Friday, February 27, 2015
Sight & Sound Theatres Adds Studer Vista X Digital Console To Enhance Productions
Console's ease of use relative to its enhanced capabilities help in further immersive theatrical experience
Sight & Sound Theatres, a leading Biblically-based theater company, recently upgraded to a Harman’sStuder Vista X digital mixing console at its primary location, a 2,000-seat theatre in Lancaster County, PA.
Sight & Sound presents an ongoing series of immersive stage productions depicting Christian stories that often break through the “fourth wall” and directly engage the audience, hosting more than 15 million guests since its founding in 1976. Productions typically begin at the Lancaster County venue before moving on to another location in Branson, MO.
“We provide our audiences with a very immersive experience that goes beyond a typical theatrical production,” says Gary Parke, audio operations supervisor for Sight & Sound Theatres. “In fact, both theaters have additional stages to the side of the main stage, so that a lot of our shows feature a 180-degree stage setup.”
The addition of the Studer Vista X digital console helps add to the experience, with its ease of use relative to its enhanced capabilities making it an attractive option for Parke.
“Flexibility and input/output count were big factors in choosing the Vista X,” he notes. “The ability to build our I/O configuration to our exact preferences was a very attractive feature.”
The theatre is currently using 64 wireless microphones and playing back the audio via Steinberg Nuendo audio software. All audio is fed into the Vista X via Audinate’s Dante networking technology.
“In addition to the I/O flexibility, the Vista X will make it so easy to be creative in developing a realistic audio atmosphere for the audience, so we can create ambient sound effects that really make guests feel like they’re part of the story,” Parke said. “The Vista X gives us so much more flexibility than we have had before.”
Berlin’s TrueBusyness Recording/Mastering Studio Expands With SSL AWS 948 Console
Co-owner and chief engineer Sascha "Busy" Bühren particularly likes the AWS's integration of Pro Tools and other applications
Sascha “Busy” Bühren, co-owner and chief engineer of TrueBusyness in Berlin, one of Germany’s premier recording and mastering studios, is getting down to “busyness” with a new Solid State Logic (SSL) AWS 948 console.
The 48-input AWS is the centerpiece of the facility’s new recording/mix studio, dubbed “New York” in homage to the birthplace of Bühren’s favorite music. Bühren, who owns TrueBusyness with his wife, Laura, is a longtime DJ, his nickname inspired by his days as a “battle DJ” in events such as Disco Mix Club’s World DJ Championship.
He established a production company more than 20 years ago, soon after his first major label release, and has served as producer and mix engineer for dozens of German hip-hop artists. He has also built a reputation as a mastering engineer, putting the finishing touch to numerous releases in a wide variety of genres and artists, such as SEED, Unheilig, Max Herre, Gentleman and Rea Garvey.
Last year, Sascha and Laura expanded TrueBusyness by an additional 190 square meters into a neighboring studio and built a comprehensive new recording/mix studio. “We knew the rooms, their sound and the vibe,” Sascha says. “It was the perfect match to blend with our mastering house. After that, choosing a console was easy. From the time I started as an audio professional, I wanted to own an SSL console; there could be no compromise. The AWS is the perfect pairing of a great-sounding console in a well-treated control room.”
Bühren particularly likes the AWS’s integration of Pro Tools and other applications. “Now, I can work in-the-box outside,” he says, “which gives any digital production a 100-percent analog feel.” Sonically, the AWS surpasses his previous methodology of combining a digital audio workstation with a summing mixer. “Summing on a real console with 48 channels totally blew me away,” he says. “You cannot compare it to the old method-I just don’t want to go back, period.”
The total control allowed by the AWS, he adds, was essential. “Fast decisions in critical situations can be a lifesaver, and with the size of the console, you can reach all knobs and faders quickly, without leaving the sweet spot.” The studio also offers an SSL X-Rack stocked with E and K Series EQ and dynamics.
“Basically,” Bühren concludes, “we wanted a high-class facility for anyone that wants the SSL sound, surrounded by a friendly atmosphere where creativity comes before business. The sound of SSL tells a story like no other console in the world and our clients agree. The first reaction is always wow. Sometimes they just sit and listen, as though they had never heard music before.”
Massive Attack Completes World Tour With Soundcraft Vi4 Digital Console & Realtime Rack
Engineer Paul Hatt utilized a Harman’sSoundcraft Vi4 console to mix monitors for the recently concluded international tour by legendary musical duo Massive Attack.
Hatt also deployed a 32-channel Soundcraft Compact Stagebox for the input extension beyond the fully loaded Vi stage rack, plus a 64-channel Soundcraft Stagebox for additional inputs.
“I think this tour has been the most arduous outing I’ve put the Vi through to date,” he says. “The Vi4 has been thrown about in local trucks, charter flights and endured both extreme heat and torrential rain.
“In southern Italy we rallied around building impromptu gazebos and placed fans around the desk as the direct sunlight and temperature became too intense, and then in Mexico we had to evacuate the stage after five songs due to an electrical storm. The desk just keeps going and duly boots up every day without issues.”
