Tuesday, September 02, 2014
Second Edition Of “Small Signal Audio Design” By Douglas Self Now Available From Focal Press
Provides an extensive repertoire of circuits that can be put together to make almost any type of audio system.
Focal Press has just released the second edition of Small Signal Audio Design by Douglas Self, providing ample coverage of preamplifiers and mixers, as well as a new chapter on headphone amplifiers. The handbook provides an extensive repertoire of circuits that can be put together to make almost any type of audio system.
Essential points of theory that bear on practical performance are lucidly and thoroughly explained, with the mathematics kept to a relative minimum. Self’s background in design for manufacture means that he keeps a wary eye on the cost of things. The book also includes a chapter on power-supplies, full of practical ways to keep both the ripple and the cost down, showing how to power everything.
The book also teaches how to:
—Make amplifiers with apparently impossibly low noise
—Design discrete circuitry that can handle enormous signals with vanishingly low distortion
—Use humble low-gain transistors to make an amplifier with an input impedance of more than 50 Megohms
—Transform the performance of low-cost-opamps, how to make filters with very low noise and distortion
—Make incredibly accurate volume controls
—Make a huge variety of audio equalizers
—Make magnetic cartridge preamplifiers that have noise so low it is limited by basic physics
—Sum, switch, clip, compress, and route audio signals
The second edition is expanded throughout (with added information on new ADCs and DACs, microcontrollers, more coverage of discrete op amp design, and many other topics), and includes a completely new chapter on headphone amplifiers.
Author Douglas Self studied engineering at Cambridge University, then psychoacoustics at Sussex University. He has spent many years working at the top level of design in both the professional audio and hi-fi industries, and has taken out a number of patents in the field of audio technology. He currently acts as a consultant engineer in the field of audio design.
Find out more and get Small Signal Audio Design, 2nd Edition here.
Church Sound: The Art of Snare Mixing…And A Frog
I created a frog. It wasn’t intentional. Naturally, I’m not talking about a real frog but just look at that photo to the left!
You’ll never read a mixing book that says, “Make the snare’s EQ curve look like a frog in water.” If you do, immediately stop reading the book. Seriously, when it comes to snare mixing, the last place you want to be is behind the mixer.
There are three factors in creating a good snare drum sound.
1. The Snare Drum
Snare drums don’t all sound the same just like all acoustic guitars sound different. Even with a house drum kit, a drummer might bring their own snare because they like it’s sound.
Know that each snare has a unique sound. This is the baseline sound for the mix. Use the same mic and the same EQ settings with two different snares and you’ll get two different results.
Consider these three different snares:
—PDP Blackout Maple Snare
—PDP LTD Classic Wood Hoop Snare
—Pearl Chad Smith Signature Snare
Just by looking at them, you can almost hear the tonal differences:
Material composite, drum size, drum head skin, all of these are factors. Even tuning makes a difference. Snares can be tuned to match whatever the drum tuner decides is to his liking. To generalize, there can be a low or high tuning (true for any drum).
Stand near the drum kit while the drummer plays the snare. This is the sound you’ll be mixing with—not against. Don’t try making it sound like something it’s not.
2. The Microphone
A microphone should be paired with an instrument and so it is with miking the snare. The Shure SM57 pairs great with a snare drum because of its polar pattern and frequency response. I polled some techs and their pairings include the Telefunken M80, Heil PR 22, Heil PR 28, DPA 4099, and the Granelli Audio Labs G5790, a modified Shure SM57 designed for tight spots.
And don’t think mic designs are the same:
Photos are nice but let’s get real—we need to look at specifics. They can have different polar patterns, different sensitivity, and they don’t have to all be dynamic mics. For example, the Heil PR28 is a dynamic mic while the DPA 4099 is a condenser. While all of these characteristics do make a big difference in how a mic treats sound, frequency response is a major factor never to be overlooked.
The frequency response of a microphone alters the tonal characteristics of the snare drum. Take just one snare drum from above, like the Pearl Chad Smith Signature Snare, and mike it with three different mikes. The result is three different sounds. And we haven’t even touched the EQ portion.
For comparison, here are the frequency response charts for the Shure SM57, Heil PR28, and the DPA 4099 (Note the charts with multiple lines are showing the differing frequency responses when not on-axis with the sound source):
Note that snares can be miked both on the top and bottom. Here are a few combinations folks sent me this week;
—Ben Salzmann: Shure Beta57 on top, Shure SM81 on bottom
—Daniel East: Audix i5s both top and bottom
—Micah Webner: Audix i5 top, SM57 bottom
—Deron Yevoli: Sennheiser MD421 on top, Heil PR31BW bottom
—Jamie Ivey: Heil PR22 on top, Sennheiser e904 on bottom
3. The EQ Work
I told you this is the last place you needed to be and now you know why. EQ’ing can only happen after we listen to the natural tone of the snare and consider the mic(s) we’re pairing with it.
Here’s an example: take a snare tuned high and pair it with a mic that has a large high-end boost. Want to cut the highs in the mix? It’s not going to be easy as you’re mixing against what is being sent, not mixing with it.
At this point, you can be as simple or as creative as you want. How do you want to mix a dual-miked snare? How do you want the snare to sound for the song?
It’s not a matter of “how do I use the equipment?” it’s a matter of “what would sound right and how do I get there?” By having the right snare and microphone combination, you’ve got the hard part out of the way. (I know this isn’t always within your control.)
I like a single-miked snare. That’s not to say I won’t fall in love with a dual mic setup next weekend. A single mic setup is a good place to start. By establishing a good single-mic sound, moving into two mics, you already know how to get a good sound from one. Make sense?
