Wednesday, June 18, 2014
Allen & Heath Launches New Addition To Qu Digital Family
The Qu-32 is a 32 fader, 38 in / 28 out digital mixer joining the rackmount Qu-16 and compact Qu-24.
Allen & Heath will be unveiling Qu-32 at Infocomm. The Qu-32 is a 32 fader, 38 in / 28 out digital mixer joining the rackmount Qu-16 and compact Qu-24.
Qu-32 shares the Qu series key features, such as total recall of settings (including faders and digitally controlled preamps), Qu-Drive integrated multi-track recorder, dSNAKE for remote I/O and personal monitoring, multi-channel USB streaming, Qu-Pad control app, and the renowned iLive FX Library to deliver class-leading audio quality. It comes with a larger, 7” touchscreen and 33 motorised faders.
Key to the design was providing a dedicated fader per mic input channel while retaining a compact footprint, the Qu series distinctive styling, and extensive I/O, comprising 32 mic/line inputs, 3 stereo inputs, 24 mix outputs including 2 Stereo Matrix Mix Outputs and 4 Stereo Groups with full processing, patchable AES digital output with a further 2-channel ALT output, dedicated Talkback mic pre input, and 2-Track output.
Qu-32 is equipped with a high resolution full color 7-inch touchscreen featuring the easy to drive ‘Touch Channel’ access to channel processing, the FX racks and all the setup and system management controls. The SuperStrip provides control knobs for a selected channel’s key processing parameters, such as gain, HPF, parametric EQ, gate threshold, compressor threshold and pan.
Qu-Drive, the mixer’s integrated 18-channel USB recorder, can record and playback multi-track and stereo audio .wav files to a USB drive. The USB interface can also be used to store scene and library data for archiving and later recall.
Qu-32 doubles up as a perfect studio mixer thanks to its 32x32 audio interface for streaming to/from a Mac or PC, and MIDI strips dedicated to control of DAW track levels, selection, mutes and solos.
A&H’s proprietary dSNAKE low latency audio connection enables the mixer to connect over a single 120m Cat5 digital snake to remote audio racks, such as the AR84, AR2412, or upcoming AB168 stagebox. It is also compatible with the ME personal mixing system.
Motorized faders provide total recall of mix levels giving the user full benefit from the scene recall system and ensuring the fader is always in the right position even when swapping between the 2 layers, which allows instant access to all channels and masters or the Graphic EQs. To customise the fader layout to suit certain applications, a third user definable layer is also available. There are also 4 DCA groups for applications where multiple sources need a single level control.
The free QuPad iPad app gives instant wireless control of the mixer’s key parameters and settings, enabling the user to tweak the PA, adjust the monitors on stage, and even mix the show from the audience.
The 32 mic/line inputs feature crystal clear AnaLOGIQ recallable pad-less preamps, optimized for transparency and low harmonic distortion. In keeping with the excellent audio quality, the Qu-32 is equipped with a selection of the iLive pro touring series’ FX emulations, used by many engineers in place of top-end plug-ins and external FX units, including classic reverbs, gated reverbs, delays and modulators.
Finally, Qu’s software also caters for different Users, with levels of user access which can be customised and protected by password to easily cope with multiple users wanting different setups at different times.
“Packed full of features and retaining a dedicated fader per mic channel, the larger Qu-32 complements the rest of the Qu family. The comprehensive array of features, such as copy and paste, soft keys, RTA and PFL options make using and setting up the Qu console easy and flexible,” comments Allen & Heath MD, Glenn Rogers. “The Qu-16 and Qu-24 have been welcomed with great enthusiasm by customers from PA companies to churches, and we look forward to more success with the new addition.”
Qu-32 will start shipping in July at a SRP of £2599 ex VAT.
Allen & Heath
Monday, June 16, 2014
Satoshi Mark Noguchi Brings Manley Labs Gear Into The Mix
Utilizes Massive Passive across the orchestra bus and Variable MU on stereo mixes
After just under a decade in Los Angeles, the list of credits of mix engineer (and Seattle native) Satoshi Mark Noguchi suggests that he does more mixing than sleeping.
Currently working on a number of film scores for the likes of Rob Simonsen, Nathan Whitehead, and Joe Trapanese, Noguchi utlizes Manley Labs gear to inject life into the mix.
“I’m a big fan of the Massive Passive across the orchestra bus, and the Variable MU on stereo mixes,” says Noguchi. “A big part of my approach to mixing is to bring life, depth, and dimension into the music, and running mixes through the natural harmonics of tubes really seems to make them come alive. I love the soul in Manley gear.”
Noguchi’s first exposure to Manley gear came while working at Hans Zimmer’s Remote Control Productions in Santa Monica. ““I was a part of many great projects, and a lot of the experience I bring to my work comes from my time there,” Noguchi notes. During his time at Remote Control, Noguchi assisted noted composer Alan Meyerson, another long-time fan of Manley’s Massive Passive and Variable MU limiter/compressor.
Some of Satoshi Mark Noguchi’s more recent mix work can be heard on Bret McKenzie’s playful tunes in “Muppets Most Wanted.” His mix on the score for the Dave Green/Relativity Media sci-fi adventure “Earth To Echo” will be in theaters this summer.
Posted by Keith Clark on 06/16 at 08:28 PM
Atlas Sound Introduces AAPHD Series Mixer Amplifiers
All five models include a patent-pending automatic system test using the front panel "Push Here Diagnostic" button
Atlas Sound has announced the release of the new AAPHD Series of mixer amplifiers, the culmination of years of development to deliver a multi use mixer amplifier that incorporating features designed to save integrators time and money.
