frequencies.

#3: This is the way I’ve always done it. This is the lie that because you learned to do something one way that it can’t be improved. It can go for microphone positioning, the use of effects, how you mix, etc. 

Believing this lie holds you back from improving your mix, improving your relationships with the crew and congregation, and ultimately holds you back from being the best you can be.

#4: I know what’s best. Similar to #3, you believe your knowledge and experience trumps everyone else. Some of my best mixing techniques came from hearing the question “do you mind if I show you a few tips on mixing this band” and giving a response of “yes.”

I’m not saying you have to accept what everyone says. We know everyone has their opinion about mixing but you do have to be willing to listen because you will learn something.

#5: Compressors can’t do that. Let’s ask this question, “do you know what your equipment is capable of doing?” Are you sure about that? I recall having a discussion with a couple of techs about a problem that had occurred when some people had “borrowed” the sanctuary for use for an event with a band. 

Apparently, they had a really odd audio problem where all sorts of bad audio was coming out of the system – weird feedback, odd sounds and all. Three of us were standing around talking about it, discussing the possible problems because we learned the gain structure was set properly, and two of us said it sounded like the compressor settings were messed up. (In fact, we learned they were). Another guy looked at us like we were idiots and said “a compressor can’t do that.”

Don’t be close minded about what your equipment can or can’t do, especially when something is going wrong.

#6: It sounds this bad because our equipment stinks. Ask any guitarist and I’ll bet they’ve heard a story where some old man or some rock star walks into a music store, picks up the cheapest “worst-sounding” guitar, and makes it sing like never before. And there is something to be learned of these stories…it’s not the equipment, it’s the person playing it.

I’ve guest-mixed for churches and a few times heard the words, “wow, it’s never sounded that good.” On an old Mackie, on an old Peavey, on an old…you get the idea. If you have bought into the lie that your sound can’t get any better because of the quality of your equipment then you have probably stopped trying to do your best. Yes, some of it can be the equipment but usually it’s you.

#7: Louder is always better. Go! Get out! Eh, just kidding. Louder is not always better because, taking a point from a recent conference, volume is like temperature. Everyone has their ideal temperature, and it’s not the same for everyone—and so it is with the house volume. 

If you’re saying “louder is better,” then at what point is it too loud? Oh, it’s whatever you think is too loud? What about the people in the congregation—do they get any say in this?

#8: I can mix how I want. “More bass!” “I love bass!” “More keyboards!” “My son plays keys and he should be heard.” Sounds a bit silly now, doesn’t it? Remember you’re mixing for a group of people so they can fully be involved in worshipping our Lord and Creator. Mix in the style they like. 

Don’t believe you can mix how you like and everyone benefits from that. If it so happens your mix styles meets up with their mix needs, then great. Otherwise, mix for them, not for yourself.

#9: Only boost frequencies that would benefit the mix and don’t cut anything. It’s not that boosting is bad, it’s that you’ll find achieving a good sound often comes by first cutting out the bad frequencies and cutting back on the energy of the frequencies that aren’t beneficial to your sound.

You don’t grow a great garden by ignoring the weeds and planting more flowers. You grow a great garden by getting rid of the weeds so the plants can grow on their own.

#10: Finish this by adding another audio lie of your own. What have you believed or what have you seen others believe?

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

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Posted by Keith Clark on 07/31 at 01:28 PM
Church SoundFeatureBlogStudy HallBusinessConsolesEngineerMixerTechnicianPermalink

Tuesday, July 30, 2013

Making It Personal: A Look At The Personal Mixing System Market

Getting up to speed on the various systems and what each has to offer

It’s hard to believe that just a few short years ago the entire personal mixing market belonged to just a few companies, replaced by a seemingly ever-expanding landscape of these systems.

The concept is simple: feed some number of inputs, typically16 to 40, to the system’s input module, connect the personal mixers to the distribution hub, and let the musicians put together their own monitor mix. Ideally, each musician gets what they want without the need for a monitor engineer.

Some would argue that a monitor engineer will deliver better results than giving musicians one more thing to do, and that may be true.

On the other hand, I’ve spoken with several bands now tour with personal mixers because they weren’t happy with the results from a monitor engineer. So we’ll let that argument alone for now.

The reality is personal mixers are here to stay, so it’s a good idea to come up to speed on the various systems and what each has to offer.

Rather than offering a comprehensive review of every model, this will be a look at newer and improved systems. And we’ll start with the company that started it all.

Aviom A360. Introduced this past January, the A360 is the long-awaited upgrade to the venerable PM-16ii. Boasting a new 36-channel mix engine, it now lets users select up to 36 sources from up to 64 on the network. Each mix channel can be mono or stereo, and has volume, reverb, tone and stereo placement controls.

New features such as a one touch ambience button, a dual profile channel that stores two configurations (useful for a background vocal that leads one song, for example), instant mix recalls, and USB setup demonstrate that this isn’t your grandfather’s Aviom.

It’s compatible with existing Aviom input modules, though to take advantage of the higher channel count, you’ll need to use the new one. The company also recently introduced a Dante module for I/O.

Roland Systems Group M-48. This one might be considered the first of the next generation personal mixers. Each unit is essentially a 40-channel mixer with 16 stereo groups. Through software, it’s possible to group the channels in any configuration for each mixer, with full level control for every input on every mixer.

Though originally designed to integrate with Roland’s V-Mixing system using proprietary REAC protocol, the S-MADI bridge allows users to grab the first 40 channels of a MADI stream for use with the M-48s.