Sum Of The Parts
The guidelines listed below are a starting point in mixing the snare. All of the sounds of the drum kit (and the whole band for that matter) have to be considered. The right sound for the snare for a particular song might be really flat on its own. (I cover this idea further here.)
Here are some mixing ideas based on my experience:
High-Pass Filter (HPF)
A mic like the SM57 has the low-end rolled off. I will roll off a bit more if I notice a positive impact on the sound. I’m not going to roll off more just to then flatten the snare sound. If you’re running an analog board, hit the HPF switch and listen for a difference.
It’s a good idea to remove low-end frequencies from all mics that aren’t focused on a low-end instrument. For instance, use an HPF on vocal mics. Snare and cymbal mics, which aren’t focused on low-end kit pieces, are another good place for using an HPF.
I’ve tried gating my snares but have never been happy with the results—at least for a general snare sound. I’ve gated the snare for a song to get a specific sound, but for all-around mixing, I tend to skip it.
Out With The Bad
I’ll sweep my mid-range with a 6 dB cut and find the area of offending frequencies. It’s that area you drop and suddenly you think, “now that sounds much better.” In the case of the frog EQ I mentioned in the beginning, I didn’t find that spot and rather found a huge boost is what was needed. Some days it’s like that.
Sculpt To Fit What You Hear In Your Head
At this point, you should have a sound in your head that you want the snare to match. It’s that internal reference sound. You know what sounds good, you just need to make it a reality. Is there too much snap? Not enough? Is it perfect the way it is? (This is where that snare/mic pairing pays off.) Or does it still need some work?
And here’s where I have to say I’m sorry. I’m sorry I can’t tell you exactly what to boost or cut and where to do it. It all depends on your snare and your mic and your room and your drummer and…eh, you get the point.
That said, here are a few places to start:
—Snap and presence, 3 kHz to 12 kHz. The higher you go, the less presence and more snap
—Body, sub-500 Hz if you need to give it some substance
The Take Away
Know the tone of the instrument, pair it with the right microphone, and then step behind the mixer. Only then can a good snare mix be created.
Ready to learn and laugh? Chris Huff writes about the world of church audio at Behind The Mixer. He covers everything from audio fundamentals to dealing with musicians, and can even tell you the signs the sound guy is having a mental breakdown. Chris is also the author of Audio Essentials For Church Sound, available here.
New PreSonus StudioLive RM-series Rack-Mount Digital Mixers Offer Recallable Touch Control (Video)
Based on the StudioLive AI-series engine and controlled with UC Surface software for Mac, Windows, and iOS
Based on the StudioLive AI-series engine and controlled with UC Surface software for Mac, Windows, and iOS, the new PreSonus StudioLive RM16AI and RM32AI 32x16x3 rack-mount Active Integration digital mixers are scalable, compact, and recallable.
New UC Surface control software was specifically designed to be an interface for live mixing but is also suited for a studio environment. The layout of the mixing features provides quick, intuitive access to key parameters.
The 3U rack-mount RM16AI provides 16 locking XLR inputs with recallable XMAX Class A preamps, 8 XLR line outs, and 3 main outs (left, right, and mono/center), 32 internal channels and 25 buses, a 52x34 FireWire 800 recording interface, 96 kHz operation, and extensive signal processing.
The 4U rack-mount RM32AI offers 32 inputs with recallable XMAX preamps and 16 line outputs but otherwise has the same features as the 16-input version. Both mixers offer individual 48-volt phantom power on all inputs, and a 48-volt meters button displays phantom-power assignment on the input meter grid.
The RM-series mixers’ Active Integration technology includes direct Wi-Fi and Ethernet networking and integrated Capture recording software with Virtual Soundcheck mode and Studio One Artist DAW for Mac and Windows.
Fat Channel signal processing is provided on all input channels and all buses, including a 4-band parametric EQ, compressor, gate, limiter, and more. However, unlike StudioLive AI mixers, the RM series’ recallable XMAX preamps mean that every parameter on the mixer can be saved and recalled.
The front panel provides input signal-present and clip LEDs (16 for the RM16AI, 32 for the RM32AI) and provides stereo RCA tape inputs, a headphone output with volume knob and source-select buttons, a mute all button that temporarily mutes all inputs and outputs, and a USB type-A jack that hosts the included Wi-Fi LAN adapter.
The rear panel contains an option slot that comes with two FireWire 800 ports, an Ethernet control port, and S/PDIF I/O. The slot also accepts the same option cards as the StudioLive AI-series digital mixers; with Dante, AVB, and Thunderbolt cards coming soon, StudioLive RM-series mixers can easily be updated to keep up with the latest networking technologies. Also on the rear are a power jack and switch, MIDI I/O, and a mirror of the line outputs on a DB25 connector for connecting to wireless in-ear systems.
In addition to UC Surface, the integrated software bundle includes PreSonus Capture live-recording software with true Virtual Soundcheck and Studio One Artist DAW. StudioLive RM-series mixers also work with PreSonus’ free QMix-AI aux-mix control software for iPhone/iPod touch, enabling musicians to control their own monitor mixes. An extensive library of tutorials and downloads completes the package.
Tuesday, August 26, 2014
New Launch Control XL Hands-On Controller For Ableton Live Now Available (Video)
Similar to Launch Control but provides twice as many buttons, an extra row of knobs, and eight 60 mm faders for high-precision level control
Novation is now shipping Launch Control XL, a new hands-on controller for Ableton Live. It’s similar to Launch Control but provides twice as many buttons, an extra row of knobs, and eight 60 mm faders for high-precision level control in integrating seamlessly with Live.
The knobs are laid out in three rows of eight, just like Ableton’s mixer interface. They also have multi-colored LEDs that illuminate to distinguish between sends, EQs or any other device.