The AAPHD Series includes five models ranging from 30 watts to 400 watts that are designed for use in 70.7-volt loudspeaker systems, all incorporating a unique, patent-pending automatic system test using the front panel “Push Here Diagnostic” button.
This internal circuitry is designed to check the connected loudspeaker lines for wiring and impedance errors. The test can be activated once all loudspeakers are connected and the circuit automatically verifies that the attached speakers’ tap settings do not exceed the amplifier’s rated power, no speakers are mistakenly tapped at 8 ohms, and the loudspeaker wire is free from shorts. This allows for quick troubleshooting on site and after the sale.
Additionally, models (except AA30PHD) include a Remote Input Selection feature that allows the user to select an input and control the volume level from an optional remote wall plate, model WPD-RISRL, that can be placed up to 200-feet away from the amplifier. Each unit also includes a very low noise floor, enhancing sound quality.
“Mixer amplifiers are the most common amplifiers used by 70.7-volt integrators today,” says senior vice president of sales John Ivey. “This new series gives integrators something that no other model offers, an automatic test of the speaker lines. Not only is this a huge benefit to the installer onsite, who now has a quick way to test the system before moving on to the next item, but it also eliminates after install service calls. The owner can activate the PHD test and if no fault is indicated, the problem is in another part of the system. This eliminates troubleshooting time and effort; this is a feature that every integrator is going to love.”
Atlas Sound is at this week’s InfoComm 2014 show at booth C10508.
Iconic World Cup Venue In Brazil Outfitted With Allen & Heath GLD Digital Mixer
Equipped with Dante network card, mixer controls audio distribution around the entire stadium
Originally built for the World Cup in 1950, Brazil’s iconic Maracana stadium will host many of the major matches in the FIFA World Cup 2014, including the final in July, with audio managed by the stadium’s new GLD digital mixer from Allen & Heath.
Located in Rio de Janeiro, Maracana was closed for a 3-year major renovation, reopening last year with a GLD-80 mixer at the heart of a new sound reinforcement system. Fitted with a Dante network card, the mixer controls audio distribution around the entire stadium through fiber optic cables.
“The main reason we chose GLD was because of Allen & Heath’s fantastic sound quality, reliability and ease of use,” states Augusto Bergamim, sound manager and system programmer at the Maracana.
The stadium will host four group games and next round games, a quarter final game, and the final.
Allen & Heath
American Music And Sound
Friday, June 13, 2014
Plugging In: The Ever-Growing Processing Stable
The other day during load-in for a production, I used something I’d not touched in more than a year: outboard gear. We were adding a small analog console as a sub-mixer for the installed system when I decided to place a graphic EQ inline to better wrangle any feedback issues from the choir and floor microphones.
It dawned on me how my tools have radically changed the last 10 years. Large-frame consoles with heavy power supplies accompanied by tons of outboard processing gear have largely been supplanted by smaller digital consoles with internal processing augmented with plug-ins.
While the first thing I look at in a digital console is routing ability, next up on my list is signal processing. Digital boards offer impressive sets of onboard processing—graphic and parametric EQs, delays, multi-effects and so on—that replace racks of analog gear, and this capability can be even further enhanced via plug-ins.
Plug-ins, simply, are software. We use them on our computers with search engines and media players. Digital audio workstations (DAWs) and their users were the first to really implement plug-ins in pro audio.
These programs, designed primarily for recording and editing, provided a palette of processing options to bring to projects. Live audio practitioners soon took notice and have come to embrace plug-ins in their realm.
Audio plug-ins come in various types, including synthesis units that create sounds, sequencer or control units, and of course, the ones live audio folks primarily use, signal processing and FX software. Plug-in software is painstakingly crafted to emulate the operation and results of vintage outboard gear, including graphical user interfaces (GUIs) that look like the faceplate of the original, or it can be a new creation offering a new take on a particular effect or processor. Some offer a single function like a specific EQ, while others incorporate a “suite” of effects (and corresponding features) to choose from.
The new AE400 active EQ Native plug-in from McDSP.
Many providers also “bundle” groups of plug-ins together as a package. In addition, there are many free plug-ins available on the web. Some of these get very high reviews, while others… let’s just say the old adage “you get what you pay for” comes to mind. Waves Live and Avid are justifiably a very popular choice in live applications, offering hundreds of lauded plug-ins, there are many providers, such as McDSP, which recently released AE400 active EQ HD plug-in offering four overlapping bands and sidechain.
Plug-ins are available in a variety of formats, such as VST, AU, UAD, AAX, TDM, SawStudio/SAC, DirectX, TDM and MAS. The differences between some of these formats are considerable. For example, a certain format of instrument plug-in might support multiple MIDI input ports, whereas another format of instrument plug-in might not. So take care to check formats when selecting plug-ins for your specific gear.
Plug-ins run with Native or Server configurations. Native utilizes a computer’s CPU, so the number of plug-ins that can be run, as well as the overall system latency, is dependent upon the CPU and sound driver capabilities.
Server configurations (such as Soundcraft’s new Realtime Rack) use a dedicated unit to power the plug-ins, enabling low latency and high plug-in counts without taxing a computer. And some Server units offer networking capabilities so that multiple consoles can access the processing.
Above, a basic Waves Soundgrid configuration, and below, a screenshot of Waves MultiRack Native.
Waves MultiRack is a software host that lets front of house and monitor engineers run multiple, simultaneous instances of the Waves plug-ins. It provides four basic components: individual plug-ins, racks of plug-ins, snapshots that store plug-in and rack settings, and sessions that store the settings for all plug-ins, racks, and snapshots, as well as information about the setup design and signal flow.