Each mixer offers a 3-band mid-swept EQ for each group, overall bass and treble controls, a built-in ambient mic, reverb send level on each group, a 3.5-mm aux input, both 1/4-in and 1/8-in headphone jacks plus a stereo pair of TRS jacks for wireless IEMs or wedges. Power comes from the S-4000D distribution module, which can handle up to eight M-48s with a cascade port for adding additional units.

Allen & Heath ME-1. One of the newest kids on the block, A&H included a whole of capability with this system. Designed to work with iLive and GLD consoles, an optional input module accepts up to 40 channels from Dante, EtherSound or MADI. They even threw in an Aviom compatibility mode.

A bright OLED screen lets you see the custom channel names, one knob sets channel volume and pan, as well as a master EQ and limiter and a clever master trim function that lowers all inputs should the user start getting them all too high.

The surface has 16 rubber keys that can be configured to control a mono or stereo input, or a group of inputs. Rather than burning auxes on the board building stems, the ME-1 does it internally. Like many of the others, both 1/4-in and 1/8-in headphone jacks are provided, as well as a built-in ambient mic.

Movek myMix. Though it’s been out for a few years, Movek has added several features to myMix that make it even more versatile. New input modules accommodate up to 32 mixers on the network, and each mixer can select up to 16 of those.

A new network control module makes set up and configuration a lot faster using nothing more than a browser. When connected to the internet, you can even manage mixer configurations from anywhere in the world.

The mixer itself has a high-resolution backlit LCD monitor showing current status and inputs. A single large knob makes quick work of building mixes, setting up the per-channel EQ, reverb and panning when used in conjunction with four buttons on the side. Each mixer is also a multi-track recorder, writing .wav files to an SD Card. Input options include analog and ADAT.

dbx Professional PMC16. This system offers plenty of conventional functionality along with some pretty cool features that will appeal to anyone invested in the Harman Pro product line. The system runs on BLU link, which means it will interface with the BSS Soundweb London line of processors.

Channel linking, grouping and multi-select are all supported. Each mixer also comes with Lexicon reverb and dbx PeakStop limiting.

The obligatory 1/4-in and 1/8-in headphone outs are included, as well as a pair of TRS and XLR mono or stereo outputs. The TR1616 analog to BLU link input module features 16 dbx mic preamps, and 16 XLR outputs, providing an analog split plus the BLU link digital output. And, this is the only system that can run at either 48 or 96 kHz.

Elite Core PM-16. In some ways, this is a “new and improved” version of the Aviom PM-16ii. Instead of a single encoder that performs many tasks, Elite Core placed 16 volume and 16 channel knobs on the surface corresponding to the 16 input channels, and 16 signal present LEDs make it easy to verify everything is working.

A 3-band graphic EQ handles tone shaping on the output, joined by a smooth sounding 1-knob compressor and a built-in ambient mic. There are 1/4-in and 1/8-in jacks for ear buds, and a set of 1/4-in TRS jacks to drive a wireless transmitter or powered wedge.

The all-steel housing looks like it will survive more than one drop to the floor, and the headphone amp is very high output. Input options are analog and ADAT at present.

Pivitec e32. Based around Ethernet AVB protocols, this approach is unique. The mixer is a small box with no controls, just a 1/4-in headphone jack, a 1/8-in stereo line in, a pair of 1/4-in TRS line out jacks, and an RJ45 network connector.

Each mixer is controlled by an iPad or iPhone, all wirelessly. As the name suggests, the e32 can mix up to 32 channels at once. Multiple 16-channel analog input modules can be added to build larger networks.

The iOS interface is quite smart looking, presenting the user with 2 banks of 16 channels. A page select button access the other 16, and a custom layer lets the user put the most-used channels on the surface at all times. The iOS app supports both iPads and iPhones in both landscape and portrait orientation. Newly added are 16 mix snapshots that can be recalled sequentially or random.

Digital Audio Labs Livemix. While this company is no newcomer to digital audio, the Livemix is the newest entry in this field, with the system scheduled to ship this fall. A number of unique features set it apart, including the fact that each surface controls two mixes, and those mixes can be up to 24 channels.

Each channel has full EQ, compression and reverb controls, along with presets to make those functions easy for novices to use. A built-in screen makes it easy to remember which channel is which, as each channel can be named.

A “Me” knob can be configured to control the musician’s inputs for easy access, and a cool “Mirror Mix” mode lets any surface on the network control any mix. That would be especially useful for a FOH engineer or MD to help musicians who are struggling with their mix.

Each mixer can also record the stereo mix to a USB drive. Finally, the stereo mix is sent back to the input module where it’s available in the rack on XLR outputs to drive wireless transmitters—very cool.

Mike Sessler has worked with audio and production for more than 20 years. Currently he is the technical director of Coast Hills Community Church in Aliso Viejo, CA. In addition, he’s the author of the blog Church Tech Arts (www.churchtecharts.org) that is also featured on ProSoundWeb.

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Posted by Keith Clark on 07/30 at 06:14 PM
Live SoundFeatureBlogEthernetMixerMonitoringNetworkingStagePermalink

GC Pro Announces RAVEN MTX Production Console By Slate Pro Audio Now Shipping

46-inch touchscreen that provides direct touch access to every aspect of the production process

Guitar Center Professional (GC Pro) has announced that the RAVEN MTX multi-touch production console by Slate Pro Audio, originally shown at the 133rd AES Convention, is now shipping.