Buttons, knobs and faders can be assigned to any parameters within Ableton, allowing users to make their own layouts. Knobs can also be customized by users with custom colors, and effortlessly switch between their own mappings and Live’s built-in functionality.
Combined with Launchpad S, it provides hands-on control over everything in Ableton at once: session view, mixer, effects and instruments. It’s bud powered, with no drivers needed.
The new Launch Control XL comes with Ableton Live Lite and a library of loops and samples from Loopmasters. And, it isn’t limited to Ableton Live, also capable of controlling other major music software, or any MIDI-compatible iOS software.
Knobs: 24 rotary pots
Faders: Eight 60mm Faders
24 assignable buttons
2 template switch buttons
26 LEDs in the buttons and 24 LEDs under the knobs
Kensington security slot
Mac OS X 10.9 Mavericks, Mac OS X 10.8 Mountain Lion
Windows 8.1, Windows 8, Windows 7
iOS 7, 6
USB-MIDI class compliant
Live 9.1.3 or later required (As with all new products with Ableton Live support, Ableton provide support only for the latest version of the latest generation of Live)
Thursday, August 21, 2014
The Blackbird Academy Engineering Program Integrates Roland’s Personal Mixing System
Roland's M-48 personal mixers are used every day in Studio I at The Blackbird Academy.
The Blackbird Academy, a premier studio and live sound engineering school in Nashville, has recently installed and integrated the Roland M-48 Personal Mixing System into their program.
The Academy’s unique “hands-on” approach curriculum provide students access to world-famous Blackbird Studio gear and engineers.
“At The Blackbird Academy, our prime emphasis is teaching the students to provide the client with high-quality sound and service throughout the production process,” says co-director and instructor Kevin Becka. “The compact, M-48 mixers punch this ticket by being easy to use, having a full set of features like EQ, reverb, panning and level controls, plus they sound great.
“I have experience with this system in a live sound setting and am amazed at how well it fits into our tracking, and overdub sessions at The Blackbird Academy.”
The M-48 mixers are used every day in Studio I at The Blackbird Academy. The studio is built around a Beatle theme, and like Abbey Road, the control room for Studio I is on the second floor. When they built the room, it was going to be a daunting task to drag copper downstairs to all the live room panels. Becka was happy that the Roland M-48 system worked with their existing Cat-5 runs.
The M-48’s personal mixers have proven to be easy for the students to grasp and get up and running quickly. With the bankable channels and multi-function encoders, its effortless for student/engineers to show the musicians how to jump between the functions and concentrate on getting themselves a great mix. The three-band EQ and reverb parameters are simple but sound great and are very usable.
Becka concludes, “We are very happy with how the M-48 works with our curriculum, studio workflows and how the students have taken to the system – they are very impressed and so are we.”
Roland Systems Group
Posted by Julie Clark on 08/21 at 10:18 AM
Monday, August 18, 2014
Backstage Class: Alternative & Effective Approaches To Sound Check
So much of what we do as sound engineers is based on habit and repetition. Better safe than sorry, if it ain’t broke don’t fix it, that’s the way everyone does it, and so on.
I enjoy questioning and testing that validity of these patterns. One of the beautiful aspects of live sound is that there is no true right or wrong way, but rather, certain approaches are more likely to result in preferable outcomes than others.
With that in mind, let’s focus on the process we most commonly call “sound check.”
Why EQ the kick drum by itself with all the other microphones turned off? How often during the actual show do you mute every other mic to just hear that kick drum sound? How relevant and useful is it to waste oh-so-valuable sound check time EQ’ing solitary mics only to start over once the rest of the stage mic interactions are introduced?
Of course I understand doing a quick test of every mic individually, but beyond that, what we really need to know is how that instrument sounds with all the other mics turned on as well. Seems we forget that every mic hears everything on stage at some level.
Want way more time to really get your sound dialed in and have the band love you at the same time? At the next gig, walk in and tell the band, “O.K., this is how I would like to sound check. After a quick tap line check to make sure everything works, you guys come on up and do whatever you want, rock some tunes, rehearse and jam.
“First we’ll get monitors sorted and close. To avoid confusion, here is a simple hand signal method, point at what you want and then point at where you want it and then point up or down so we know what to do. And while you’re rocking out, I’ll get all your sounds dialed in out front. I may stop you for a moment if there’s a particular problem, but what’s best for me is for you to play as many tunes as possible and get comfortable on this stage.
“Oh, and drummer person, if you can, lean into some extra toms so I can grab them as well.”
Congratulations—you’ve just gone from having your band annoyed with being subjected to 50 hits on each drum to having happy musicians doing what they (hopefully) truly love.
With the artists playing, bring up each instrument and get a rough EQ and meanwhile, you also learn important things like how much the cymbals bleed into the toms, or how much guitar is getting into the vocal mics while you are EQ’ing them.
As the band kicks out the jams, my approach is to bring up drums one at a time and do a rough EQ, then all the drums and refine the EQ. Add bass, check the drums and bass combo, then EQ the bass.
Next, lay guitars on top, get a rough EQ, and touch up bass and drums. Then give a listen to just guitar and bass without drums, and EQ them to fit.
All the while, I’m dialing in my compressors and gates. I finish with muting all the other mics and EQ’ing vocals with the full band playing. Then vocals, add in guitars, add bass, and then add drums. My headphones are always at the ready for cuing up and checking certain things.
If specific issues come up while the band is playing, hey, don’t worry about it yet. Get the rest of the sounds together first. The goal is obtaining a solid grasp of the bigger picture in the time it takes to test one mic at a time.
Plus you’re actually mixing, and are free to make drastic changes to hear those blends and combinations in a way that you can never do during an actual show. You’re also now the coolest engineer the band’s ever worked with.
