Friday, May 06, 2016
MULANN Launches Recording The Masters Analog Tape Brand
New brand of professional recording, mixing, mastering and archiving media expands market in the United States, Europe and other territories.
MULANN Group announces “Recording The Masters” a new brand and identity for its AUDIO professional activities. Creating a new visual identity for the audio products with a new logo reflects MULANN’s commitment to professional audio recording and strengthens its presence on the worldwide music, archive and instrumentation markets.
“MULANN owns the original formulas of analog recording, some of which date back to 1950, created by AGFA and BASF. These magnetic formulas deliver a very high sensitivity and dynamic sound quality. They also offer the capabilities to store data for several decades, far beyond what digital and optical media offer today,“says Jean-Luc Renou, MULANN CEO.
Oriented to professional and semi-professional recording, mixing, mastering and archiving, MULANN is expanding its position in this market in the United States, Europe and other territories.
With the introduction of this new brand, MULANN products speak directly to both the professional sector and to audiophiles. The analog recording tapes created by AGFA, BASF then EMTEC, are now manufactured by MULANN group under the brand “Recording The Masters”. This new visual identity energizes the market searching for sound authenticity, sustainability and the original technical qualities of audio recording developed in the mid-twentieth century and never equaled over the years.
“The visual brand identity “Recording The Masters” is radically new,” explains Renou, “This new brand identity is dynamic, modern and it indicates that analog recording is active more than ever. Its technology values offer a future and an important role to capture, transmit and store audio tracks and sounds for the long term. The analog recording flame shines even brighter for an everlasting light.”
Posted by House Editor on 05/06 at 10:46 AM
Wednesday, May 04, 2016
Leona Lewis Tours UK With Solid State Logic
Front of house engineer Dave Wooster selects the SSL L500 console for 14-date tour.
Soulful, multi-talented vocalist and performer Leona Lewis has been on a 14-date tour of the UK with a Solid State Logic L500 console and K-array Slim Array technology, chosen by front of house engineer Dave Wooster.
Lewis’ I Am show provides Wooster with around 65 inputs from the stage. “It’s a full-on set-up,” he says. “The drummer alone has two mics on the bass drum, three snares - each with at least two mics on, four toms, all the cymbals, electronic kick and snare… So he was up to about 20-odd channels on his own.”
According to Wooster, the show absolutely benefited from the L500’s signal path - from the SuperAnalogue mic inputs, through the flexible channel path, and comprehensive internal FX Rack: “What really separates the L500 from the competition is the sound.”
“The effect on Leona’s vocal was very noticeable in the system,” he continues. “I think the 96kHz operation makes a difference, but the pre-amps make a huge difference as well, and whatever it is SSL has done on the EQ is stunning…. You really hear the HF.
“With Leona I have to deal with a massive dynamic range within every song… The mix has to be able to go right down to almost nothing and then build to everything. The way the SSL input section handles that is fantastic.
“Of course, it’s natural that when she whispers I get a load of low end from the microphone that I don’t need, and when she’s screaming down it there’s too much high end and not enough lows. I use a dynamic EQ from the internal FX rack to sort that out. The standard EQ helps calm down some resonances, though there were only two cuts with low and high pass filters that I needed to make with that.
“The channel compressor is the first layer of dynamics control, just to help take out any real big peaks; then across her stem- which includes her reverb and delay returns, as well the main vocal path - I put an SSL Bus Compressor; it’s very good.”
Wooster’s approach to the console surface configuration takes full advantage of the L500’s layer & bank approach to layout, as well as the Super-Query (‘Q’) function - a forward and reverse interrogation feature with fast-assignment feature.
“I have all my input channels as a sub-layer,” says Wooster. “That’s where all the programming is. Then I use Stems on the top layer. I completely isolate them from any recall and end up with kick, snare, hat, toms, overheads, a bass channel, guitar channel, keyboard channel, lead vocal and BV stem faders that are always below my fingers… That’s my mix.
“All the automation and scene recalls are still going on underneath, so if I then hit the Q button on any Stem all of the underlying contributions pop up from below. I can make a quick adjustment in that scene, save it, then go straight back to the Stem layer and carry on mixing.”