The RAVEN MTX is outfitted with a 46-inch touchscreen that provides direct touch access to every aspect of the production process, from recording, editing, plug-in control, MIDI composition, and final mixing.

The console’s RAVEN MIXER software is a powerful multi-touch mixer that can control all major DAWs, and the RAVEN TOOLBAR provides intuitive control familiar to slick mobile computing devices. The console facilitates quick navigation through any virtual production environment.

A digitally-controlled analog monitoring section delivers multiple inputs and outputs, plug-and-play integration with digital audio interfaces and multichannel artist cue mix systems, 7.1 surround capability, and on-board Laptop Reference loudspeakers.

“We are thrilled that the long-awaited RAVEN MTX console is now shipping, and we have begun fulfilling our initial orders to our customer base,” states Rick Plushner, GC Pro vice president. “In order to support and meet the demand for hands-on demonstrations, GC Pro will have RAVEN at select GC Pro facilities around the country, including Los Angles, New York, Atlanta, Dallas, Washington D.C., Nashville, Miami and Chicago.

Plushner continues, “For a limited period of time, the surround version of RAVEN MTX will include the Penteo 3 Touch Stereo-to-5.1 Surround Plug-in, the world’s first multi-touch plug-in for the Raven MTX which transforms any stereo recording, soundtrack, DJ mix or live television broadcast into perfect surround sound. It’s a very comprehensive package for the pro audio and post-production markets.” 

“We’re aware the RAVEN marks a significant moment in time and perhaps in audio history,” notes Alex Oana, RAVEN co-creator and product manager. “The sheer size of this realization has been driving us, the Slate Pro Audio development team, since the public’s first reactions to the RAVEN prototype.  We fully intend the product we ship to live up to the buzz and most importantly the expectations of our discerning customers, our peers and friends.”

GC Pro
Slate Pro Audio

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Posted by Keith Clark on 07/30 at 10:09 AM
Live SoundRecordingNewsProductConsolesDigitalDigital Audio WorkstationsMixerSoftwareStudioPermalink

Wednesday, July 24, 2013

Yamaha Training Sessions Head To New York City In August

Sessions include M7CL training for beginners and advanced users

Yamaha Commercial Audio Training Seminars (YCATS) will be holding training sessions in New York on August 13 and 14, 2013.

Sessions include M7CL for Beginners, M7CL for Advanced Users, and Yamaha Technology Partners.

M7CL for Beginners (August 13, 1-5 pm)
Attendees will become familiar with console signal flow, dynamic processors, equalizers, busing, and sound system hardware. In addition, they’ll have an opportunity to apply what they’ve learned using multitrack audio and the M7CL digital mixing console.

M7CL for Advanced Users (August 13, 8 am - Noon)
This course reveals the depth of the M7CL console and teaches how to utilize the console to the full extent of it’s capabilities. And, there is also the opportunity to apply knowledge using multitrack audio and the M7CL.

Yamaha Technology Partners (August 14, 8 am—Noon)
A discussion on the setup, operation, and optimization of Aviom personal monitors, Dugan auto mixers, Lake signal processors, and Waves plug-ins that incorporate into Yamaha digital consoles through the MY-Card slot.

There is no charge to attend Yamaha training sessions. For more information and to register, go here.

Yamaha Commercial Audio

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Posted by Keith Clark on 07/24 at 07:03 AM
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Tuesday, July 23, 2013

Allen & Heath Launches OneMix iPad App For GLD

GLD OneMix iPad app gives musicians wireless control of their personal monitor mix.

Allen & Heath has launched GLD OneMix, an iPad app that gives musicians wireless control of their personal monitor mix.

GLD OneMix locks control to a single Aux mix, providing instant access to a customized easy-to-use monitor mix environment.

Multiple iPads can be set up by an Admin user to give numerous musicians personalized monitor control without affecting each other or the FoH main mix. A musician’s own aux monitor mix is assigned and locked into the ‘My Mix’ layer, and a selection of input splits dedicated to the individual musician can be added.

Similarly, all other instrument send levels can be assigned to any of the three extra layers, allowing unique personal monitoring configuration. The range of accessible settings is defined by custom permissions for each layer.

When in User mode, the musician is presented with simple-to-use access and control of their aux master level and processing, instrument send levels and processing, if enabled. Depending on the application and the performer’s technical knowledge, the layout and level of access can be kept minimal or extended to a more complex musician’s monitor mix involving a high channel count across multiple layers.

Notably, the mix can be tweaked and listened to by both the user on stage and the sound engineer at the desk, enabling easy interaction between the two. Up to 16 iPads running OneMix can be connected to a GLD system.

“Joining GLD Remote, OneMix offers yet more control options for the GLD system. OneMix is a customizable solution for personal monitoring, drilling down the mix into very simple local control for performers without affecting the greater mix environment. OneMix also complements the ME distributed personal monitor system and ideal for wireless In-Ear users who can have a complete wireless system with this app,” comments Nicola Beretta.

Allen & Heath

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Posted by Julie Clark on 07/23 at 10:36 AM
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Tuesday, July 16, 2013

In The Studio: How EDM Is Changing Mixing And Mastering

There's a different kind of finesse involved in its creation
This article is provided by Bobby Owsinski.

 
Both mixing engineers and mastering engineers are tied at the hip, though many don’t realize it.

Yes, it’s true that many mastering engineers are dependent upon a mixer’s business to keep the doors open, but that’s been changing, since many times there’s a shoot-out between mastering engineers to see who gets the gig.