Back In The Day
I started using a version of this approach about 25 years ago with a 60-piece orchestra I mixed weekly. Due to wind and the size of the area being covered, there was the need to rely on fairly close mic’ing, and ended up with about 24 inputs.
It dawned on me pretty quickly that the whole “O.K., now will the third flute please play” method was a complete waste of time and left me scrambling to try to scrape a mix together when the show began.
So I devised a plan. Turn every gain knob all the way up, and have every channel muted with the fader down. As the various orchestra members showed up, tuned their instruments, and began playing, the clip light for the mic(s) near them would come on.
I would crank the gain down to below clip, PFL that channel to make sure it sounded fine, and then un-mute so I knew which channels had the gains set. As the channels were un-muted, I brought those faders up and began blending and EQ’ing, while waiting for the next clip light.
Finally, the conductor would have the orchestra play a short segment, and that was that. The whole process took about 15 to 20 minutes, I had the mix together, and then had time to go to the stage to fix any issues before the show started.
Dave Rat heads up Rat Sound Systems Inc., based in Southern California, and has also been a mix engineer for more than 30 years.
CADAC CDC four Digital Console Fronts Large-Scale System For Celebration At Iconic Ibiza Club
Console heads up large-scale system incorporating Funktion-One loudspeakers powered by Full Fat Audio amps with XTA processing
Iconic Ibiza club Space recently celebrated its 25th anniversary with a birthday bash centered on its outdoor Flight Club arena, with regular Ibiza sound specialists Project Audio providing a CADAC CDC four compact digital console in front of a large-scale system incorporating Funktion-One loudspeakers powered by Full Fat Audio amps with XTA processing.
The club’s earlier 2014 season “Opening Fiesta” in late May saw the Funktion-One rig fronted with a CADAC LIVE1 analog console. Ibiza, noted for its legendary nightlife, is the third largest of the Balearic Islands in the Mediterranean Sea, 50 miles off the coast of the city of Valencia, in eastern Spain.
Dave Millard, founder of Full Fat Audio, was Project Audio’s sound engineer for both events at Space, working in partnership with Funktion-One chief Tony Andrews and Project Audio’s Ibiza system technician George Yankov.
“We used the LIVE1 on the opening party, but for the 25th Anniversary we needed to wireless mic a troupe of flamenco dancers on stage and use some effects on them, so we went with the CDC four for that,” says Yankov. “It was a real pleasure to use both the analog LIVE1 and digital CDC four. The audio performance of both consoles is equally excellent. I cannot recall another desk so transparent and with so much drive and finesse.
“The CDC four really allows the audio to breath and just does not sound digital at all,” he continues. “Every nuance of a recording or live input can be heard, with even subtle changes to the controls. Bass performance is exciting with every note precise. Build quality is also first class and user interaction is straightforward.”
The 25th anniversary party featured a line-up of Playa d’en Bossa regulars and legends, including Nina Kraviz, Carl Craig, Jimmy Edgar, Shaun Reeves, Layo and Bushwacka!, Alfredo, José Padilla, Jose De Divina and César De Melero, a four-hour set from Erick Morillo, and ‘cameo’ spots from Fatboy Slim and Annie Mac.
Synchro Arts Introduces Revoice Pro 2.6, Offers Numerous Enhancements (Includes Video)
Workflow speed-ups and enhancements to Cubase and Nuendo users and speed improvements for Pro Tools 11 users
Synchro Arts has announced Revoice Pro 2.6 for both Mac and Windows, offering significant workflow speed-ups and enhancements to Cubase and Nuendo users and speed improvements for Pro Tools 11 users.
New features in Revoice Pro 2.6 include the ability to instantly import audio and clip information from Cubase and Nuendo to Revoice Pro using either Copy and Paste or Drag and Drop. Instant transfer of Revoice Pro’s processed audio back via Drag and Drop or export audio function.
Cubase 5.0 and later and Nuendo 5.0 and later—including Nuendo 6.5—are supported. For Pro Tools 11 users of Revoice Pro on Windows, the slowdown in Pro Tools when the Revoice Pro plug-in is open have been removed.
Revoice Pro has been designed for professional audio editors and provides easy-to-use tools for manipulating audio features (timing, pitch, vibrato, inflection and level) with precision and quality. There’s also unique automation of tedious audio feature editing tasks, which can save hours and improve results.
Overview of Revoice Pro
Revoice Pro is a purpose-built, stand-alone program that includes two unique, automated processes:
1) Audio Performance Transfer (APT) process. Automatically transfers selected timing, pitch, vibrato, inflection and/or loudness characteristics of a good “guide” audio signal to one or more target audio signals (“dub”). APT is powered by and includes an advanced version of VocALign.
2) Doubler process. When double tracks haven’t been recorded, Revoice Pro’s Doubler creates incredibly natural-sounding ones. And when you a manual adjustment of timing or pitch is needed, there are numerous simple-to-use tools available.
Applications include tightening the timing, pitch and vibrato of “stacked” lead and backing vocals or instrumental tracks, as well as creating one or more realistic double tracks from a single input track. In addition, users can lip-sync dialog (ADR) and vocals by the same or different performers, even when there are noisy guide tracks. In addition, the inflection in dialog (ADR, voice-overs etc.) can be changed with the desired Guide pattern provided by recording the director or dialog editor.
“Revoice Pro saves me and my team hours of work when it comes to vocal pitch and timing correction with doubled vocals and backgrounds. “It’s amazing how fast the workflow is,” states Tony Maserati, Grammy winning producer (Lady Gaga, Jason Miraz).
“I started using Revoice Pro on American Hustle and I was surprised at how quickly and perfectly it matched sync without sonic artifacts,” adds Renée Tondelli, dialogue and ADR editor (American Hustle, Django Unchained). “I now use the pitch function to match performances, and it works incredibly well. Because Revoice Pro is so fast and precise, it’s now my go-to tool.”