As well as the console, the tour rig included the innovative K-array ‘Firenze’ Slim-Array PA system with acoustic steering. Wooster is convinced that the combination of the two was unbeatable: “In the 33 years I’ve been doing this,” he says, “I’ve never mixed on a system this good…”
Front of house engineer Dave Wooster at the SSL L500
Solid State Logic
Kingston College Outfits Creative Industries Centre With Audient
UK school selects ASP4816 and ASP8024 analog consoles to compliment additional Audient gear in new facility.
Boasting two brand new studios, fully kitted out with two Audient analog mixing consoles, two iD22 audio interfaces and three 8-channel mic pre ASP880s, Kingston College’s Creative Industries Centre was opened at the beginning of this academic year.
As they advance into their second term, Music Technology instructor and studio technician, Chris Winter has noticed how the students have been enjoying the recent studio upgrade.
“They love using both the ASP4816 and progressing to the ASP8024 with Dual Layer Control,” he says, pointing out that they find them “easy to understand and clearly laid out. They love that analog sound, too.” Whilst the desks are located in Studios 1 & 2 - the commercial and the primary teaching studio respectively - the outboard gear is predominantly used in the four production rooms that are linked to the rehearsal and live rooms.
“The students love the way iD22 allows you to assign a mic pre as talkback mic. The ASP880s are also really useful when you want to DI an instrument quickly.” He continues with his own thoughts on the new Audient outboard gear: “The iD22s are so easy to use and have great, big console features. The ASP880 are amazing too, and really work well with the iD22s. The idea was that we have the same mic preamps across the facilities so there is continuity in the sound. And we love them.”
Winter describes the new building that houses the studios as “…an amazing space. We are very lucky to have the opportunity to build industry standard studios to such a high specification, and work with great companies like The Studio People on getting wonderful sounding rooms that look fantastic too.” Comprising a 3D workshop, TV studio, as well as a mixture of studios and classrooms designed to enhance the learning experience, the Creative Industries Centre is designed to encourage cross collaboration on projects with other courses taught at Kingston College. “The students love the studios’ design with all that natural light; it has a big impact on their learning and creative flair,” says Winter.
“We teach from Level 2 up to level 5 courses in Music Technology and Performance,” he explains. “The great thing about these Audient desks is that for beginners right up to advanced users, they are easy to understand with no gimmicky features.” Winter is very clear that teaching signal path to students is “…fundamental to their training. So many people have tried to teach ‘in the box’, but have reverted to analogue board as a physical piece of equipment that you can touch and see where the signal starts and ends up.”
Indeed all the new technology in the Creative Industries School helps with that. “Audient has high quality products that are easy to use and are at a professional level. We wanted our students and users to learn on industry standard equipment, so they can transfer their skills learnt at Kingston College straight through to employment.”
Audient wishes staff and students all the best with the new facility.
Posted by House Editor on 05/04 at 09:48 AM
Meter Madness: What Your Level Meters Tell You And What They Don’t
In May 2014, the VU meter celebrated its 75th birthday. It has served the industry well, and when properly interpreted, it’s still useful.
However, today’s digital recording processes have caused us to take a hard look at the usefulness and inadequacy of both the traditional VU meter and its modern replacement, the LED level meter, as tools for signal-level management.
The classic VU meter, though relatively rare today (primarily because of cost), has long been the most common audio-level indicator. VU stands for Volume Unit, and the VU meter indicates how loud something sounds.
The traditional VU meter is mechanical, analog, and has a standardized (even the color scheme is standard), logarithmic (decibel) scale that runs from –20 to +3, with usable resolution over about a 15 dB range. The zero mark is about two-thirds of the way up the scale. This 0 VU mark is what we mean when we say the meter “reads zero,” not where the pointer rests when the equipment is turned off.
Resolution is very good near the high end of the scale: generally 1 division per dB between 0 and +3. The scale resolution becomes progressively lower (more dBs per scale division) below zero VU.
These days, the mechanical meter’s replacement is usually a column of LEDs, with individual lights serving as the meter scale ticks. On a computer-based digital audio workstation, a meter is often simply drawn on the monitor screen. The Cool Look Factor is lower, but it’s a lot cheaper than using mechanical meters. However, some analog audio devices—notably “boutique-style” preamps and channel strips such as the PreSonus ADL 600, ADL 700, and RC 500—still sport traditional VU meters.