Usually, the one who can provide the loudest master wins (there’s that loudness war again).

But that’s not the real issue, nor where mixing and mastering engineers are mostly tied together. In fact, the concept of a separate specialized mixing engineer and a creative mastering engineer both began at nearly the same time during the late 70s, and continued to grow in prominence from that point until today.

Before then, engineers were somewhat interchangeable and came with the studio that you rented. Usually the same engineer that recorded the project would mix it, since the projects were generally short (as in a few weeks) to begin with.

As for mastering engineers, they were just part of the process of transferring the audio signal from tape to vinyl disc (and later CD). It wasn’t until legends like Bernie Grundman, Doug Sax and Bob Ludwig began to make mixes sound better, and louder, than the mixer could, that the mastering engineer came to be what he is today—the last part of the creative process.

But EDM is changing all of that. Today there’s less perceived need for someone to mix an EDM track. The writer/programmer gets the sound he wants right from the start of the track, and since the kick and bass are already out in front and have a lot of impact, most feel that there’s no reason to hire a specialized mixer for that particular bag of tricks.

The same goes for mastering engineers. Thanks to some great tools from a variety of plug-in companies like Waves, Slate Digital, Universal Audio and iZotope to name a few (the same tools that many mastering engineers use), EDM mixers can pretty much make their mixes as loud as needed, so it’s not surprising when they ask, “Why do I even need a mastering engineer?”

One of the things about EDM is that there’s a different kind of finesse involved in its creation from what a great many of the industry veterans are used to, where manipulation of the sound is encouraged and celebrated, and distortion is viewed as simply a byproduct of that manipulation.

That’s the antithesis of most mix and mastering engineers that don’t deal in EDM, where in their world distortion is something to be avoided. In fact, getting impact from the rhythm section without it is almost revered.

As my buddy (and mixing legend) Dave Pensado recently expressed to me, “We’ve (mixers) been too concerned with sonic quality, and it’s hurt mixers when it comes to EDM as a result.” It should be noted that Dave is one of the few mixers who does a fair amount of EDM, so he can speak with some authority on the subject.

Is this trend going to kill the market for mix and mastering engineers? Probably not. When it comes to music made by real instruments instead of samples and loops, it takes a great deal of expertise that only comes from experience with that type of music. I have a friend who creates fantastic electronic music, but is hopeless when it comes to either recording or mixing real instruments (especially the drums).

In many ways, it’s apples and oranges, but EDM is an ever-growing musical genre that now dominates the music business. As Aaron Ray, a principle in the management company The Collective said last week during a talk that I attended, “EDM has decimated rock. It’s now an entirely different business.”

The point of this post is to open up the eyes of those in our business who may be a little too tied to the past way of doing things, since there’s a whole genre of music that’s mostly ignoring you. In the end, we’re all in a service business and the client is still king.

It’s great to have principles, but if you hold them too tightly, you might find yourself not working as much as a result. If a client wants something that violates your aesthetic sense, in today’s world, you might consider suppressing your arty urges and give them what they want, because there’s a whole group of people right behind you that are more than willing to do just that.

Bobby Owsinski is an author, producer, music industry veteran and technical consultant who has written numerous books covering all aspects of audio recording. For more information be sure to check out his website and blog.

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Posted by Keith Clark on 07/16 at 02:33 PM
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It Takes Two: Double-Miking Approaches For Drums & Other Instruments

The technique can provide a whole new palette of tonal colors

If you’ve never experimented with double-miking a musical instrument, you’re in for a treat. Properly utilized, the technique provides a whole new palette of tonal colors, along with surprising ease of control. It’s especially useful when working with an unfamiliar console, one that has limited EQ capability, or when multiple operators are working together on the same control surface.

Further, with two or more microphones on key instruments, there is built-in redundancy. If one mic fails, falls off its stand, or gets whacked by a drum stick, mic number 2 is likely to still be in-service and able to keep the show going along relatively unscathed. For this reason, it’s a good idea to mix condensers with dynamics whenever possible, so that a failure of a phantom supply won’t cause both mics to go down.

Let’s start with kick drum, as it provides the foundational anchor for many modern musical styles. Certain shell materials, heads, and beater combinations can lack definition, sounding big but muddy and indistinct. Or at the other end of the scale, definition might be fine but the desired low-frequency “whoomph” is less than inspiring.

The usual practice of placing a single mic in front of the outer head or the sound hole (if there is one), or inside the shell on a pillow, may not provide the desired sonic quality – though each position will certainly produce different tonalities.

But even if you find a sweet spot with a single mic, you may not want that same tonality for every song, or you might have settled for a tonal compromise to begin with. Traditionally the problem is solved by applying EQ, maybe also a compressor and a gate, or endlessly changing out mic types to try to get closer to the mark. But there’s another way. It’s faster, easier, and comes with collateral benefits that solve other problems at the same time.

One or the other… or how about both? (click to enlarge)

On The Kit
With more than one mic to work with, it’s possible to create a wide range of tonal colors merely by blending the channel faders together proportionally to obtain the sound quality you’re after. In practice the technique is fast and simple. Depending on the sophistication of the console, the relative levels can either be recalled by using presets for different songs, or, on a modest analog mixer, the useful range of relative levels can simply be marked on tape alongside the faders. The only downside is the requirement for additional mics, multi-core channels, and console inputs.