Since most of the programming was done in production rehearsals, a typical show day for the technical team starts with loading in early in the morning, followed by a line-check and a rollback on all the gear. They then return in the evening for another line-check before show time. For Hatt, the absence of backline amps or loudspeakers on-stage and a good separation of microphones to minimalize spill created a pleasant mixing environment.
“The shows we do are dynamic and scene-heavy from a mixing perspective,” says Hatt. “Everyone is on IEMs and there are two drum kits, acoustic and electric, the latter providing what can be quite radically different triggered sounds from one song to the next. It’s actually been great on this tour to really get a bit deeper into the snapshot capabilities of the Vi4. I’ve used this desk as a layered analog console for years, but now I’ve had the chance to deal with it on a more technical level, and I’ve been very pleased with the results.”
Hatt also had the chance of using the Soundcraft Realtime Rack over the course of the tour. It provides UAD studio plug-ins to the Soundcraft platform, while its SHARC-based processors run audio over MADI, and ensures DSP stability and sound quality.
“The Realtime Rack gives me a blank canvas to introduce some very high-end plug-ins to the mix,” Hatt notes. “Though I’m only scratching the surface of this technology on this tour, it’s a very elegant solution, and the more I use it the more I like it. Ultimately, it hooks up easily via MADI connections and is a very exciting addition to the Vi environment.”
Czech Republic’s Sound of Innovation Adopts Soundcraft Vi3000 Digital Console
Sound of Innovation, a leading sound company based in Prague, Czech Republic’s premier live sound contractors, recently added a Harman’sSoundcraft Vi3000 digital live sound console to its inventory.
“Since we were founded in 1990, we’ve been fond of Soundcraft mixing consoles for their exceptional sound quality, going all the way back to the Spirit LX7, and we still find the MH3 to be a dependable workhorse,” says Frantisek Petrik, Sound of Innovation’s owner. “In 2011 we purchased the Vi1, which we found to be a huge step up thanks to its intuitive user interface and ease of operation.”
When the time came to make the move to a larger-format console, Sound of Innovation stepped up to the Vi3000, working with Czech Harman Professional distributor Audio Master CZ. “Audio Master CZ has always been very supportive to us and provided exceptional customer service,” Petrik notes. “In fact, over time they have become our sole technology supplier.
“The advances in technology and internal sound processing are even more apparent in the Vi3000,” he adds. “There are many major as well as incremental improvements. We are totally excited about the sound quality and the ability to manipulate every parameter, while the clear layout of the interface makes the console even more ‘comfortable’ to operate.”
Petrik finds the console’s Dante compatibility to be a major advantage, and appreciates the fact that the console relatively lightweight and fits inside a single transport case.
He also notes, “Although we’re old-school analog fans we are aware of the amazing advancements in digital technology. We’ve always considered the quality of a console’s preamps as the foundation of live sound mixing. We can assure any live sound people who still might be hesitant about entering the world of digital audio that the Vi3000 is going to blow their minds.”
The Alero ALR-AEC-8 is a dedicated microphone mixer, designed specifically for web conferencing applications in medium to large meeting rooms.
AMX, a member of the Harman Professional family, recently announced its new Alero Web Conferencing Audio Mixer to bring premium conferencing experiences to more meeting spaces.
The Alero ALR-AEC-8 is a dedicated microphone mixer, designed specifically for web conferencing applications in medium to large meeting rooms.
PC-based web conferences are becoming increasingly popular because of their ease of use and low cost; relative to traditional, VTC conferencing. This popularity is driving more conferences to larger rooms that can accommodate more guests. But to ensure good audio quality, larger rooms require more microphones and a mic mixer.
AMX designed the Alero Mixer with support for up to eight boundary mics and automatic monitoring and adjustment of audio levels to deliver a great sounding conference. And with included line level inputs/outputs the Alero is scalable to support traditional VTC conferences as well.
The ALR-AEC-8 includes two additional features few mixers offer. A USB audio interface is included for simple, direct output to a PC to enable a Lync, Skype or other PC-based web conference. In addition, onboard web configuration makes the Alero easy to install and configure. The technical training required to set up larger mic mixers has been eliminated and just about anyone can confidently host an Alero-managed conference.
“AMX saw a real need to design a mixer that was specifically targeted for large room web conferencing,” said Shaun Robinson, AMX Vice President of Product Management. “There were plenty of great mixers out there for standard VTC conferences but none that supported easily managed web conferencing. The Alero is not only able to adjust everything on the fly but its drop dead simple to install, manage and use, something we think will be a welcome addition for mainstream users.”
The Alero Web Conferencing Audio Mixer (ALR-AEC-8) is expected to ship in the US and select markets in Q3 2015 with global availability expected Q4 2015.
Church Sound: Correcting Phase Cancellation With A Plug-In
An approach to getting things in phase in your mix, which can make a big difference...
I tend to go back and forth on some of my mixing approaches. For example, I thought I had sworn off using drum overhead mics in exchange for under-miking cymbals, but lately I’ve been flying my good ol’ overhead mic on the drums again.
When I do use an overhead mic, I’ve become a fan of using an X-Y stereo mic to capture more of an overall kit sound than simply cymbals. Using a stereo mic is advantageous for this because it ensures phase coherency at the capsule.