In this article, we’ll review the characteristics of the traditional analog VU meter and also address audio-level measurement and management in the digital domain, the concept of headroom, and how loudness and audio level are related—as well as how they’re not related.
What’s a VU?
The Volume Unit meter was originally designed to help broadcast engineers keep the overall program level consistent between speech and music. There’s a well-defined standard for the mechanical rise-and-fall response characteristics of the pointer (to which few of today’s VU-like meters actually comply).
A standard VU meter responds a little too fast to accurately represent musical loudness but fast enough to show movement between spoken syllables. Unscientific as it may seem, the dynamic response of the VU meter was tailored so that the pointer motion looks good when indicating speech level. It was easy to tell at a glance whether speech or music was going out over the air, whether it was about at the right level, and when something wasn’t working. Engineers learned that brief excursions up to the +3 dB top end of the scale rarely caused distortion in the analog equipment of the day, nor did they sound too loud.
How is Audio Level Measured?
Once sound is converted to electricity, we can represent the audio level by measuring the alternating electrical voltage. A symmetrical signal, such as a sine wave—a single-pitched note with no overtones or distortion—spends equal time on the positive and negative sides of 0 volts.
The numerical average of the positive and negative voltages over many cycles is zero—not a very useful measurement when we want to know how loud a sound that voltage represents.
The Classic VU Meter
Today, a VU (or pseudo-VU) meter is most often used for little more than to indicate an impending overload. We don’t watch a recorder’s VU meter to tell how loud our recording is, but rather, to ensure that we don’t exceed the available headroom—the 10 dB or so between 0 VU and the point where THD reaches 1%.
However, the VU meter is only a good headroom indicator if you’re working with program material that’s fairly consistent in level and doesn’t require a generous allowance for surprise peaks.
Take a close look at the VU meter scale. While the meter scale has a total range of 23 dB, fully half (the top half) of the scale represents only 5 dB.
This is good resolution for measuring steady tones when calibrating the recorder or setting levels within a system but pretty wasteful when working with a recorder that’s capable of handling a dynamic range between 65 dB (analog tape) and better than 90 dB (garden-variety digital [QQQ 16/44.1?]).
There’s no usable resolution below -10 VU. If we assume that we have at least 10 dB of headroom above 0 VU, a VU meter is really only informative within about a 13 dB range.
The classic VU meter scale
Why such a compressed scale? Practicality. Perceived loudness is a logarithmic function. It takes more than twice the signal voltage for something to sound twice as loud. A linear scale that represents loudness just wouldn’t look right. Remember that one of the design criteria for the VU meter was that the pointer response looked good for speech.
It’s no surprise that modern, highly compressed music will shoot the meter pointer well up scale, and it’ll stay right there until the fadeout. On uncompressed material with a wide dynamic range, there’s plenty of audible material down below the -20 mark on the VU meter, but an inexperienced engineer who trusts the meter rather than his ears can be misled into thinking that anything that barely moves the meter is too soft.
Today, many meters, both mechanical and LED or LCD ladder-style, have scales that look like a VU meter but don’t meet the VU standards. These are useful for establishing steady-state calibration levels when setting up a system, but they don’t accurately represent loudness or headroom.
LED ladder meters are often found on digital equipment but until you dig into the inner workings, you usually don’t know whether the meter indicates an analog voltage or the amplitude of a digital sample. In either case, the garden-variety meter doesn’t provide the same dynamic response of a real VU meter. It can show you average and sometimes peak level but it won’t tell you much about apparent loudness.
Zero VU is an arbitrary voltage level. It’s whatever is “normal” at the point in the circuit where it’s measuring. A meter that indicates input or output level is generally calibrated according to one of several industry “sort-of” standards. The most common is that 0 VU represents a voltage level of +4 dBu, about 1.23 volts RMS.
Most modern mixers are calibrated to this convention: When the output meter reads 0 VU, the device is putting out +4 dBu RMS. But this isn’t always the case. For many years, Mackie believed this was too confusing so they calibrated their meters so that 0 VU = 0 dBu. The “semi-pro” recording gear popular throughout the 1980s was generally calibrated for 0 VU = –10 dBV (10 dB below 1 volt, about 0.32 volts). Professional recorders are usually calibrated so that their meters read 0 for an input level of +4 dBu. In the broadcast world, “line level” is often +8 dBu, so that’s where their VU meters are calibrated.