I’ve found that the combination of a “half-cardioid” mic, such as a Shure Beta 91A placed inside the kick drum, coupled with a traditional dynamic cardioid such as an Shure SM7B or a Beta 52A located outside the sound hole or near the center of the front head, are a solid pairing. The 91A inside the shell captures a sharp, well-defined attack, while the SM7B outside the shell provides punch, and with a bit of EQ it can add a good measure of “thunder” when it’s needed.

This Ludwig snare is miked with a Sennheiser 442 on top and a Brüel & Kjær 4007 on the bottom. Its character can be changed in an instant by blending the ratio of the two mics, and also reversing their positions. (click to enlarge)

By blending the two faders together in different ratios, it’s possible to radically alter the timbre without ever touching the EQ, which can be held in reserve for creating additional layers of sonic potentialities.

Other mics can accomplish much the same thing while adding their own particular “flavor.” Good candidates are the AKG D12 or Electro-Voice RE20 used outside the shell, paired with an AKG C547 or Audio-Technica U851R inside the shell. The real value here is making use of the differences in physical placement and the differing types of the mics, rather than adhering to specific models. 

It’s no rarity to see a snare miked from both the top and bottom heads. This is perhaps the most common usage of dual-miking, and again gives the mix engineer a lot to work with. Want a more snappy sound to cut through screaming guitars? Increase the level of the bottom mic. Need to mellow it out some? Take the bottom mic down or out altogether. Try an AKG C451 on the bottom head and an SM7B on the top. Then reverse their positions and see what happens. It can be an ear opener. 

The same concept can be applied to toms, especially if they’re fitted with bottom heads. Depending on how the toms are tuned, there can be significant differences in what each mic picks up, thus creating an opportunity for making the toms sound larger than life on one end of the scale, or providing only mild accents on the other end. All without using EQ.

In situations where time allows, such as preparing for a lengthy tour, installing mics inside the shells will provide tremendous isolation from drum to drum, as well an extremely sharp attack that you can’t get from exterior miking.

However, the mids and lows tend to be very thin, so additional support is likely to be needed from exterior mics. On a big kit this might eat up a lot of channels, but if they’re available, the level of control you’ll experience is astounding.

You can even delay the exterior mics by a few milliseconds relative to the interior mics, producing the effect of a bigger and longer “body” that follows the impact of the initial stick contact. This is highly recommended for complex, demanding music such as fusion and progressive rock.

Spreading The Concept
Other percussion instruments can also benefit from a double-miking approach. Try miking both the top head and the bottom flare of a djembe. Listen to what happens. The depth of LF content from the bottom mic will put to shame many large-diameter kick drums. The modest djembe now becomes a powerhouse!

The same approach can also be applied to congas whenever a “power factor” is desired. (If recording, be sure to allocate two tracks for maximum flexibility when mixing.)

If a system is configured in stereo, there’s nothing like a pair of mics positioned over the tray of percussion “toys.” When a shaker is being moved around the stereo pair, the wide expanse and motion of sound in the loudspeakers can be breathtaking.

A conga miked with a Sennheiser e835 on top and a B&K 4007 on the bottom for a deeper, more defined LF response. (click to enlarge)

If taking this approach, it’s important that the percussionist has stereo stage monitors in order to provide some idea of the effect the movements are having on the spatial localization in the main system.

When a traditional lead instrument is present – one that plays an important role in the music such as a sax, trumpet, clarinet, or flute – miking in two places can capture the essence of the instrument’s voice in a way that no amount of EQ will achieve. A lot of tone emerges from traditional instruments, and it’s not all coming from the bell or the mouthpiece.

On a large baritone sax, for example, the lower part of the instrument often provides a resonant character that’s an important component of the sound the player is hearing and working off of. Fortunately with today’s miniature clip-on mics and readily available wireless transmitters, it’s not as hard as it once was to capture the very best that a given instrument has to offer without cramping the stage movements of the performer. 

Next time I’ll continue the discussion by focusing on bass, guitar, other stringed instruments, and more.

Ken DeLoria is senior technical editor for ProSoundWeb and Live Sound International and has had a diverse career in pro audio over more than 30 years, including being the founder and owner of Apogee Sound.

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Posted by Keith Clark on 07/16 at 02:06 PM
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Fishman TriplePlay Software Update Adds New Features & Enhancements

Now 64-bit compatible on both Mac and Windows platform, and increased functionality

Fishman has announced a comprehensive new software update for the TriplePlay wireless guitar controller.

The new TriplePlay v1.1 software contains more than 50 new improvements, features and updates.

With software update v1.1, TriplePlay is now 64-bit compatible on both Mac and Windows platforms and provides improved support with Windows 8.

A new Basic Enhanced Mode allows iPad GarageBand users to easily activate Pitch Bend. Factory patches now can use Komplete sounds directly instead of requiring Elements.

The new Import/Export feature allows the transfer of patches from one TriplePlay installation computer to another. Improved hardware support allows the entire patch list to be loaded into the controller for use in the Hardware Mode. The TriplePlay software also now lets users know when new software releases are available.

Other features include improved Encoder/Receiver communications to ensure greater connection stability; new plug-in support to improve scan time and accuracy; new Windows support to improve overall performance, and new support to enhance TriplePlay’s robustness in popular DAWs such as Cubase and Ableton Live 9.

Two new TriplePlay mixer audio options allow the Guitar Channel Level to be adjusted independently of Synth Channels and also allow the final Output Volume to be controlled by assigning CC80 to an external MIDI pedal.