MultiRack Native relies on a host computer to process the audio and uses an ASIO/Core Audio interface to connect to a console, while MultiRack SoundGrid uses an external DSP server to achieve low-latency performance and run multiple plug-ins in real time, working with select consoles.
MultiRack Native works with consoles from virtually all of the major manufacturers, and further, DiGiGrid MGB/MGO interfaces let users plug any MADI-enabled console into the SoundGrid platform. This fosters recording, processing and playing back up to 128 audio channels, using hundreds of Waves plug-ins, with latency of only 0.8 milliseconds.
Most Yamaha Commercial Audio digital consoles can host WSG Y-16 I/O cards (up to four) in mini-YGDAI slots to incorporate MultiRack and SoundGrid, as can Allen & Heath iLive and GLD consoles via M-Waves I/O cards.
Several Midas models, including the PRO Series also link to both via an MGB or MBO interface, while the SSL Live simply integrates via an MBO interface. And naturally, it also accommodates SSL plug-ins.
Soundcraft Vi Series consoles can also utilize MultiRack and SoundGrid via an MBO interface, in addition to the previously noted Realtime Rack hardware/software package that integrates Universal Audio UAD plug-ins, along with others from Neve, Studer, Lexicon, Manley, and more.
All DiGiCo SD Series consoles accommodate SoundGrid with a Waves I/O card as well as MultiRack Native with an MGB I/0 interface. And just a FireWire cable between computer and console is required for PreSonus StudioLive, Mackie Onyx and Behringer X32 Series models to interface directly with MultiRack Native.
Avid was one of the first console manufacturers (if not the first) to integrate plug-ins for live usage, a natural transition from the company’s tremendously popular Pro Tools DAW. All Avid consoles ship with a collection of plug-ins. VENUE consoles use the VENUE TDM format, while the new S3L console uses the AAX format that was introduced a few years and supports third-party plug-ins as well.
Some of the growing stable of Avid AAX plug-ins.
For example, McDSP has qualified all of its AAX plug-ins to operate on the S3L systems, while earlier this year Avid expanded the AAX live sound platform further to offer choices from providers such as Crane Song and Sonnox. Waves Audio also supports the AAX Native platform.
Engineers are increasingly utilizing plug-ins, and sound companies are working to support them. For example, Madrid, Spain-based Fluge has been steadily outfitting more of its consoles with MultiRack and SoundGrid servers.
Raúl Méndez (left) and Alvaro Ureña of Fluge with consoles being outfitted with Waves capability.
Recent applications include consoles for the musicals Hoy No Me Puedo Levantar and Grease and for artist tours by Melendi (DiGiCo D5), Pablo Alborán (Avid D-Show), Alejandro Sanz (DiGiCo SD7), Dani (Yamaha CL5), and Violetta (DiGiCo SD5).
“In the past few years, we’ve been investing in Waves tools with the objective of having Waves plug-ins available for our range of Avid Venue consoles,” notes Fluge technical director Raúl Méndez. “Now with SoundGrid technology and the ability to run Waves plug-ins in low latency with MultiRack on all consoles, we’ve decided to equip our stock of Yamaha and DiGiCo consoles with Waves tools as well.”
Front of house engineer Peter Keppler used SoundGrid with a DiGiCo SD10 console on David Byrne and St. Vincent’s 2012 and 2013 tours, and states that he plans to use it with a DiGiCo SD7 with Katy Perry this year.
“MultiRack SoundGrid with Waves plug-ins has been vital to my sound and workflow with the DiGiCo desks,” he says. “My go-to plug-ins are the Waves C6 Multiband Compressor, CLA-76 and TrueVerb, and of course I use several others as well. Using C6 on vocals works beautifully. I can use much more gentle compression on the channel, because any of the ‘peak-y’ frequencies are taken care of beforehand with C6, and subsequently retain more of the singer’s dynamics.
“Also,” he continues, “I do some basic EQ with the C6, and that leaves more of the console channel EQ available to me if I need it.”
Recently on tour with German rap artist Casper, front of house engineer Oliver Voges (Faith No More, Tangerine Dream, Sarah Brightman) used MultiRack with a SoundGrid Server One with a Yamaha CL5 console.
“Casper performs with a 5-piece rock band,” he says. “My main goal is to transfer the musical vision and energy from the studio recording to a live performance. In order to achieve this, live instruments and electronic sounds need to be glued together with a huge foundation of low end. This production is extremely challenging since it encompasses such diverse musical styles—hardcore, hip-hop, indie—creating an immense variety of sound.”
For a solid sub range, he uses a Waves MaxxBass plug-in, tuned to 43 Hz, with the input routed via an aux send, using it as a special FX sub for different sources, finalized by a L3-LL Multimaximizer in order to eliminate peaks and gain on the headroom. And for drums, Voges continues, “A combination of the very clear Yamaha CL5 input channel EQs to work on resonances, and different Waves EQ plug-ins for coloring. I love the V-EQ4 on kick drums and the SSL G-Equalizer on snare. For these channels I also use the Renaissance Axx compressor in order to get these pumping sounds I’m looking for.”
Engineer Michael Brennan mixing Primal Scream on an Avid S3L.
Front of house engineer Michael Brennan, a long-time Avid user, utilized an Avid S3L for a world tour last year by Scottish alt rockers Primal Scream. He primarily chose the S3L because its very compact footprint overcame some pretty serious size restrictions. In addition, the show file he had already created on his VENUE console was compatible with it, so he was confident in a smooth transition.
But what I found really interesting n was another plug-application. Brennan explains: “With the plug-ins and output patch function on the board, I’ve been able to use it as a system controller and crossover on several occasions. This has saved the show both times that the system controller broke, which was awesome.”