A 14-day free trial license (iLok-based) for Revoice Pro can be obtained from www.synchroarts.com along with downloads of the Revoice Pro program, online manuals, demos and tutorial videos. Full licenses (iLok-based) can be purchased from Synchro Arts’ dealers or on-line from www.synchroarts.com/store.
Recommended retail price of Revoice Pro is $599 for North America, £374 (ex VAT) for UK and the rest of the world, and €449 (ex VAT) for Europe. Discounts are available on trade-ins for current VocALign owners.
Friday, August 15, 2014
Emulation Destination: Plug-Ins For Enhancing Live Applications
Ask live mix engineers their favorite effects and processors, and you’ll get dozens of different answers.
Some still prefer outboard gear, ranging from the more common to the esoteric. Vintage purists may want older tube gear that’s no longer even manufactured, while others in this camp aren’t satisfied with anything less than a single particular unit that was only used (and possibly built into) a specific recording studio.
In the past it was difficult, if not impossible, for a production company or venue to assemble a rack of outboard gear that would satisfy every engineer. It was also hard for concert tours, particularly those comprised of one-off fly dates, to carry all of the outboard pieces that they wanted, or more importantly, that the artist that they worked for needed.
Those days are largely in the past, as consoles with onboard digital processing and software plug-ins have taken over effects duties. Do you prefer the onboard effects and processing in your digital console? No problem, just patch them in where want.
Need a particular effect or processor that’s not available onboard? Again, no problem as you can probably use a plug-in directly with the console or augment it with a full server-type plug-in platform such as Waves SoundGrid or the Soundcraft Realtime Rack. Further, Avid consoles come with a collection of plug-ins, with VENUE consoles using the TDM VENUE format and the more recent S3L console accommodating the growing AAX format.
In a recent article (here), I provided an overview of the various types of plug-in formats as well as the platforms that support them, along with some general applications. I’m continuing the discussion here with a look at specific plug-ins that I’m using and/or that have peaked my interest.
For most bands, I keep things relatively simple from an effects standpoint. I begin by patching in a quality vocal reverb, an adjustable delay for vocals, and a solid snare drum reverb.
While the onboard effects and verbs of most consoles range from pretty good to great (and a few are quite stellar), there are times when I want something more specific to suit the needs of a particular performer or to better duplicate the recorded sound in the live realm. And for me, this is where plug-ins can come into play. Here are a few that have caught my eye.
Studio To The Road
An effect that has been limited to studios is double tracking, where a main track is duplicated on a second track, lending a fuller sound—and many times, with the two tracks being panned left and right, providing a stereo signal from a mono instrument.
We can sort of emulate double tracking live by applying a very short delay to a signal and feeding both the original and delay signals to the PA. This gives a “thicker” sound especially useful on guitars and vocals.
However, Waves recently introduced the Abbey Road Reel ADT, designed to emulate Abbey Road Studios’ process of Artificial Double Tracking, a signature effect created at the studio in the 1960s for The Beatles. The Reel ADT was developed in association with Abbey Road and the process has been a closely guarded secret of the facility until recently.
The original process was created by Abbey Road engineer Ken Townsend, who connected a primary tape deck to a second speed-controlled tape deck allowing two versions of the same signal to be played back simultaneously. By varying the speed of the second machine, the replayed signal could be moved around to simulate a separate take.
A retro look to go with the classic sounds supplied by the Waves Abbey Road Reel ADT.)
The Abbey Road Reel ADT can even emulate the sound of tape complete with wow and flutter effects, providing the closest thing yet to real double tracking. Where was this a few years ago when I mixed a Beatles tribute band?
Lexicon has long supplied go-to reverbs, with the PCM96 stereo verb and effects processor one of the most popular. It has 28 legacy and new reverbs, delays and modulation effects that can be integrated into both digital audio workstations (DAW) as well as live rigs.
While the 1RU package is compact, it still takes up space and adds weight, while its price may not be in the budget. Plug-in format to the rescue, with Lexicon offering the PCM Bundle that utilizes the same algorithms and presets from the PCM96 hardware unit at about a quarter of the price.
A Vintage Plate from the Lexicon PCM Bundle.
A really cost-effective direction for getting the vintage Lexicon sound is the Native Instruments Reverb Classics plug-in that provides emulation of some reverbs based on the classic Lexicon 224 and the 480L, which are still used in many top studios. Although the number of room and hall choices compared to the PCM Bundle is limited, the Native Instruments Reverb Classics could be just the ticket for a vocalist who is singing classic hits.
Another classic tool that’s hard to drag out on the road is a real plate reverb.
These large units operate by inducing vibrations into a large plate of sheet metal with an electromechanical transducer and then using a pickup (or two for stereo) to capture the vibrations.
Elektro-Mess-Technik (EMT) introduced the EMT 140 in the late 1950s, and it became a popular model featured on many hit records.
But instead of lugging around a 600-pound plate reverb to gigs (and ticking off the stagehands), you can load up the EMT 140 Classic Plate Reverberator plug-in from Universal Audio. It replicates the sonic signatures of three different EMT 140s that are installed at The Plant Studios in Sausalito, CA.
Compressors are sometimes tricky, with each different model subjectively better or worse than others depending on the source material and the expected results. Most engineers have their favorites for use on vocals, bass, drums and percussion, and I’m no exception. Some of these classic hardware units are still popular in the studio today, but aren’t too practical for live use due to cost and durability factors.
One solution stepping up to meet these need from a plug-in standpoint are Waves CLA Classic Compressors bundles, providing numerous emulations created with the aid of multiple Grammy Award winning mixer and producer Chris Lord-Alge. They offer takes on the classic Teletronix LA2A and Urei 1176 units that I love for giving a “classic squeeze” on both live vocals and instruments.