Are you beginning to see what the “madness” is in the title of this article? Wait! There’s more!
A digital meter—which is not a VU meter—has a scale with 0 dB all the way at the top. (Shown below is the Selected Channel level meter from a PreSonus StudioLive 32.4.2AI digital mixer.)
This doesn’t represent a specific analog voltage level but rather represents maximum digital level – a sample represented by a binary number with all the significant bits turned on.
Digital meters (regardless of what the scale says) measure dB relative to the maximum value (“full scale”) rather than a nominal value, as with an analog or standard VU meter. We call this dBFS, where the all-bits-on value is represented by 0 dBFS.
Unlike the VU meter, with a digital meter there is no headroom above 0. You can’t turn on more bits than the system’s word length. It’s up to you, the engineer, to decide where to set the analog-to-digital converter’s input gain to allow as much headroom as you’d like.
The object is to leave enough analog room so that only the loudest peaks approach a digital level of 0 dBFS. While today’s digital systems typically accommodate internal processing that yields a word length greater than what goes in or comes out, a 24-bit analog-to-digital converter is flat out when that 24th bit is turned on.
The digital LED meter scale
Perceived loudness is the same whether the source is analog or digital but we view headroom and operating range differently in the two worlds.
But How Loud is It?
Loudness is a function of both the recorded level and how far the listener has the volume turned up. In the past few years, commercial (and following in their footsteps, independently produced) recordings have been in a race to make each recording sound louder, when played at the same volume setting, than the previous one. A whole segment of our industry has sprung up as a result.
In a presentation at the October 1999 Audio Engineering Society convention, mastering engineer Bob Katz of Digital Domain described the concept of monitoring at a calibrated sound-pressure level. He proposed a new type of meter scale that essentially displays headroom relative to the calibrated monitoring SPL rather than the digital level. His meter scale looks like a VU meter in that it’s calibrated both above and below 0 VU, but unlike a VU meter, it has a linear scale.
The K-System, then, is an integrated metering system tied to monitoring gain, and it is intended to standardize the levels at which sound is mixed and mastered.
Those Pesky Reference Levels
The real meter madness associated with digital level measurement is when you have devices with different reference levels (volts or dBu vs. 0 dBFS) in the same system.
To make this even more befuddling, you don’t always know a device’s reference level unless you measure it. A lot of gear is specified only in terms of a nominal analog reference level, without revealing the corresponding digital level.
Further, many digital I/O devices have no input or output level controls, so you can’t easily calibrate the reference level to match other gear in your system – you have to accept whatever calibration the manufacturer gives you. It’s bad enough when there’s a standard and not everyone follows it, but in this case there’s no standard for the analog level equivalent to 0 dBFS.
This leads to the common complaint of “my mixes aren’t hot enough” or conversely, “it plays much too loud.”
A dirty little secret is that budget-priced A/D converters (whether a stand-alone converter or integrated into another device) tend to be a bit on the less-sensitive side. It’s more likely that +4 dBu going in will give a digital level in the -20 dBFS ballpark than -12 dBFS. The reason is that with less gain on the front end, they’re digitizing a lower quiescent noise level, so the manufacturer can advertise a lower noise floor on the digital side. You need to hit the input pretty hard in order to get to 0 dBFS.
Sometimes you can crank up the output level of the source—for example a mixer—and sometimes, as with a microphone, you can’t. If there’s something you can adjust, you must take care that you don’t push the source feeding the A/D converter into clipping before the converter reaches maximum digital level. This can occur if you have a mixer with little headroom (a greater risk with older semi-pro mixers than with modern ones) or if you have a mismatch of nominal analog operating levels between the mixer and recorder. If you’re using a recorder with +4 dBu input sensitivity together with a mixer with a nominal operating level of -10 dBV, the only way you’ll be able to get the recorder’s meters to approach zero on peaks is to run the mixer’s output level meter well above zero, which will seriously compromise, and may exceed, the headroom in your mixer.