Phase coherency is one of those things I probably overlooked for too many years in my career, until the light bulb finally clicked and I could hear what the lack of it was causing. A phase-coherent overhead mic guarantees that the snare and toms captured by that mic won’t have phase problems resulting from the left and right capsules “hearing” those drums at different times. So, aside from the effects of the mic’s frequency response, I know that the drums coming through that mic will sound like the actual drums.
Phase issues can still come into play with this approach, however, when I start blending in close mics with the overhead. If my overhead mic is three feet above the kit and my snare mic is only a few inches off the snare, the snare will arrive at the overhead roughly 3 ms later than at the close mic.
If the snare in these two mics combines in my mix at more-or-less the same levels, that time or “phase” offset will create phase cancellation—also known as “comb filtering”—in the sound of my snare. If I employ a little bit of math from my Smaart training, I can estimate that the time difference will create a null right around 160 Hz, which means it will take some of the meat out of our drummer’s wonderfully fat snare. Additionally, the time offset can also smear the transient of the snare and dull its punch in the mix.
One of the tools I’ve been using to approach these types of phase issues is the Waves InPhase plug-in. I keep InPhase instantiated at all times on a stereo utility channel. I would love to use it all over the console since there are so many things I like to check with it, but this seems to be the best way to optimize my DSP use on the Avid VENUE.
Screenshot of the Waves InPhase plug-in.
To make use of my one instance, I have a series of snapshots programmed to patch different instruments into the plugin where I might want to check phase. For example, I have snapshots that patch my top snare mic or tom mics into one channel of the plugin and one side of my overhead mic into the other.
I also have snapshots to patch my multiple guitar mics on the same amp so I can compare the phase between those mics. Basically, if I’m using multiple mics on an instrument, chances are good I have a snapshot programmed to patch those mics to InPhase.
So here’s a typical scenario involving my snare and overhead mic. I’ll start my process during sound check or rehearsal by recalling my snare snapshot and pulling the plu-gin up on the console.
A quick click of InPhase’s Capture switch gets the ball rolling with a visual inspection of the waveforms captured from the mics. InPhase’s window makes it very easy to see the difference in time between the two mics, but I try not to get too hung up on what I see because ultimately what we hear is the overriding factor.
Next I’ll click the “Mono Mix” button and solo my InPhase channel up in headphones or on my nearfields to hear what’s happening when those two mics combine. If I like what I hear, I’ll leave things alone, but in most cases I’ll start adjusting the delay on my snare mic to line up its waveform with the overhead right in InPhase’s window. I listen to the combined result as I do this, until the snare gets its body and punch back. Then I’ll take the delay time from InPhase and dial it in on my snare mic before moving on to another set of mics.
I used to employ a more convoluted version of the same process with recorded audio in Pro Tools, trying to measure the distances between peaks in the waveforms, but it was always time-consuming and not something I could easily do until there was downtime. InPhase, on the other hand, lets me do this on the fly while the band is playing, and I get to hear the results in real time. I love this because it saves me time from dealing with the technical side of mixing and allows me to spend more of my time approaching things musically.
If you’ve never experimented with the difference getting things in phase in your mix can make, I highly suggest you give InPhase a try.
David Stagl serves as audio director at North Point Community Church in Alpharetta, GA, and is also the author and publisher of the Going To 11 (www.goingto11.com) blog, which is full of excellent audio “how to” and advice.
Studio Bohemo Enjoys Analog Advatages In Switching Over To An Audient ASP4816 Console
Facility in rural New Hampshire that provides project services to songwriters and musicians transitions over from a digital approach
Studio Bohemo, located in the White Mountains of New Hampshire, recently incorporated an Audient ASP4816 compact analog recording console in switching from a digital approach.
“I suddenly knew what I had been missing,” says studio owner Wes Chapmon, describing his initial reaction after running a mix on his new ASP4816. “The sound was alive, three-dimensional, revealing and much easier to work with.”
A small studio built on the side of a mountain, Studio Bohemo provides comprehensive project services to songwriters and musicians.
“The ASP4816 was a big shift for us,” Chapmon notes, having used a digital desk previously, running four 96 kHz I/O cards to two Lynx AES16 cards on the computer side. “We normally set this up in two layers for 24-24-bit/96k ins and 24-24-bit/96k outs.
“Before, we were locked in at 24-bit/96k,and honestly we usually worked at 24/48 for higher track counts, resources etc,” he explains. “Now we are not limited by the console and are running everything to and from two Lynx Aurora16 converters that are capable of much higher sample rates and fidelity.”
The time to revert to his “analog roots” had been approaching for some time, notes this son of an engineer and a classical pianist.
“I was always surrounded by music: two pianos going and a healthy diet of microphones and recording gear,” he says. “I grew up listening to works created on analog equipment; creating and mixing music on analog equipment.” The final straw came when re-listening to a couple of tracks he’d recorded years ago on an analog desk, which “... planted the seeds of discontent with our digital set up.”
Enter Audient and the compact ASP4816 desk, which with the features of a larger desk and fully-featured inline architecture, fulfilled his desire for his studio to have an analog heart.