Times have most certainly changed in terms of audio tools, and it’s all for the better.
Senior contributing editor Craig Leerman is the owner of Tech Works, a production company based in Las Vegas.
Wesley United Methodist Church In Texas Reinforces With Soundcraft Vi3000 Digital Console
Also, new Si Performer 3 with an additional Dante card used for recording in a separate room
Wesley United Methodist Church in Beaumont, TX recently improved its audio capacity with a new Harman’s Soundcraft Vi3000 digital console and Soundcraft Si Performer 3 digital console. Installed by MSC Systems of Beaumont, the consoles are part of a sound reinforcement project aimed at providing better audio fidelity to the 100-year-old church, which provides contemporary services for up to 500 people.
The newly launched Vi3000 is outfitted with SpiderCore DSP plus extensive onboard options such as Dante and MADI interfaces. The expanded capabilities of the new 40-bit floating point DSP engine mixes FPGA and DSP technologies with a footprint small enough for it to be included within a control surface.
“The capabilities of the Vi3000 are amazing, as it has plenty of ins and outs,” notes Chase Daigle, lead designer/installer of MSC Systems. “The operators can now mix 12 stereo in-ear monitor mixes right on the console, and have 48 local inputs in addition to the 48x32 stage box run with fiber and 64x64 capability of Dante to the Si Performer.
“We can also configure presets for every channel, which means we can now recall settings for every single musician on different weeks, instead of resetting the whole board. Operation was so easy that the technical staff often involves the assistance of local volunteers, who despite lacking professional training were able to operate the Vi3000 and the Si Performer 3 after a few pointers.”
The Vi3000 handles FOH and monitors on different sections of the same board, while the Si Performer 3 with an additional Dante card is used for recording in a separate room. Feeds from the Vi3000 are also routed through the Si Performer 3, so the church services can be streamed directly online.
“The audio has been dramatically improved over our old system,” says Garret Kyler, worship and technical arts director of Wesley United Methodist Church. “The new Vi3000 console does not sound thin and digital, but very warm and analog-like. It’s also not like other consoles such as those made by Yamaha, where you go through layers of menus. The whole thing is incredibly responsive, because whatever you put in, you get back immediately.”
Church Sound: A Variety Of Ways To Handle Auxiliary Mixes
As more churches put their entire services online, the need for a quality broadcast audio mix becomes more critical. By “broadcast,” I’m referring to a mix that leaves the building, whether by actual broadcast or internet delivery. It could also be the same mix sent to the lobby, overflow rooms and other areas.
Why not use the main mix? While it’s technically possible to just take the L/R mix from the console and send it to video, the result usually isn’t ideal. This is true for several reasons.
The first – and biggest – issue is dynamic range. In a typical modern service, you’re likely to have 30-plus dB of dynamic range in the room. That sounds great – in the room. But on a laptop or in a cry room, people will be reaching for the volume control. A lot.
The second issue is the contribution of ambient sounds. There may not be a lot of drums in the main mix because they’re are already pretty loud in the room. (I hate seeing a video shot of the drummer when I don’t hear any drums.) The same may be true for guitars. Smaller rooms are more prone to this problem, but it’s an issue for everyone at some level.
Finally, the main L/R mix doesn’t have much, if any, ambience in it. Without some sense of what is going on in the room, the mix will feel dead. We’re not capturing sound in a studio; we’re in a live worship setting. Thus we need to hear people worshiping.
There are several ways to arrive at a good broadcast mix.
Use The House
This is the easiest approach, but for the reasons mentioned above, it’s also the least effective way to do it. Some house microphones could be fed to a matrix to add some ambience, but that means a lot of the aforementioned dynamic range. Subsequently running it through a compressor will likely make the music feel squashed. There are leveling products available and they work “OK,” but there really are better ways to go about it.
Some would argue this is the best way to get it done. A separate console is set up in another room with access to all the inputs from stage. A split – either analog or digital – provides all of the inputs the house console sees. An operator mixes these together with complete freedom with processing, mixing and effects.
Stems are an alternative when a full split and large broadcast console aren’t in the budget. The broadcast position might get a set of mono or stereo mixes: drums, guitars, keyboards, vocals, speaking mics, playback channels, etc. The broadcast engineer mixes and level-balances these stems while adding in some house mics. It’s a good way to go, though it does eat up groups or auxes on the house console.
The downside is that another console is still needed, as well as a room and an operator. For some churches, staffing another mix position is tough to do. But there is another approach.
This method is like a board mix, but it does differ. Basically, it involves taking the inputs and splitting them up into groups.
These groups don’t go to the main L/R bus, but rather, feed into the matrix mix of the console. Inside the matrix, they’re combined at the proper level so that when they come out, it feels right. We’ll talk more about that in a bit.
How the inputs are arranged into groups will depend on the band and your equipment. Currently, I’m using two mono and three stereo groups, and I also add several direct channels for walk-in music and audience mics. The beauty of this approach is that each element of the service is level-balanced to the correct perceived volume.
Different processing is also available at each stage of the mix. This provides more control and keeps the processing more transparent.
I had several goals in seeking to create a high-quality broadcast mix. First, I wanted it to sound good, even when I’m not mixing FOH. Second, the process has to be pretty seamless, and must work regardless of who is at the console.
Third, I wanted to create an accurate representation of what’s happening in the room – capturing the live energy is important to me. Finally, I wanted to do as little post production on the mix as possible, and doing the hard work up front helps in achieving this goal.
These groups form the basis of the author’s broadcast mix.