The Softube Summit Audio TLA-100A plug-in sounds and looks like the original hardware.
Softube has teamed up with hardware manufacturer Summit Audio to develop the Softube Summit Audio TLA-100A, based on the Summit Audio Tube Leveling Amplifier, another staple in many studios.
This plug-in adds a saturation control, allowing the user to pump up the character of the processing without overdriving the input that was required on the hardware version to get the same effect. It should find many uses onstage, especially with acoustic guitars and vocals.
Need more choices? Look no further than the new McDSP 6030 Ultimate Compressor, an AAX plug-in offering 10 different compressors in a single interface. Some of the units are emulations of existing gear with unique variations by McDSP and other units were designed from the ground, providing a wide range of processing and dynamic options in one package.
The McDSP Ultimate Compressor provides 10 options in a single package.
For those who are doing live broadcast remotes or needing to tame the feeds to live webcasts and video recording, the Waves MaxxVolume may be just the ticket.
It combines the technologies from the L2 Ultramaximizer, C1 Parametric Compander, Renaissance Vox, and Renaissance Compressor, offering up a slew of versatile processing. From leveling dialogue at the podium to helping control acoustic instruments in the mix, MaxxVolume is handy indeed.
Senior contributing editor Craig Leerman is the owner of Tech Works, a production company based in Las Vegas.
Thursday, August 14, 2014
AMX Integrates Next Generation NX Series Control Processor Into Enova DGX Digital Switcher Line
Enova DGX models now have the fast processing power, security features and scalability of the new NX Series, with no change in pricing
Harman’s AMX announced that all models of Enova DGX digital media switchers are now equipped with the new NetLinx NX Series central control processor. Enova DGX models now have the fast processing power, security features and scalability of the new NX Series, with no change in pricing.
DGX models (DGX 8, 16, and 32) incorporating the new NX control processors began shipping this month, while NX Series controllers have been shipping in the Enova DGX 64 chassis since its release in April 2014.
The new DGX models come with a long list of immediate enhancements, including a high performance architecture that includes a 1600 MIPS processor that can manage the most complex and expansive applications. And with the NX Series under the hood, the Enova DGX supports wired 802.1x for enhanced security authentication and IPv6 for the most modern IP addressing.
Migrating to the new DGX models is simple, as they are fully compatible with code developed for the previous AMX NI Series control processors. Should they wish, existing DGX customers will have the opportunity to upgrade their current DGX switchers with NI Series controllers to the new processor by purchasing a CPU Replacement Kit. Concurrent with this upgrade, AMX is implementing a simplified model numbering scheme for the DGX family to streamline the ordering process.
The Enova DGX functions as a centerpiece of a complete integrated solution that manages and distributes analog and digital audio and video, including HDMI/HDCP, control and Ethernet. A comprehensive set of Enova DGX hot swappable boards can be used in conjunction with DXLink transmitters and receivers to provide an end-to-end distribution system over twisted pair cable or fiber.
Seattle-Based PNTA Joins Yamaha Commercial Audio Dealer Network
Company adds new Yamaha QL5 digital console to its growing inventory
Seattle-based PNTA has joined the Yamaha Commercial Audio dealer network, recently adding the new Yamaha QL5 digital console to its growing inventory.
The company opened its doors in 1975 as a local Seattle-area supplier of theatrical equipment and consumables, and since that time, has expanded staff and capabilities to provide a wealth of services for customers nationally. Employee-owned since 2008, PNTA has increased its services to include audio, video, and lighting.
With an experienced technical services staff, PNTA installs and repairs equipment and supports both non-profit and private customers through its event services department, providing production requirements for events of all sizes. The company is located in a 20,000 square-foot facility near downtown Seattle, offering a sizeable brick and mortar retail store supporting its dealer activity for entertainment and live production manufacturers, now including Yamaha Commercial Audio products.
“We did an exhaustive internal review of what audio vendor we needed to strategically add to best accelerate our market progress in growing our overall audio business,” says Dave Vaught, Event Services production manager at PNTA. “We were particularly interested in broadening our offering in professional products appealing to our client base. We focused those efforts initially in audio control and sound reproduction as we needed more tools for sale, rental, and event services support. With the substantial market position and reputation of Yamaha and its many new and highly regarded products recently introduced, the answer became clear. We are happy to have become a Yamaha dealer and integrator of Yamaha’s Commercial Audio Systems products.”
Vaught said that PNTA wanted to add consoles that are expected to see high rental demand from both existing customers as well as those traveling into the state of Washington.
“With the announcement of the QL Series, we found a very forward-thinking technological capability in a footprint that was easily transportable while being exceedingly powerful,” he adds. “The expandability of the QL Series clinched our choice and buying depth at key levels. Based on our client history, every need we could think of could be answered by some or all of the features of Yamaha QL desks. We were impressed with a number of the features, including onboard recording, signal and effect processing, auto-mixing, and iPad integration.”
For rentals and event services, PNTA also invested in a loudspeaker line extension, moving into the Yamaha DXR and DXS Series.
“While our company size has grown along with the expanded services we now offer, PNTA retains what we started with when we first opened our doors—a dedication to meeting the needs of our valued customers and finding creative solutions to tough problems,” states Richard Carlson, president of PNTA. “By adding Yamaha products to our product line, we can now offer the ultimate in quality and reliability to our audio customers. Yamaha consoles are specified on many riders and fill rental needs as well as being versatile for our production needs.”