“As soon as it arrived—after we managed to get it up the mountain—we ran some basic patches through it just to hear those gorgeous EQs,” he says. “I want a clear and audible shift in the sound so I can make clear and artistic decisions without second guessing or wasting valuable time with ‘maybe this plug-in or maybe that emulation.’ What makes a great EQ is one that with a well-recorded sound can twist it through all the bands and frequencies and shapes, make it sound radically different—yet musical—and potentially useful in every permutation. Maybe I’m just still on the honeymoon but I think that describes the ASP4816 pretty well.
“The inserts are also critical to digital recording,” he adds. “It can be important to have dynamics processing between the preamp and AD converters. A plugin after the fact just can’t do what this does. Audient is one of the few companies to understand this and get it right, as seen through the product line, especially the iD22 and ASP880.”
Studio Bohemo is a product of Chapmon’s dream “...to build an artist retreat/studio in a contemplative setting.” He continues, “We didn’t know anyone here or have a job and neither of us were from the northeast. We literally looked at properties online in various places that inspired us, took a few weekends to visit them and picked this place to relocate and build.”
Sixteen years on and this bold move is paying off. “We’ve worked with several local artists and a few artists have flown in for projects and a welcomed retreat. We have a few things on the go at the moment, including a multi-album release of restored and re-mastered material from jazz great and friend Betty Johnson who sang with Sachmo (Louis Armstrong).”
In Bohemian style, Chapmon is content living ‘in the now’ and is hugely driven by his latest endeavors with his new console. “Really, though I’m always most proud of the artists that put themselves in that personally vulnerable place behind a microphone because of some compulsion to bring art and beauty into the world. Whatever the outcome, that always blows me away.”
South Africa’s Black Coffee Heats Up Inventory With Soundcraft Vi3000 Digital Console
Long-time Soundcraft supporter adds Vi3000 to handle larger and more demanding projects
Black Coffee of Durban, South Africa recently added a Harman’sSoundcraft Vi3000 digital console to its inventory to handle larger and more demanding projects.
The events company has been a long-time Soundcraft supporter, still using the company’s folio mixers for smaller and non-complex applications.
While Black Coffee has been happy with its other Soundcraft digital consoles (including the Si Compact and Si2+ models), owner Brandon Bunyan and head of sound Brad Ellapen wanted to up their offerings with the latest mixing console technology and quality.
“We chose the Vi3000 based on our passion for Soundcraft products, in the reliability and ruggedness of the products on the road, and the compatibility of the Vi3000 not only with their existing stage boxes, but with other Dante-enabled products as well,” Bunyan says.
Black Coffee currently owns 31 mixing consoles, with the majority being Soundcraft, complementing the company’s range of Harman Professional products including Crown I-Tech HD amplifiers and JBL VRX, PRX and SRX Series loudspeakers.
The Soundcraft Vi3000 offers a new ergonomic design, 96 channels to mix, internal SpiderCore 40-bit floating-point DSP, Dante compatibility and extensive connectivity and routing options, as well as a host of built-in Lexicon rever-delays, dbx compression and effects. The control surface provides four Vistonics II touchscreen interfaces so that it that can be used simultaneously by two engineers at the same time.
Recently I had an experience I’ve had before. I was working on mixing down a song we did a few years ago, and I just couldn’t get it working.
I do this stuff for fun now that I have more free time, and I enjoy playing with different techniques in the studio that I wouldn’t be able to do live. I had been working on the mix for quite a while, and it wasn’t happening.
I rendered it out, sent it through my mastering process then went and listened in a few spaces. Nope. Not working.
Back in the studio, I kept picking at it, but it wasn’t getting better. Finally, I took the nuclear option. I saved the file, renamed it and started over.
I pulled out all the plug-ins, muted all parallel processing and pulled all the faders to off. I began to re-build the mix from scratch, doing only as much processing as I absolutely needed.
Within an hour, it was sounding pretty dang good. Another hour later and I was really digging it. A test mix down revealed a few things to tweak, but overall, it was finally where I wanted it.
The first time I saw this done live was about 10 years ago. I was at church, working with a guy on the sound team. He’d been a touring engineer in a past life, and generally knew what he was going.
But that day, the mix wasn’t working. We both tried to fix it, but we just couldn’t get it there. Finally, in what I saw as an act of desperation, he just pulled all the faders to off. “That’s it,” he said, “I’m starting over.”
For the next few minutes, he rebuilt the mix channel by channel. And when he was done, we looked at each other and nodded. It was working. I’m not entirely sure what changed; the board didn’t look that different from where it was before he killed the mix. But it was different enough.
Sometimes, we can get ourselves off in deep weeds and lose sight of what we’re trying to do. And, like being lost in a field of deep weeds, we can keep going, but never get to our destination because we can’t see it.
There’s so much noise happening in our minds at that point that nothing works right. Pulling all the faders down is like having a giant brush hog come in and mow the field. Finally, we can see where we’re going.
Clear The Decks
When you clear the faders, you can re-start the mixing process. This is like re-booting your computer. You get a fresh start at the mix.