In The Grouping
What follows is my approach, which I offer here to get you thinking. This is descriptive, not prescriptive. The worship team splits into two groups, stereo band and stereo vocals. I typically add an extra 1-2 dB on vocals, which helps them stand out on video. I also do a little compression on each group.
A mono speaking mic group includes the pastor, plus any interview or announcement mics. Another stereo group handles playback of videos and the occasional Skype interview. A recent addition is what I call “Worship Leader Speaking.” When the leader talks during the worship set, it’s usually a lot quieter. This works in the room but feels (sounds) weird on video, so this mono group gives a little boost when they talk. Snapshots or macros are used to get those inputs in and out of the group.
Finally, a stereo pair of mics in the house picks up the audience and some ambience. The walk-in music playback channel is also routed straight to the matrix at the appropriate level. Because it’s post-fade, our opening transition is now cleaner.
On The Level
As noted earlier, it’s not uncommon to see a dynamic range of 30 or more dB (SPL) in a typical service. Speaking mics might run in the mid to high 60s, while music may be anywhere between the mid 80s to the low 100s (all dB SPL). The matrix mixing approach is designed to narrow that gap.
The initial temptation will be to balance out all of the various groups so they meter the same. So let’s say you want to hit the recorder at -12 dB FS (full scale). You’ll be tempted to set the levels for the music first, then dial up the speaking mic group until it hits -12 dB FS. But if you do that, the pastor will likely feel too loud.
That’s because in the real world, we don’t experience music and talking at the same volume. So don’t make them the same on video. Close is OK, but speaking will have to be less. I usually shoot for the speaking to be somewhere between 6 and 12 dB lower than the music, but that’s just a starting point. You have to listen to it and make adjustments accordingly. It has to feel right, not just meter right.
So that’s a little glimpse into my process. Next time I’ll share some of the “secret sauce” that has taken the mix from good to even better.
Mike Sessler serves as church sound editor of Live Sound International, and has worked with audio and production for more than 25 years. Currently he is the technical director of Coast Hills Community Church in Aliso Viejo, CA. Read more from Mike at www.churchtecharts.org.
Thursday, June 12, 2014
In The Studio: Mixing With Distortion—Where To EQ? (Video)
Distortion can be a very handy tool for doing a lot of fun (and useful) things in a mix—helping a track cut through the mix, giving something more grit, and so on.
In this video, Joe takes you through a recent project where he utilized distortion to attain a distinctive vocal treatment/effect, and how it was furthered through the effective use of EQ.
A big key is understanding where and how to apply EQ when it comes to distortion. Joe goes through his specific process and also provides some “before and after” takes to demonstrate (and clearly hear) the process.
Joe Gilder is a Nashville-based engineer, musician, and producer who also provides training and advice at the Home Studio Corner.Note that Joe also offers highly effective training courses, including Understanding Compression and Understanding EQ.
Live Recording: Splitting The Microphone Signals
Following is an excerpt from the just-released Second Edition of Recording Music on Location by noted LSI/PSW author Bruce Bartlett and Jenny Bartlett, published by Focal Press.
Let’s consider a different way to make a multitrack recording. Plug each microphone into a mic splitter, which sends the mic signal to two destinations: the PA mixer and recording mixer. The splitter has one XLR input and two or more XLR outputs per mic.
Some splitters have a third output which goes to a monitor mixer, and a fourth output might go to a broadcast mixer. (In Chapter 1 we described transformer-isolated splitters and Y-cable splitters.)
Splitting the mics is the most expensive method, but is the most professional. It gives you and the PA operator independent control of each microphone’s recording level and signal flow.
Pros: Ultimate sound quality. Independent control at each mixer. Consistent sound.
Cons: Complicated. Expensive if transformer splitters are used.
Equipment: Mic splitters, maybe mic preamps, mic cables, mic snake, recording mixer and multitrack recorder or audio interface and laptop, mixer-to-recorder cables, headphones or powered monitors.
(click to enlarge)
There are many advantages of splitting the mic signals. You use your own mic preamps, so you are not dependent on the quality of the PA console mic preamps.
Also, you are not hassling the operator about adjusting gain trims. Each mix engineer can work without interfering with the others. The FOH engineer can change trims, level, or EQ and it will have no effect on the signals going to the recording engineer.
Another plus: a splitter provides consistent, unprocessed recordings of the mic signals. This consistency makes it easy to edit between different performances.
(click to enlarge)
What’s more, splitters let you use mic preamps on stage if you wish. That way, the cable from each mic to its preamp is short, which reduces hum and radio-frequency interference.
As shown in the illustration, connect the outputs from all the splitter channels to the PA snake and to your recording snake. Connect the recording snake to your recording mixer mic inputs.
This mixer is used to set up your own monitor mix and to set the recording levels. Connect the recording mixer’s insert sends to a multitrack recording system of your choice.
Whether you record in the local rock club, jazz café, or in an orchestra hall, the Bartletts offer sage advice on each stage of the process of location recording. The Second Edition of Recording Music on Location has been thoroughly updated and includes new sections on iOS devices, USB thumb-drive recorders, and digital consoles with built-in recorders, along with updated specs on recording equipment, software, and hardware. The book provides an exceptional collection of information regarding all aspects of recording outside of the studio, and is available from Amazon and Barnes & Noble.
PreSonus Celebrates Opening Of New Headquarters
Company unveils facilities to research, innovate, and develop its next generation of audio products
Music-industry VIPs, business-community leaders, and PreSonus employees, distributors, and dealers converged on Baton Rouge at the end of April to celebrate the grand opening of the company’s new headquarters.