Fishman

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Posted by Keith Clark on 07/16 at 12:58 PM
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Monday, July 15, 2013

Church Sound: Seven Steps For Cleaning Up Your Music Mix

You like the mix but think it could be better. Here's how
This article is provided by Behind The Mixer.

 
Saturday, I was conducting an audio training session and I was given the opportunity to work on their mix with the worship band. The mix was pretty good so instead of re-building their full mix, I focused on cleaning up the mix.

There are seven areas to consider when cleaning up a mix. Mind you, like I said, this assumes you have a pretty good mix to start. 

For the sake of this article, let’s say you’re in the middle of your sound check, you like the mix, but you think it could be better.

This is a great time to focus on cleaning up the mix, and here are seven steps to do it.

1. Check your volume balancing. Go through each channel and use this process; mute the channel, listen to your mix, then un-mute the channel. If the instrument or vocal seems to jump way out in your mix (too far out), then you need to pull the volume back a little. If the muting didn’t make a difference, then you didn’t have it loud enough.

2. Did you cut before boosting? This is an easy mistake to make, especially on an analog mixer. What is that old commercial for BASF?  Something along the lines of, “We don’t make baseball helmets. We make them better.” It’s better to have the best from the beginning. 

Regarding the situation I was in, cleaning up the mix, I thought something with a singer’s vocal didn’t sound right. It was close, but not where I thought it should be. They had a boost in the vocalist’s mid-range. Maybe in the 6 kHz range, I don’t remember. Using the sweeping mid, I moved the sweep frequency way down around the 800 Hz mark and did about a 4 dB cut. This really warmed up their vocals and gave their voice a great tone.

The first part of creating a mix should be cutting out the offending frequencies. Once you do that, then you can consider boosting when it’s necessary.  Remember, boost wide, cut narrow when you have control over the frequency range with a Q control.

3. Review your gating. Gating is often used on drum kit microphones to minimize audio bleed – when one kit piece is played but a different kit microphone picks up the sound. Where else could you use gating?

Consider the vocals a place for gating. A vocal microphone near a drum kit could easily pick up the drums. While that would happen when the vocalist is singing, what about when a different vocalist sings lead for a song, like in the case of a pianist who plays and sings. Why let those other sounds into the microphone? Gate the mic.

4. Hit your high-pass filter. You don’t need low end coming through a lot of your channels, so stop it. I’ll enable the HPF on my vocals and guitars with one exception. If I don’t have a bass guitar on the stage, then I’ll allow an electric guitar to give me some of that low end. In some cases, you can control the point of your high-pass filter. I’ve used an HPF in the 200 Hz mark on vocals to clean up my bottom end.

Tip: when altering any setting like boosting, cutting, gating, compression, or a high-pass filter, go to the extreme so you hear a clear difference in the sound. Once you know how “extreme” it can sound, then back off the setting until it’s to your liking. Don’t turn a knob or press a button just because you think you should. Let your ears make that call.

5. Check your subs. The signal to your low end subs might be one that you can control. Therefore, you have this additional means of altering the mix sound. Don’t be afraid to pull them back or push them louder if that’s what’s needed for the mix. While you’re building your basic mix, you should have your subs at an average level. Once you have set your overall mix, then you should consider the sub volume.

6. Consider microphone polarity. When two microphones pick up the same sound, the combining of those sound waves may or may not cause problems. This is where you get into sounds being out-of-phase. 

For example, if the two incoming sound waves are in phase with each other, this means the sound waves, when compared side-by-side, look identical. As the distance from the sound source to one microphone changes, so does the point in which the sound waves enter the microphone. 

When this happens, the sound waves start to get out-of-phase (compared side-by-side, the wave peaks are at different spots. In the extreme case of being 180-degrees out of phase, the sound waves look like a mirror image of each other. Combine these sound waves and you lose a lot of the audio signal because it’s like math; +10 + -10 = 0 (flat-line)

A simple way for ensuring you are getting the best sound from an instrument, where phase could be an issue, is by switching the polarity button on the channel. If you get more bass response, then you have found a better setting. By switching polarity, you are inverting the sound wave.

7. Re-visit your effects. Once you’ve cleaned up your mix using the above methods, you should re-visit your effects settings.  If your board enables it, turn off all the effects and listen to the difference.

Otherwise, go channel by channel. What worked before might not work now. It might not be needed to the same degree or it might not be needed at all. You might even need MORE of an effect because your cleaner mix enables you to use more of the effect to reach your mixing goal.

The Take Away
The process of cleaning up your mix is best explained with the words of author Antoine de Saint-Exupery,

“…he has achieved perfection not when there is nothing left to add, but when there is nothing left to take away.“

You aren’t going for perfection in the traditional sense, but the idea applies just the same.

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

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Posted by Keith Clark on 07/15 at 03:04 PM
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Friday, July 12, 2013

DiGiCo SD10 At The Heart Of Mega Monitor Mixes On Frampton’s “Guitar Circus”

Engineer Matt Fitzgerald kicks off tour with new audio footprint at monitor world to handle ramped-up requirements

Peter Frampton reclaims his guitar throne on the “Frampton’s Guitar Circus” tour, which kicked off at the end of May in Nashville, TN at the Ryman Auditorium.

The guitarist-turned-frontman/singer is sharing the stage with a dizzying array of axemen on the summer outing including Steve Cropper, Dean DeLeo (Stone Temple Pilots), Don Felder (formerly of The Eagles), Mike McCready (Pearl Jam), Vince Gill, David Hidalgo (Los Lobos), Davy Knowles, Roger McGuinn (founder/lead guitarist of the Byrds), Richard Thompson, Vernon Reid (Living Colour), Vinnie Moore (UFO) and Rick Nielsen (Cheap Trick), with B.B. King, Steve Lukather, Kenny Wayne Shepherd and Sonny Landreth trading off opening the shows.