Now, you can start from the rhythm section as I often do, or start with the vocals. I’m not sure one way is right or better than the other. Maybe try both and see what works better for you.
I tend to think in terms of a foundation of drums and bass, layer in guitars and keys, then put vocals on top. But others prefer to work the other way.
The funny thing about this process is that most of the time, you won’t be able to tell what was wrong with the mix before. But it will be obvious to all that it is better.
As a word of caution, if your band is on wedges and not in-ears, you may want to warn them before you do this. If you pull down the house during a song, the sudden loss of volume from the house may freak them out.
And while it probably goes without saying, do this during rehearsal, not the service.
Mike Sessler now works with Visioneering, where he helps churches improve their AVL systems, and encourages and trains the technical artists that run them. He has been involved in live production for over 25 years and is the author of the blog Church Tech Arts.
Before there was digital recording, before spring reverb, even before analog tape, there was EQ. Equalization is one of the oldest tools in the audio engineer’s arsenal, and one of the most useful.
Used judiciously, EQ can do wonders to de-clutter a crowded soundscape. Used with precision, it can remove offending sounds we hadn’t necessarily intended to capture. Used correctly, a bit of EQ can be all that’s needed to make peace between dueling guitars, scoop the mud from the heaviest drums, or make a mundane vocal stand up and shine.
But all too often, EQ is misused and misunderstood, typically in a vain attempt to fix a poor recording.
Rule Number One in recording still applies: garbage in equals garbage out. A little EQ is great for helping make a good track sound better, but no amount of EQ will make a bad track sound good.
The best mix starts with the best recording, so try to capture the best sound you can to begin with. Move mics, listen at the source, not just in the control room or in your headphones. Make sure what’s being recorded sounds as close to what’s being played as possible – before it’s too late to do anything about it.
Your ears are the bottom line when it comes to applying EQ. While we can talk about a few general principles, every instrument has its own unique characteristics and timbre, and will react differently to boosting or cutting specific frequencies. So take these and all suggestions with a few grains of salt; use them as a starting point but make your decisions based on what sounds good.
EQ Giveth & EQ Taketh Away
When it comes to EQing, less truly is more, and in nearly all cases it’s better to take away than to add. Many less experienced users have a tendency to make an instrument stand out by boosting frequencies, but the cumulative results can be dangerous.
Adding just 2 dB of gain to two different instruments means that when they excite the same frequencies (and trust me, they will, and probably at the worst possible moment), you’ve got 4 dB of gain. Add too much EQ and your mix can easily turn to mud. It’s often a better idea to try attenuating those same frequencies in other instruments instead.
Another good reason to minimize your use of additive EQ: while cutting frequencies is a passive process, boosting frequencies makes your EQ function as a preamp within the signal flow. Adding any preamp means adding noise and distortion, and the preamps in most EQ circuitry are less than optimal.
All those arguments aside, sometimes it’s simply more effective to boost one element of the mix, rather than rolling off dozens of others. Once again, the operative word here is moderation – a little boost of 1 or 2 dB goes a long way.
EQing Drums – If It Doesn’t Fit, You Must EQ It
If your mix includes drums, it’s a good bet you’ll spend a considerable portion of your mixdown time EQing them. Because drums cover such a wide tonal range, there’s plenty of other stuff in the mix that can compete with those frequencies. Kick and snare in particular tend to be prominent parts of the song’s sonic fabric, and when it comes to helping them play nicely with other instruments and vocals, EQ is your best friend.
Of course, assuming you’re working with a live drum kit (as opposed to isolated drum samples), you’re not working in a vacuum. Since every drum track also contains leakage from other mic, boosting a frequency on one track can also bring up the off-axis sounds of adjacent mics, potentially creating more problems than it solves.
For a dull sounding kick drum, adding a slight boost anywhere around 80 Hz to 120 Hz will produce more boom and a more rounded “thud.” (Typically, the kick tends to compete with the bass guitar for that frequency range, and it’s a good idea to decide which of the two should occupy the lower and upper edges of that zone. See the section on bass later in this article for more on this.)
Adding a tiny bit of 500 Hz can bring out the “click” of the beater hitting the drum head, and can be helpful in preventing the kick from disappearing once your track hits the listener’s earbuds in the inevitable low-fi MP3 version.
Snares come in such a wide range of sizes and materials, it’s a bit tough to generalize about frequencies.
But the sound of the snare wires rattling lives in the 5 kHz to 10 kHz range, and a bit of gain there is great for brightening up a dull snare. If you’re plagued with a boxy sounding snare, try rolling off a bit of 300 through 800 Hz.
With toms, a common mistake is to try boosting low end to make them stand out. Adding a couple of dB at 100 Hz will increase their power, but at the expense of muddying the mix. A better strategy for perking up those tom fills is to leave the bottom end alone and add a tiny bit of 5 kHz to bring out the attack. And as with the snare, play around with rolling off that same 300 through 800 Hz range to eliminate boxiness.