Louisiana governor Bobby Jindal opened the ceremonies, and PreSonus chairman of the board Kevin Couhig, CEO Jim Mack, co-founder/chief strategy officer Jim Odom, and co-founder/VP of manufacturing Brian Smith each offered their thoughts about the new building at 18011 Grand Bay Court, just off Highland Road.
“This new facility enables us to continue to grow PreSonus and at the same time improve the lives of our staff,” Mack stated. “Its architectural design enables us to work better as a team, be more productive and better create, market, and sell our products.”
“We now have the facilities to research, innovate, and develop the next generation of music and audio products,” Odom added.
Designed by local architects Ritter Maher, LLC, and developed by Moniotte Investments, the new 44,000-square-foot building cost approximately $8.8 million, including land and development.
Among the unique features is a specialized high-tech recording studio/R&D space that was custom-designed for PreSonus’ engineering and testing teams by the Walters-Storyk Design Group of Highland Park, NY. The 2,500-square-foot studio features a control room, a 500-square-foot live-sound room, two isolation rooms, a video-production suite, and five test labs, as well as a separate 1,500-square-foot live sound room.
The grand opening celebration used virtually every inch of the building, with live bands entertaining the crowd in the parking lot, the foyer, and the live sound room, as well as jamming in the studio. Featured performers included Tab Benoit (backed by Louisiana’s LeRoux, with Jim Odom on guitar), Edwin McCain, Briana Tyson, and Chris LeBlanc.
The PreSonus management team with Louisiana governor Bobby Jindal in front of new PreSonus headquarters in Baton Rouge.
PreSonus employees, distributors, and dealers converged on Baton Rouge at the end of April to celebrate the grand opening of the company
Digital Audio Labs Shipping Livemix Personal Monitor System With Dante
Available with both analog and Dante digital inputs
Digital Audio Labs is now shipping the Livemix personal monitor system with both analog and Audinate Dante digital inputs, and it will be on display at the upcoming InfoComm 2014 show in Las Vegas at booth C12116.
“One of the things we hear a lot about personal monitor mixers is that they can be complicated to use, especially for the volunteer or non-technical user,” says Ted Klein, president of Digital Audio Labs. “Ease of use is a big thing for us,” he continues “so we built Livemix to be simple to use, and this extends to setting up Livemix to work with your Dante network.”
Each personal mixer allows for two completely separate mixes, reducing stage clutter and overall per node cost. Custom channel names and easy navigation is possible via a touch screen display. Livemix also offers remote mixing, permitting any user in the system to hear and adjust the mix of another mixer, allowing more experienced musicians or sound techs to assist novice users.
Setting up Livemix with a Dante network is straightforward—the entire configuration can be done with the Livemix CS-DUO personal mixer, removing the need for a PC application.
Digital Audio Labs
Wednesday, June 11, 2014
Silence Can Be Golden: The Value Of Selective Muting
In high school, I held a stressful job: paint mixer at the local hardware store. The equipment looked like it was from Dr. Frankenstein’s lab – and I was Igor. One extra drop of dye could turn Orange Ruffy into Tangerine Dream, and there was no going back.
I learned two valuable lessons from that job: don’t rush a delicate process, and always check your work. On that second point, after the new color was mixed, a small wooden stick was dipped into the paint and then blown dry with a hair dryer – if it matched the color swash, it was good.
Years later, I’m applying the same concepts to audio mixing. It’s truly a building process. What starts as a bank of muted channels ends as 18 (or more) live audio channels. Color upon color is added to the base, and eventually, it’s regarded as finished.
But is the mix the desired color? We do ourselves a disservice by assuming it’s right – time to pull out the metaphorical stick and hair dryer in examining an audio mix for what it is and what it should be.
Not hearing can be good. Through muting an instrument or singer, the mind of a good sound tech can imagine what he wants to hear once the channel is un-muted. This gives the brain the opportunity to compare “what should be” against “what ya got.”
Mute mixing, for lack of a better phrase, enables volume problems to be fixed, EQ oddities to be corrected, and the overall mix to be improved. This process happens in two ways: channel-level muting and group-level muting. Let’s start with single-channel muting.
Volume balancing is an integral part of mixing, and by muting a channel it’s easy to evaluate the volume level. Start by listening to the whole music mix. Give it some time to sink in, and then mute a channel, such as rhythm guitar. Listen to the mix without that channel.
Next is the biggest step that has improved my mixes. Un-mute the channel, and it will be instantly noticeable if the volume of the channel is too loud, too soft, or just right. Make the appropriate adjustments and then move on to the next channel. This can be done for overall channel evaluation or fixing specific problem channels. For anyone new to mixing, definitely use this process for channel volume evaluation.
EQ correction via muting is similar to volume balance correction, with a twist. Imagine the electric guitar riff that starts the classic rock song Layla, where Eric Clapton’s riff has a very distinct sound. One could listen to five alternate lead guitar mixes and still know which one was from the original recording. We know what sounds right for a song.
Enter muting for EQ correction. Listen to the overall mix, and then mute the problem channel, such as the electric guitar lead. While listening to the mix without the lead guitar, imagine how it should sound if it was present in the mix. Un-mute the guitar and decide if it meets the expectation or not. If it doesn’t, make the necessary EQ tweaks. (And sorry, getting a different guitarist isn’t an option.)
Muting also helps to identify the natural room volume of an instrument. This can be applied to drums, brass instruments, and any instrument using a stage amp. In the cases of drums and percussion, using groups makes this easy. Any sound emanating from the stage with enough volume can affect the house mix. In some cases, one discovers the stage volume is greater than what is sent through the house loudspeakers.