Yamaha Commercial Audio
Wednesday, August 13, 2014
Solid State Logic To Host Free SSL Live Console Training At UK Headquarters
Training sessions will include hands-on experience for all participants
Solid State Logic (SSL) has announced Live Console Operator Training Days, a series of free one-day courses that will include hands-on experience for all participants, aimed at serving as an introduction to the SSL Live console.
Training sessions will take place at the company’s Oxford, England-based headquarters this coming September 17 and 18, October 1 and 2, and November 5 and 6.
While the sessions are free, numbers are strictly limited, so all interested parties are encouraged to apply via the online application form (here).
SSL Live is the company’s first console for live sound production, suited to touring or installation, and for front of house or monitor systems for venues, arenas, houses of worship and concert halls.
Solid State Logic (SSL)
In The Studio: The Five Levels Of Mixing Quality
The meaning of the word “good” is one of my odd and recurring fascinations. What makes a “good” mix?
Music is a subjective field with many general principles but very few hard and fast rules.
The arts are inherently up for interpretation — and so “good” to one person may be “bad” to another. I could site examples of this, except there’s so many I feel there’s really no need.
So I often repose the question to myself: what makes a “good” mix? After all, that’s what I get paid for right? To make “good”/”great”/”unfrickin’ real” mixes.
Keeping in true-to-blog format, here’s a list of what I feel makes for the levels of “goodness” in a mix.
Level 1: Getting The Sound “Out of the Way”
At the most fundamental level, recordings are ultimately adulterated forms of a musical performance.
The fact is nothing really equates to the sound in the room, and when we start putting microphones in between the performance and the two measly speakers that are attempting to regurgitate that performance, it’s going to fall flat.
Couple that with any deficiencies of the recording space, equipment, or (hey, hey) tracking engineer — or lack thereof — and we soon find that the record pales in comparison.
Level 1 is the recognition that mixing is a necessary evil — someone has to compensate for all of this and “get the sound out of the way.”
Because it’s really hard to enjoy a performance when the guitar sounds like it’s under a blanket and the vocal sounds like the singer was chewing on the microphone in a space that sounds like a space-cavern and coffin at the same time.
No matter how good the performance is, bad sound is going to interfere with the listener’s experience.
Level 2: Making The Sound “Larger Than Life”
Once the sound is out of the way the most important job is done. But now we get to view mixing as a creative medium.
While a recording will never have the power and impact of a live performance, the actual sound performance can do things that can’t happen in nature. And that’s a fantastic thing.
Records, like film, have grown into their own art because of the manipulation that can occur. We can create a sound that is “larger than life”.
This comes in many forms: giving the sound a greater space, elements that are more vivid than we would hear them even in the best of sound systems, shaping sounds to have a stronger perceived impact than they normally would, etc.
Level 3: Enhancing The Musicality
This is the level that separates the aspiring engineers from the inspired engineers.
The ability to “help the music along” is often lost on the bands and artists who need it the most — but for the vetted artists, bands, producers who hear and feel musicality — this is the real litmus test.
The engineer hears the basic mix and begins to interpret the musical intentions.
There’s a myriad of means in which musicality is expressed, interpreted, and subsequently helped along — and a great deal of it is instinct — but when you hear it you hear it.
All I can say is a great deal of this process involves automation — bringing key elements out at exacting moments.
Level 4: Understanding The “Bigger Picture”
Music does not exist in a box.
Having an appreciation for the culture of people creating and listening to that music is paramount.
This doesn’t mean strictly playing to the aesthetic of the audience, but also knowing how to manipulate their expectations. This means not only understanding what the listener wants, but also why, and what the effects of altering their expectations may be.
Level 5: Doing Everything To Serve The Song
The mixer’s role is generally understood to be the tailoring of elements within the production.
However, the mixing phase is still a production phase, and as such, there is still time for adding, removing, or changing the vision of elements.
I have done everything from adding crazy effects, muting instruments, replacing drums, overdubbing guitars, and even added vocals onto records. The cornerstone to all of this is doing so in good taste.
The other important consideration is that the mixer sometimes must sacrifice their own importance. The things which “feel” the best aren’t necessarily the things that “sound” the best.
Putting things out of balance, leaving them muddy or thin, overly reverberant or awkwardly dry, can all go towards the main goal: the success of the song. The most successful songs are the ones that are the most compelling to the listener and that doesn’t always mean perfection.
There are two points I’d like to make about this article.
First, all five of these “levels” correlate. They’re not in fact separate stages or concepts, but more like degrees of mastery.
To this day I am still refining my skills in levels 1 & 2, even though my main goals are mastery of levels 3, 4 & 5.
My second point is that I didn’t choose these levels randomly. I put them in order of primary importance and difficulty of mastery.
The vast majority of mixes I hear do not have proper negotiation of levels 1 & 2 — far be it from 3, 4, or 5.
It takes a great deal of study, practice, and discipline master the art of mixing, so don’t ever be afraid to revisit the foundation during your journey!
Matthew Weiss engineers from his private facility in Philadelphia, PA. A list of clients and credits are available at Weiss-Sound.com. To get a taste of The Maio Collection, the debut drum library from Matthew, check out The Maio Sampler Pack by entering your email here and pressing “Download.”
Also be sure to visit The Pro Audio Files for more great recording content. To comment or ask questions about this article, go here.
Church Sound: Mistakes Worship Teams Make That Can Compromise Services
The mistakes worship teams commit while approaching God do not preclude His presence, but they do erect obstacles to the flow of the Holy Spirit.
Here, then, are 10 common errors churches can avoid in the pursuit of God:
1. Turning minor mistakes into public spectacles. When a vocalist forgets to turn on a wireless microphone or a technician commits a track cueing error, the worst thing a worship leader can do is to proclaim the mistake to the entire congregation.
If the audience didn’t notice, why bring it up? And, if it was an obvious error, then everyone already knows about it.