Almost every tom has a resonant ring, and some can be problematic. Of course, the basics apply: tune the toms first and foremost to reduce or eliminate ringing. Whatever problem resonance remains can be addressed using a surgical approach with a multiband EQ. Select a narrow Q and boost the gain as you sweep the midrange band. When you locate the offending frequency, apply a few dB worth of cut to make it go away.
Overhead mics can be a mixed blessing. Their position and relative distance from the kit makes them great for adding air and ambience, but loud cymbals can overpower the mix. Try adding a bit of 10 kHz to brighten the track, and then backing off the overall level to get the air without too much metal.
The Bottom Line On Bass
Since bass and kick occupy the same frequency range and (hopefully) work together, it’s almost always necessary to use EQ to differentiate them in the mix. As mentioned earlier, it’s best to pick one as the rounder, bottom-y sound and make the other a bit more bright and punchy; which is which will be dictated by the song.
One of the questions i hear most often is whether it’s best to record the bass direct or mic the amp. The answer, as you might expect, is “it depends.” Ideally, many engineers opt to record both the amp and a direct track simultaneously, balancing them in the mix for the best possible tone.
Of course, in today’s project studio world, it’s not always possible to record live at the volume you’d like. If you’re working with a bass track that was recorded direct, chances are it’s a bit flat and nondescript compared with a mic’ed bass amp. The good news is, that flatness will ultimately make EQing the DI track far easier, since there’s less coloration to begin with.
Like the kick drum, boosting the 80-120 Hz range on an electric bass will add roundness and bottom end. To add presence and attack, go for a slightly higher range than with the kick, around 1 kHz. Don’t add too much or you’ll bring out the finger noise as well.
Making Space For Guitars And Keys
Guitars are among the most versatile instruments; that same versatility can make them a real challenge. With electric guitars, if you’re fortunate to have a player who knows their amp and their sound, your best bet is to change as little as possible.
If you’ve got two rhythm guitar parts going, a bit of panning and EQ can help distinguish one from the other. Try a slight boost at around 100 Hz on one to bring up the lower mids (with perhaps a corresponding cut on the other guitar). Experiment with higher frequencies on the second part – boosting different frequencies between about 750 Hz and 10 kHz will each bring out a different type of sparkle. Scooping out a bit of 250 to 500 Hz can help eliminate some harshness and woofiness.
Acoustic guitar is a very different animal. Each has its own unique tone and timbre, and much will depend on the player, the sound of the room, the mics you’ve used and where you’ve placed them. A mic too close to the sound hole will deliver a boomy sound; a slight cut at 100 Hz can help. Close miking can also pick up some boxiness from the wood’s resonance, especially around the midrange. Try dropping a bit of the 300 to 400 Hz range. And of course, bring out the shimmer and strumming sound by boosting the upper ranges, from 750 Hz up to around 10 kHz (watching out again for finger noise).
Acoustic pianos, like their guitar counterparts, are organic instruments subject to a number of unique conditions. Every piano has its own character and tone to begin with, further affected by the room, the mics, mic placement and of course, the player. Few instruments cover as wide a range of frequencies and overtones as the piano, which can be both a blessing and a curse. What you do with regard to EQ depends largely on the song - a dense part with close-clustered chords is probably best treated with subtractive EQ, while a spare, melodic passage might benefit from a bit of boost in the upper mids.
Keyboards are a whole other issue, and could easily be the subject of an entire article alone. Synths cover such a wide range of sounds, it’s impossible to generalize about what will work on any given patch. For the most part, you’re quite literally playing it by ear.
Listen Before You Look
I’ll close with the same point I opened with: take this and all advice as nothing more than suggestions. There are no hard and fast rules except one: use your ears. If it sounds wrong, it probably is. So close your eyes and listen. Adjust your EQ, close your eyes and listen again. Don’t just solo the track, either - listen to your changes in the context of the whole mix.
Especially in today’s DAW-oriented world, we all have a tendency to stare at the screen. But it’s important not to depend on spectrum analyzers and meters instead of listening. Try out these suggestions, but then try something totally different. Innovate - don’t blindly follow. Every song is unique, every instrument and room is different, and every artist and song is unique.
What worked for one person on one recording won’t necessarily be what’s right for you.
Daniel Keller is a musician, engineer and producer. Since 2002 he has been president and CEO of Get It In Writing, a public relations and marketing firm focused on audio and multimedia professionals and their toys. Despite being immersed in professional audio his entire adult life, he still refuses to grow up. This article is courtesy of Universal Audio.
From a borrowed garage space in Baton Rouge to 44,000-square-foot building, more than 130 employees and additional offices in the U.S., Ireland and Hong Kong
It’s been a long road since Jim Odom and his former high school and Louisiana State University classmate Brian Smith built and sold their first piece of professional audio equipment back in 1995. Working in a borrowed garage space in their home city of Baton Rouge, Louisiana, they created the first affordable product that could digitally control analog audio signal dynamics.
“At the time we designed our DCP-8, other options were esoteric gear costing tens of thousands of dollars,” Odom recalls, “We were recording and wanted the same equipment as the big studios but we couldn’t afford it, so we built our own, starting PreSonus with three employees and an output of five or six units a week. Then and now, our philosophy has been to design products that people can afford.