Evaluating The Many
Muting a full group of channels is equally beneficial in assisting with volume and EQ work. Muting an individual channel might not be enough to help fix a problem.
In some cases, the problem exists across channels. By pulling out a group of channels, the source of the problem can be found. Group muting enables focusing on larger areas of the mix such as low end, guitars, and backing vocals.
My standard console configuration includes five mix groups: guitars, vocals, piano/keyboards, drums, and low end (kick drum and bass). Pulling out the low end and the keys groups, one hears the primary sounds of guitars and vocals that drive most songs.
Any time a guitar-centric mix isn’t coming together, drop all groups except the guitars and vocals. As long as those two sound good, the others can be reintroduced, one group at a time, to identify the problem area – usually it’s in the overall backing vocals or the overall drum mix.
Drum group muting can work in two different ways; it depends on the reason for the muting. Need to hear the difference in the mix with and without the drums? Use a single group mute. Need to fix a problem within the drum mix? Use the group concept but apply it to the channel level as follows.
Start by muting all of the drum channels, leaving the group level un-muted. Listen to the mix without the drums. Next, introduce the kick and consider how it sits in the mix. Continue through all of the drum kit pieces from the low-end kick up to the highest-pitched tom and then the snare. Optionally, add the snare after the kick and then work through the toms. Finally, add in the cymbals.
Another method for tweaking drums, rather than muting, is boosting the volume of the kit piece, adjusting the EQ, then lowering it back to the proper volume. The benefit of the mute method is allowing the brain to imagine what it wants to hear and then mixing to match that sound.
Sound techs working with the same band all of the time should have those sounds imprinted in their heads and can use either (or both) method(s). To them, I suggest giving the mute mix concept a try.
Muting groups also helps pinpoint a channel problem. For instance, a low-end frequency problem due to a bad keyboard EQ can be narrowed down to the keys by dropping out the low-end group containing the drums and bass. In this case, the low-end from the keyboard would stand out in the remaining mix.
The process of group-mute mixing enables one to identify a volume or EQ problem related to a group of channels. It also speeds up the investigation into a single channel-related problem by quickly eliminating many channels at once. Meanwhile, the process of channel-level mute mixing enables one to easily correct volume and EQ problems. It also leads to an overall mix improvement.
In contrast with mixing paint, audio mixing allows us to mix and re-mix as many colors as we want until we find the right combination. Dr. Frankenstein created a monster in his lab, but he did something far more interesting: he gave his creation life. I’ll let you draw the parallels.
Chris Huff is a long-time practitioner of church sound and writes at Behind The Mixer (www.behindthemixer.com), covering topics ranging from audio fundamentals to dealing with musicians – and everything in between.
Huntsville, AL Church Relies On Muzeek World For Dual Yamaha CL5 Consoles
New consoles joined by two Rio3224-D input/output boxes, one located on the stage and the other at front of house
First Seventh Day Adventist Church in Huntsville, AL worked with Muzeek World of San Juan Capistrano, CA on implementing dual Yamaha Commercial Audio CL5 digital audio consoles in its new 1,200-plus seat sanctuary.
The two Yamaha CL5 consoles were installed along with two Rio3224-D input/output boxes, one located on the stage and the other at front of house. “The purchase of the dual CL consoles was based upon the fact that the church previously owned a Yamaha console in their former sanctuary and had a wonderful experience using it,” states Muzeek World’s John Sardari.
“There are several new features in the CL5 which helped us decide this was exactly what we needed for the new sanctuary,” adds Julian Ray of First Seventh Day Adventist, who designed and installed the new system. “For starters, the expandability of the number of channels we can use. We enabled the board with 64 channels initially, and if necessary, we can expand in the future to its maximum.
“Second, the ability to have the sound engineer control the monitor mix on stage with the musicians, and while doing a sound check using an iPad, he/she can fine-tune and make changes on the monitors and have an accurate overview of the monitor mix,” Ray continues. “This eliminates the potential discrepancies between what the vocalists’ need and what they get in the monitors. We use floor monitors for the vocalists while the entire band is on in-ear system.”
Ray notes that another reason for the selection of the CL5s was based on the framework of the church’s previous Yamaha console. “We used an M7CL for about six years and had the team already trained on the same software, so the learning curve moving to CL5 was a breeze,” he says. “Other than the initial network programming and a few changes in the setup, we had the entire team up to date very quickly.”
The contemporary worship services at First Seventh Day average five to nine musicians at any given time. There is also a seven-member praise team and several choirs ranging between 30 and 100 members.
While the CL5 at FOH is in the center of the sanctuary on the balcony level, the second CL5 is in a room used to mix sound for recordings and video streaming over the Internet, also utilizing Yamaha CL Nuendo Live software.
Yamaha Commercial Audio
In The Studio: Lessons Learned From A Weekend Tracking Gig
The majority of my freelance audio work comes to me through the internet from all over the world. It’s maybe 1 in 10 projects that I do for musicians in my area. When I’m recording it’s usually overdubs.
Recently I had the pleasure of tracking a 3-piece band live in another studio. That’s not sarcasm! I really do love working with people in real life, setting up microphones, and making loud noises. I don’t get to do it nearly enough.
Thinking back on this session, a few important topics come to mind that will help you get through your next tracking session.
The headphone mix for the drummer and bassist were constantly changing throughout the two days of recording—more bass, less drums, more guitar, less guitar, more bass, less vocals, more bass. I kept tweaking their mix, filtering sub and compressing it in an attempt to get them more level, and the bassist kept playing harder and harder in an attempt to hear more clearly.