With Clair Global providing the consoles and full stage package, Frampton engineer Matt Fitzgerald—who has worked with Frampton for the past four years, and with Ringo Starr and Blue Man Group prior—kicked off the tour with a new audio footprint at monitor world to handle all of the ramped-up monitoring requirements.

Fitzgerald chose a DiGiCo SD10 for Frampton’s 5-piece band and the guest guitarists. “This was my first experience using one of the new SD consoles in-depth,” he explains. “I’d had lots of mixing experience on a D5 working with opening acts. But for this summer tour, I opted in favor of a smaller footprint and more flexible solution by trading up two linked digital desks that I was using prior for an SD10.

“Not only was it difficult mixing inputs between the two desks, but it was sluggish also as I had to treat each one as an individual mixing surface.

“Obviously making a big switch was a bit nerve-wracking,” Fitzgerald continues. “Peter’s the kind of guy that wants to be able to walk into rehearsal and just go. I did a lot of research and spent time with DiGiCo’s Ryan Shelton in Nashville and worked with the offline software to build my big session files.

“We had a decent amount of rehearsal time, about two weeks, because the band was learning the other guitar players’ material, so that was of huge benefit to me. I was able to get really comfortable on the desk, storing snapshots, making scene-to-scene files, etc. Literally after the first day, I was shocked and awed at how easy it was to get around on the desk and how small the learning curve was. I felt like a burden was lifted off my shoulders and I was able to just concentrate on mixing, not on the new gear.

“I was really blown away by the desk and really liked that everything was laid out so naturally. The buttons were right where I needed them, not to mention that the features are fantastic and you can have the channel strip wherever you want and create the desk to make it the way you want it to be. It’s really comfortable and I really enjoy it.”

With all of the band on Westone ES2 in-ears, Fitzgerald was looking at a lot of variables with the guest guitarists—some would be on ears, some on wedges; some had big stereo rigs and others had simple combo amps, acoustic guitars, etc.

“I wanted to have a lot more flexibility and room to grow,” he says. “With the SD10, I’m able to build a bigger show file as well as have extra wedge and ear mixes built-in. And it’s been awesome as far as changing the layout of the desk from show to show. For instance, we just played with Steve Cropper, who uses wedges, and I was able to move my wedge mixes to my top layer and restructure my file, and it was so cool and easy.”

Fitzgerald makes use of all of the onboard effects from the desk: reverbs for drums, acoustic guitars and keys, slap delays for vocals, plus a bit of multiband compression for vocals and DiGiTuBes on the bass for a little bit of drive.

The only external effects he’s employing is an Eventide H3000 harmonizer for the acoustic guitars.

One of Fitzgerald’s challenges is creating a controlled environment on a day-to-day basis, blending the audience mics and stage sounds in everyone’s in-ears: “The tonality, spaciousness and imaging are really great and big-sounding on this desk. There’s a real clarity to the sound. Any inputs like Peter’s LCR Marshall and stereo Leslies for is his guitar rig, the Hammond organs and keyboards that are hard-panned left and right, sound so clear and natural. The imaging is excellent and sounds like you don’t have ears in at all.

“I’m able to fine tune the EQ of the audience mics to each room, so the audience sounds the same even though you’re in an arena, theater, or event hall… When someone in the audience yells in between songs I want Peter to be able to look into the audience and know where and who is yelling and make eye contact with them. With the SD10’s imaging and clarity I can pan audience mics exactly where I want. It’s as if I’m mixing the audience just as much as I mix the band.

“Peter’s mix is a very full range, dynamic mix of the entire band with his inputs just top,” he adds. “He wants to hear everybody and what they’re playing. If someone solos I ride my programed control groups to give them a nudge in his ears so he hears all important parts of the songs.”

With the tour well underway, the feedback on the new console among band and front man has been unanimously positive. “This band is so great to work with because they respect my input and ideas as we move forward with this new console. I try to open up the spectrum in their ears and create a ceiling for them instead of monaural mixes.

“I want it to sound as natural as possible and at the same time have full control of what’s going on onstage so I’ve created a big open stereo spectrum with Peter in middle and the guitar players panned on each side to create that image of what’s really going on onstage. If you take your ears out, that’s what it’ll sound like. And the fact that I’m able to recreate all of this in the in-ear monitors and have it sound so natural is really cool.

“This DiGiCo desk is amazingly ‘analog’ sounding,” adds Frampton. “It’s very warm and, with its full band width, has incredible smooth high-end response. Matt can run monitors and multi-track record every show with ease. This is due to the foresight of design. It’s like not having ‘ears’ in but instead, listening to a really great pair of studio monitors. But… I’m playing live on stage.”

DiGiCo

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Posted by Keith Clark on 07/12 at 10:02 AM
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Thursday, July 11, 2013

Church Sound: Understanding Signal Flow & Console Operation

How well do you really know your console? If a problem crops up are you prepared?

Knowing the audio path through a mixing console is absolutely critical to sound tech/engineer success.

Using this information, an engineer can quickly troubleshoot the likely causes of common problems, and can even narrow down the possibilities of unexpected major problems.