Far from being a way to humanize the proceedings, public notifications only hinder the work of the Spirit and demoralize the person responsible. It’s better to go on with the service and discuss the incident in the context of love at a later debriefing.
2. Playing too much. Some musicians live to play and feel compelled to use every chord they know each service.
Just as too many cooks spoil the broth, so does too many notes spoil the song. If each segment can be given some air to breathe in the form of silence around the song, then each part that is played takes on added value and weight.
Ed Kerr says it best, “Make every note you play count toward the goal of communication and away from a focus on your ability.”
3. Playing too loudly. Worship “wars” are known for their resounding barrage of noise.
The goal of the band should not be to destroy the congregation’s hearing, but to play music that encourages the audience to participate in a journey to the throne of God. How loud is too loud is a question each team must answer based on the culture and circumstance of the local assembly.
However, a rule of thumb is to keep the stage level low enough that unamplified voices can be at least partially understood from a one-foot distance. The house mix level should be below 95 dB-A average response.
4. Choosing inappropriate material. I recently attended a worship service designed for 40-year-olds that incorporated a musical style more appropriate for 20-year-olds. While the audience seemed to appreciate the band’s efforts, they never became engaged in the proceedings. There were, though, a few “Gen Xers” in another room who were drawn to the sounds emanating from the sanctuary.
As a church consultant, I’ve been asked to referee many battles between the old and new, and have discovered the new is more readily digested when coated with cues from the old. No one wants to be outmoded, and there will always be someone who lives to hear Journey-esque music performed by a Steve Perry wannabee.
Keeping everyone happy is one way to direct people to Christ.
5. Selecting songs average people can’t sing. In a recent informal survey of non-participatory church goers, the majority cited the frustration they feel when they desire to worship in song, but are hindered by a musical selection beyond their range.
While the team members may impress themselves with their virtuosity and skill, the average Joe in the pew just gives up and stares into space.
Engaging people is never accomplished by making them feel inferior and inadequate.
In the words of Chariya Bissonette, “It doesn’t matter what you [the vocalist] can do. It only matters what Christ can do through you.”
6. Starting the service late. If the service is to begin at 10 am, then that’s actually when it should start - lest those who made the effort to be there promptly are disenfranchised while those who failed to arrive early are greeted with an “it doesn’t really matter mentality.”
One of the most precious commodities people have is time, and starting a service late implies their time gift is not important to the staff and team.
7. Treating rehearsal time as practice time. As Jamie Harvill states, “Rehearsal is crafted to polish the song, not to learn it. Individual practice time is when learning occurs.” Curt Coffield uses the time/money scale to weigh the value of rehearsal. If each member’s time is worth $25 per hour, imagine the total value of every rehearsal event and treat it appropriately.
8. Buying a Hyundai, then driving it like a Ferrari. Audio and video systems cost what they are worth. There is no way a modest system can perform like an expensive, properly designed system.
Churches love to set system budgets, and then try to force the integrator to “make it work.” Unfortunately, God’s laws of physics apply in His house just like they do at an Eminem concert.
As the cliché says, you get what you pay for. If a church needs to reproduce video and audio at a high level, it takes the right equipment and personnel to achieve the goal.
9. Presenting a hip image of Christianity in place of the image of Christ. God does not call us to make Christianity cool. There is nothing cool about suffocating to death on a cross while stripped naked.
The Gospel is a wonderful message and conveys hope, but not at the expense of truth. Our message must be applicable to all people for all time in all circumstance.
10. Creating virtual music. Performing “Muzak” versions of rock tunes with guitars played through modeling modules and drums banged out on electronics drums does not endear the message to someone raised on real rock ‘n’ roll.
If the situation is appropriate for virtual instruments and the room acoustics are atrocious, then virtual may be the answer. However, if authenticity is the goal, then authentic instrumentation is the means for success.
Discernment is needed to understand when to wail and when to use in-ears.
Kent Morris is noted for his church sound training abilities. He has more than 30 years of experience with A/V, has served as a front-of-house engineer for several noted performers and is a product development consultant for several leading audio manufacturers.
Tuesday, August 12, 2014
In The Studio: Five Steps To Checking Your Drum Phase When Mixing
One of the most important yet overlooked parts of a drum mix is checking the phase of the drums. This is because not only will an out-of-phase channel suck the low end out of the mix, but it will get more difficult to fix as the mix progresses.
I covered how to check the polarity of the drum mics a few weeks ago (here), but here’s an excerpt from my Audio Recording Basic Training book that covers a way to check the phase when you’re setting up for a mix as well.
A drum microphone can be out of phase due to a mis-wired cable or poor mic placement. Either way, it’s best to fix it now before the mix goes any further.
1) With all the drums in the mix, go to the kick drum channel and change the selection of the polarity or phase control. Is there more low end or less? Chose the selection with the most bottom end.
2) Go to the snare drum channel and change the selection of the polarity or phase control. Is there more low end or less? Chose the selection with the most bottom end.
3) Go to each tom mic channel and change the selection of the polarity or phase control. Is there more low end or less? Chose the selection with the most bottom end.
4) Go to each cymbal mic or overhead mic and change the selection of the polarity or phase control. Is there more low end or less? Chose the selection with the most bottom end.
5) Go to each room mic channels and change the selection of the polarity or phase control. Is there more low end or less? Chose the selection with the most bottom end.
You’d be surprised how many times that flipping the phase on one or two of the drum mic channels results in a better, fuller sounding kit, even on one that’s well-recorded.
Bobby Owsinski is an author, producer, music industry veteran and technical consultant who has written numerous books covering all aspects of audio recording. For more information be sure to check out his website and blog. And go here for more info and to acquire a copy of Audio Recording Basic Training.