“The DCP-8 had reasonable success but missed on some features the customers really wanted,” he continues. “We were close but had to listen more closely to our customers. It was an enormous lesson for me as a product designer. Our second version, the ACP88, has been a tremendous success and remains part of our product line today.”
Moving twice to larger quarters, Odom and Smith went on to introduce products such as DigiMax (2000), the first 8-channel microphone preamplifier with ADAT Lightpipe output; Central Station (2004), the first monitor controller with talkback; and some of the first multi-channel FireWire interfaces, including the FirePod (2004), FireStudio series and AudioBox USB (2008).
According to Odom, “We were providing a piece of the solution with our hardware but we weren’t satisfied with existing digital audio workstations.” In 2006, a Hamburg, Germany, startup company founded by ex-Steinberg developers began working on Capture and Studio One in cooperation with PreSonus. The partnership brought great results, leading to the acquisition of the company as PreSonus Software Ltd.
By 2008, sales were up to $20 million annually, and Odom decided it was time for more professional leadership so that he could assume the role of chief strategy officer. Longtime industry veteran Jim Mack was hired as CEO. According to Mack, “Our overall strategy was to marry software and hardware to a degree never done before in the industry.”
In 2009, the company launched its StudioLive series of digital mixers, which is integrated with bundled PreSonus Capture live-recording software (2009), Studio One Artist DAW (2009), Virtual StudioLive bidirectional control software (2010), QMix monitor-mix control software (2012), and a custom version of Rational Acoustics’ Smaart audio-analysis software.
In 2014, PreSonus addressed the software/hardware mixing relationship by introducing UC-Surface touch-controlled software for the new StudioLive RM-series rack mixers. This software will be ported to support StudioLive AI console mixers in early 2015. The adoption of Dante and AVB networking support for StudioLive AI mixers, loudspeakers, and WorxAudio line arrays will allow the company to provide even more live sound solutions in 2015 and beyond.
Today, the long road from the garage has led to a new 44,000-square-foot building with a Walters-Storyk-designed recording studio at its heart. With an employee count of more than 130, PreSonus also has offices in Ireland and Hong Kong, and subsidiaries WorxAudio and Nimbit have offices in North Carolina and Massachusetts, respectively.
Turbosound Launches iX Series Loudspeakers Offering Onboard Remote Controllable Mixing
Self-powered loudspeakers outfitted with Klark Teknik digital processing, Class-D amplification, mixer and Bluetooth audio streaming
Turbosound has launched the new iX Series of loudspeakers, equipped with Klark Teknik digital processing, Class-D amplification, and an onboard 2-channel digital mixer that’s remote controllable via dedicated iPhone/iPad app or locally via dedicated LCD-based user interface.
A proprietary dual angle pole mount socket allows the enclosures to be pole mounted above subwoofers either straight on, or raised higher and angled downwards.
Both models in the series, the iX12 (12-inch) and iX15 (15-inch), also include wireless Bluetooth stereo audio streaming. DSP presets have been optimized for a wide range of applications and orientations, as well as for seamless integration with subwoofers and enhanced intelligibility in floor monitor applications.
Both of the 2-way models are driven by amplification of up to 1,100 watts. The Klark Teknik DSP provides a range of presets and dynamic EQ to enhance full-range response at both lower and higher output levels.
Estimated U.S. “street” pricing: iX12—$349; iX15—$449. Both models are covered by a 3-year warranty.
U.S. Retirement Community Specialist Selects Symetrix Processing For New Development
SymNet Radius and Solus open architecture DSPs are integral to the audio configuration
Lutheran Senior Life is a leading name in senior living and health care, and operates a number of CCRC properties throughout the state of Pennsylvania.
Pittsburgh-based integration firm ASCC’s Pro A/V arm has implemented Symetrix processors to provide suitably flexible audio in a number of Lutheran Senior Life properties, and in its latest project has installed Symetrix SymNet Radius and Solus DSPs at the Passavant Retire Community complex in Zelienople.
In a move designed to better serve its residents, Passavant recently completed work on a new site, the Abundant Life Center. The facility hosts a chapel, fitness center, restaurant, conference area, retail outlets, independent living and personal care apartments, nursing beds and more.
To deliver anywhere-to-anywhere source routing and intuitive local control at the Abundant Life Center, ASCC made extensive use of Symetrix technology. A SymNet Radius 12x8 DSP resides at the heart of the system, and an xIn 12 and an xOut 12 connect to it via Dante, bringing the main system I/O count to 24x20.
One Symetrix SymNet Solus 8 open architecture DSP each resides in the chapel and in the conference complex, and although those systems usually operate autonomously ASCC’s Ken Coey and Ryan Hesske designed the system so that users could still tap or send signals to or from anywhere else in the system via analogue feeds and interactive preset changes.
In addition, seven Symetrix ARC-2e wall mounted remote controls, spread across the zones, permit easy source selection, volume adjustment and preset changes. Paging centres located at the main reception desk, fitness centre reception desk and media area use a Symetrix ARC-SW4e paired with an ARC-EX4e wall panel remote to allow intuitive paging to any zone in the building.