What was happening? I was listening to the same mix as them off my interface, but I did not check it on their headphones. As a result, their mix was totally distorted from driving the input of the headphone amp too hard. They messed up by not telling me it sounded like crap until 4 hours into the second day of recording.
Note to musicians: speak up, the headphone mix must sound good, tell the engineer if it doesn’t. I only found out the mix was jacked because I heard the bassist’s headphones buzzing when he was in the control room for an overdub. The point here is that you need to listen to the musicians’ headphones to adjust their mixes correctly.
I don’t always use contracts; in this session, I’m glad I did. It was great to have everything in writing and agreed on, simply because it gave me the power to say “No.”
If it’s not in the contract I don’t have to do it. The price and terms were set, all in plain English. There were a few important points in the contract:
—Price per day and when it is due
—Client pays for the studio and equipment rentals
—The service was defined as audio capture, including basic editing
—The service did not include advanced editing—vocal correction, drum editing, noise reduction
—No files or rough mixes released until paid in full
—Sessions last no more than 10 hours, including setup and breaks
—My recording credit specified
If the idea of “fix it in the mix” came up, I could shoot it down: “Nope, not in the contract, you have to do another take.” This is not me being lazy, this is quality control.
In case you’re wondering, I did say “Yes” much more often than “No” in this session.
Something Will Go Wrong
Aside from the headphone mix issue, my mobile DAW system caused a lot of frustration with intermittent glitches. I thoroughly tested the system at home before the session but issues still came up. Now I can’t recommend OSX Mavericks for audio.
We also had a phone go off in one of the best takes. The bass player had tape all over his fretting hand to keep from bleeding everywhere. Worse things can and do happen! You just have to stay positive, keep the morale up and work around the issue.
Know When To Move On
With that said, sometimes you just have to give up. One song the band was recording has a drastic change in feel between verse, pre-chorus and the chorus. It was a train wreck every time. If they did make it through, then the tempo would be way off. Most takes are incomplete.
We gave up on this song early on the second day and came back to it at the end. On the first take of the second attempt, a guitar string broke. Fixed that, then computer glitches, tried with a click track and without—it’s just not working.
I tried to keep it positive for them. We DID get through 19 other songs, and we started late on the first day and went home early. This one song needs more work and now was not the best time. Time to pack up and go home.
Anyone can mic a snare drum, that’s easy. Recording sessions are less about the equipment you’re using and more about working with people. That is the priority on these projects, keeping everyone happy and playing their best. You can’t fix that in the mix.
Jon Tidey is a Producer/Engineer who runs his own studio, EPIC Sounds, and enjoys writing about audio on his blog AudioGeekZine.com.
Monday, June 09, 2014
Artist Bill Fontana Enlists Meyer Sound For Chilling WWI Soundscape In UK
Remotely powered IntelligentDC system to accurately reproduce soundscape and immerse museum visitors
In his newest sonic sculpture, Vertical Echoes, noted sound artist Bill Fontana brings the horror of the WWI battlefield to IWM North, part of the Imperial War Museums, in Manchester, England.
As in several of his sound installations, Fontana has selected a remotely powered IntelligentDC system from Meyer Sound to accurately reproduce his soundscape and immerse museum visitors.
Vertical Echoes is housed inside a 55-meter-high tower known as the AirShard, located at IWM North. To evoke battlefield dynamics, Vertical Echoes juxtaposes the crescendos of battle against a backdrop of softer nature sounds.
Featuring recordings of a Sopwith Camel warplane and vintage field gun, the soundscape is reproduced by eight self-powered UPJunior-XP VariO loudspeakers with IntelligentDC technology. The loudspeakers are evenly spaced along the vertical axis of the structure.
“There’s extreme dynamic range in the work, from artillery fire and the buzzing biplane, to wind in the trees and twittering birds,” says Fontana. “The UPJunior-XPs were ideal for this piece because they offered both clarity and dynamic range.”
Providing the same sonic performance as the company’s self-powered AC systems, Meyer Sound loudspeakers with IntelligentDC technology are driven by a rack-mount IntelligentDC power supply, with one cable carrying both power and balanced audio to reduce installation times and costs.
“It’s just so much easier to work with in situations like this,” notes Fontana. “You don’t have to run AC or long speaker cables, but you still get true bi-amplified systems. The UPJunior-XPs were perfectly suited to this application.”
The Meyer Sound loudspeakers were provided and installed by Pro Audio Systems of Bradford, West Yorkshire, with project management headed by Lee Unsworth. A Meyer Sound dealer in the north of England, Pro Audio Systems is also a long-time supplier for IWM North.
“I was really impressed by the quality of the loudspeakers, and their capacity to play such clear and loud sound,” observes Zoe Dunbar, head of exhibitions at IWM North. “The vertical positioning of the speakers creates a breathing wave of sound that you experientially feel. Our aspiration was to mark the centennial by engaging visitors in a challenging way and giving contemporary relevance to the First World War. I believe we have achieved those goals with Vertical Echoes, and our visitors have reacted positively.”
Vertical Echoes is the first in a series of artistic responses to World War I commissioned by IWM North in honor of the war’s centennial. The installation runs through September 21. (Go here to listen to a sample track from Vertical Echoes.)
Recognized as one of the world’s foremost sound artists, Bill Fontana has created innovative and intriguing sound sculptures around the world since 1976. In 1999, Fontana’s requirement for very small, high-quality loudspeakers for his installation along the Lyon, France tramway provided the impetus for Meyer Sound to develop the MM-4, the first in the company’s popular line of miniature loudspeakers.
Find out more about IntelligentDC systems here, and check them out in live demos at InfoComm 2014 in Las Vegas at demo room N110.