It can also prevent mistakes because you know what the audio is doing at each stage of the console, and it instills confidence as you sit behind the console, fulling knowing the the ins and outs (sorry for the pun) of the equipment.

Finally, it provides a foundation of understanding which makes it easy to move from room to room or console to console and not be thrown for a loop.

For instance, you might think “the second red knob on my old console was always set to 12:00, does that mean the second blue knob on this console should be set the same?”

However, after carefully studying a console’s signal path, you’ll know exactly what that knob is and where it is in the audio signal chain (even consulting the owner’s manual if necessary.)

You want to be an excellent all-around driver of vehicles, not a specialist who only knows and drives a Chevy Malibu 2-door with the small V6.

Generalities
In general, the controls that you tend to “set and forget” are at the top of the console, meaning you have to actually reach for them. The controls that need more adjustments along the way are closer to your hands.

The channel strip tends to lay out generally “in order” as it applies to the audio signal flow – Gain, then EQ, then the Fader, for example. But this is a very broad overview. There is much more detail to be examined..

Figure 1: Yamaha DM2000.

So how do you learn the signal flow of your particular console? You break out the manual!

It will contain what is typically called a “block diagram”. Now, block diagrams like Figure 1 for a Yamaha DM2000 can be headache-inducing nightmares.

So I recommend that you take the time to create your own simplified signal flow. Just follow the lines on the block diagram to determine the signal path.

It’s also recommended that you make it in linear, vertical orientation so that it helps you visualize the flow better. You can use any drawing or paint program to make one.

Examples
I’ve created a few signal flows for study. These can be extremely valuable learning tools.

Here is the signal flow of a Mackie 1604 VLZ:

Figure 2. Click to enlarge.

 
 
 
 
 
 
 
 
 
 
 

And here is a larger format Yamaha IM8•40:

Figure 3. Click to enlarge.

 
 
 
 
 
 
 
 
 
 
 

Finally, here is an APB Dynasonics Spectra-C/56.

Figure 4. Click to enlarge.

 
 
 
 
 
 
 
 
 
 
 

Basic Definitions
Now that’s we’ve taken a look at some different signal flow diagrams, let’s review exactly what the different components you’re likely to see on block diagram are doing.

Gain: A level adjustment designed to optimize each signal coming into the console.

Pad: If you turn the gain all the way to the left and the signal is still too hot, then you should engage the pad, which will reduce the incoming signal by a preset amount (usually 20 dB or so).

HPF: A high-pass filter is a circuit which sharply decreases low frequencies, reducing mike handling noise, stage rumble, and plosives (p-pops).

Polarity: A simple switch which flips the polarity of the input. (Polarity is sometimes incorrectly called “phase”). Useful for eliminating phase-cancellation when using multiple mics on the same source (both the top and bottom of a snare drum, for example).

Insert Loop: A patch point for connecting outboard gear, such as a compressor or effects unit.

Direct Out: An individual channel output after the gain stage, but before EQ or fader involvement. Most often used for feeding multitrack recorders.

Aux Mix: A separate mix of each channel which has its own output, which can be used to feed stage monitors, a recording mix, sends to a reverb unit, or other uses.

Pre/Post: An indication of where the Aux mix splits off from the main signal. If it’s labeled as “Pre” or “PreFade” mix, then its level is completely independent of the channel’s fader. If it’s labeled as a “Post” or “PostFade” mix, then the aux’s level will also be affected by the channel fader as it is adjusted.

PFL: The Pre Fade Listen works as a “solo” button for the engineer’s headphones. You can isolate an individual channel, and hear changes you make with the EQ. Because it is pre-fade, it does not matter where the fader is at the time.

Group/Subgroup: A tool used to help the audio tech during a service or performance. Rather than have to independently mix 32, 40, or even up to 56 channels on a console, you can assign, for example, all of the drums to one fader called a “Subgroup.” The Subgroup does not affect any aux sends, it only affects the main mix. So I can raise or lower the level of all eight drum mics on one fader.

VCAs & VCA Groups: A VCA stands for Voltage Controlled Amplifier and is a common way to “automate” certain things on a mixing console. You can assign multiple channels to a VCA (just like a Group), but the difference is no audio is passed through a VCA.

Instead, the VCA acts like a remote control to channels which are assigned to it. Where it gets really interesting is that channels that are assigned to a VCA Group do not have to share a common audio path at all.  This means you can have the entire band on one VCA fader, even if they all are routed to different mixes and Subgroups!

Something to keep in mind with VCAs that you don’t have to worry about with Groups: a VCA provides the exact same function as adjusting a channel’s fader (including any changes to it’s Post Aux mixes). This is different from a Subgroup, as a sub would only affect the house mix.

Bus: This is an electrical term rather than an audio term. Technically, an aux mix, a Subgroup, a master mix, a mono output, a matrix output, etc. are all buses. The only way this term becomes important to an audio tech is in the possibility that you get some “bus distortion,”  which may not show up on the meters.

Matrix Mix: A completely different kind of output available only on the larger consoles. It’s sole purpose is to create an alternate mix to be used for recording, for routing a different mix to a different room, or for any other specialized purpose. You will not see a Matrix split on the above audio signal flows.

Why? Because they are not made up of individual channels. A Matrix mix is created solely from mixing the Main Outputs and Subgroup Outputs. So a Matrix Out is created downstream from any individual channel functions.

Jeremy Carter is a veteran of the pro audio industry with extensive experience designing and operating church audio, video, and lighting systems.

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Posted by Keith Clark on 07/11 at 12:35 PM
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