With regard to the five principal areas that comprise the Abundant Life Center, Coey – who is pro A/V design specialist at ASCC – says that “it made sense to tie all the systems together. That way, in addition to being able to page specific areas or across the entire system, they could send programme input sources from anywhere to anywhere else. It’s very flexible, and yet with the ARC controllers still very easy for them to operate.”
A long-time user of Symetrix systems, Coey remarks that he has “always been impressed with the quality, functionality, reliability and ease-of-use of their products. Moreover, they sound great and we’re able to programme the user interfaces so that staff can use them intuitively.”
Inputs to the system include seven Tascam CD-200i CD players with integrated iPod dock, a Tascam TU-690 AM/FM radio, an Antex XM-100 Satellite radio, three Audix ADX-12 gooseneck paging microphones, 15 Sennheiser evolution wireless systems, three Sennheiser SKM-Series wireless microphones, four Galaxy Audio HSE-UBG-SENN headworn microphones and a PreSonus StudioLive 16.0.2 digital mixer.
Eleven QSC ISA-, CX- and RMX-Series amplifiers power a large collection of QSC AD-series surface-mounted loudspeakers and Quam System ACT-Series lay-in loudspeakers, as well as a handful of Electro-Voice ZX1-SUB subwoofers and SoundTube RS400i pendant loudspeakers. Furman sequencers condition the system, while Listen Technologies provides assisted listening.
Yamaha Unveils AG Series Compact Hybrid Mixers & USB Audio Interfaces
Designed for general audio applications as well as webcasting, podcasting, gaming and music production
Yamaha has introduced AG Series hybrid mixers and USB audio interfaces, which includes two models (AG03 and AG06) that are designed for general audio applications as well as webcasting, podcasting, gaming and music production.
The AG Series offers high-resolution (24-bit/192 kHz) audio recording and playback, iOS compatibility and battery operation.
A key feature of the AG Series is a TO PC switch that allows users to select which inputs get routed back to a desktop or iOS device. Selecting Dry CH 1-2 allows computer or iOS recording of inputs 1 and 2, while the INPUT MIX switch routes all inputs to the computer or iOS device via USB for standard music production applications.
The LOOPBACK function in the TO PC section, which is particularly handy for podcasters, routes all inputs to the stereo USB output, along with the USB input from the computer. This makes mixing mics and instruments with music beds, sound effects or the audio from computers and iOS devices a breeze.
Users are also able to adjust their sound in real-time with the included headset interface and hardware controls, removing the need to open a software control panel.
“USB audio interfaces typically rely on software to control routing, effects and monitoring,” says Nate Tschetter, marketing manager, Music Production, Yamaha Corporation of America. “The AG series eliminates the need for software and brings easy-to-use routing and DSP controls to the front panel. Combine that with high resolution 24-bit, 192kHz audio and you have a device that serves as an audio Swiss Army Knife.”
The AG series also includes Yamaha D-PRE microphone preamps and onboard DSP effects, along with an AUX input for a music player and foot switch jack to control effects. Separate headphone/monitor volume controls are provided as well.
The layout brings hidden or difficult-to-find software controls to the front panel with easy-to-use faders, knobs and switches.
The AG06 offers six (6) channels: a main input optimized for vocals with a D-PRE mic preamp, a second flexible channel for another mic or a guitar and two stereo inputs for keyboards or other line level sources. The second channel features 1-TOUCH AMP SIM DSP to optimize guitar tone at the touch of a button.
The AG03 has three (3), with a single 60mm fader for the main input with D-PRE mic preamp and a Hi-Z input that allows direct connection for guitars and basses.
New Yamaha AG Series mixers are expected to ship in April 2015. MSRP: AG03—$199; AG06—$249.
Allen & Heath GLD-80 Consoles Deployed For House, Monitors On Tour By Of Mice & Men (Video)
Both engineers needed a mix solution that could handle larger arena shows while keeping to a compact footprint
California band Of Mice & Men, recently out in support of Linkin Park on the European leg of the Hunting Party arena tour, were mixed with Allen & Heath GLD-80 consoles at both front of house and monitors.
It was the first experience of using GLD for engineers Dave Nutbrown (front of house) and Ian “Squid” Walsh (monitors), who both needed a mix solution that could handle larger arena shows while keeping to a compact footprint.
A GLD-80 connected via Cat-5 to AR2412 and AR84 AudioRacks provided 32 XLR inputs for stage sources and delivered five in-ear mixes, as well as mixes to side fills and a pair of center wedges.
The consoles were fitted with Dante cards, and mic preamp signals were split digitally via Dante networking over Cat-5 to front of house. The analog gain was set at the start of the tour, with the engineers using the mixers’ digital trim (+/- 24 dB) to independently make adjustments during the shows.
‘’It’s really easy to use,” Walsh notes. “It’s such a small rig, we were in and out of there so fast.”
Nutbrown adds, “We’re happy with it—it’s a good, all round rock ‘n’ roll board. Our show finishes at 8 o’ clock. I carry it back to the bus at 8.30 and at 8.45 I’m in the bar. What can I tell you?”
Find out further details from both engineers in the following video.
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