Loudness in the 21st Century
In the past few years, the audio industry, primarily in response to complaints that television commercials are too loud, has developed a new set of standards for loudness. This is primarily a broadcast thing but there’s now a variety of loudness meters that measure compliance with such loudness standards as ITU-R BS.1770. This is pretty complicated stuff and beyond the scope of this article but don’t be surprised if your DAW gets a plugin for it eventually.
Keep Your Eye on the Ball (Not on the Meter)
Once you’ve properly calibrated the gear in your studio, meters can tell you a lot about what’s happening, but don’t become a slave to them. Use your meters as tools, and you’ll spend more time recording and less time worrying about whether your levels are set properly. Don’t forget that you have a playback volume control. Leave getting the hot levels to the end of the process.
Mike Rivers has an electronic engineering degree and over 35 years as a design, system, and government engineer, while also operating a part-time recording studio and remote-recording truck. He is retired, but he continues to take on occasional projects, such as writing for PreSonus Audio Electronics. The original article, along with plenty of additional links to reference material, can be read here.
Monday, April 25, 2016
The High Pass Filter, Your Best Friend
What Is It?
A high pass filter, or HPF, is exactly as it sounds. It is a filter we can use on our soundboards that ONLY allows the higher frequencies pass. It is sometimes referred to as a Low Cut filter for a similar reason. It is also the most overlooked tool in the sound engineer’s arsenal.
Where It’s Found.
Some soundboards only have a High Pass Switch which is fixed at a certain frequency, often 80Hz or 100Hz. This includes most Mackie, Behringer, Allen & Heath and similarly priced consoles. Usually, only on higher priced consoles do you find the most amazingly useful type… the coveted golden ticket… the end-all-be-all… the “Variable High Pass Filter.”
The variable high pass filter is more useful because it allows you to change the frequency where the cut off begins, or more importantly where the lows no longer muddy up the bottom of our mix. But rest assured, I have a little trick for you folks not yet blessed with a variable HPF.
Why We Need It.
Well, simply put, the more low frequencies allowed into a mix, the more muddy or unintelligible a mix usually is.
Let’s take a violin for example. For the most part the violin is made up of mostly mids and highs. So if we have 4 mics on our violin section, we are probably picking up a good deal of low frequency content from the timpani, bass guitar, kick drum, and so on. The problem is that the leakage from the other instruments, into our violin mics, is out of time with any of the close mics on the low frequency instruments.
Let’s take a short trip back to physics class. Sound is made up of waves, waves take time to move through air, and low frequency waves are longer than high frequency waves. Son if one mic hears two sound sources arriving at the mic at different time, we can say they waves are out of sync. When waves are out of “sync” with each other we have cancellations and/or additions.
It is best to not have multiple mics picking up multiple instruments, especially if they have the same frequency content, but are different distances from the source.
Like I mentioned above. If the violin mics were picking up the bass guitar, it would be safe to say that the low frequency leakage of the bass into the violin mics is not “in time” with the actual bass input. Which would result in some of the bass guitar sound being compromised because of the out of time (or out of phase) leakage into the violin mics.
What Do I Do With It?
If you are lucky enough to have a Variable High Pass Filter the trick is to engage it and while listening to the violins play, sweep their HPFs up until you hear their lower notes change. At that point, back it off just a little bit, and know that the bass guitar leakage has been eliminated from the violin channels.
Did you follow that? By making the HPF higher, but not so high it altered the low notes of the violin, we have effectively eliminated any lower frequencies from leaking into those inputs and ultimately into our mix.
What If I Don’t Have A Variable HPF Or I Have A Fixed Frequency One?
So you more moderately priced Mackie, Behringer, and other folks are feeling a bit left out at this point. I wish we all had unlimited budgets to buy the consoles that had this feature, I realize they are expensive consoles, and sadly I know how that one goes.
Here is the trick for you guys. Almost all soundboards have at least a HPF switch that can be engaged. First trick… engage it on all channels except things like Kick, Bass, CD, Video, and anything else that has the potential to make really low notes.
Now since you do not have a variable HPF what else can you do… well you can use your low EQ to do a similar trick.
The low eq knob on most of the consoles at this price point are what is called a shelving filter. Which means everything below that frequency is attenuated similarly. So even though you still can not sweep it up to hear the low notes cut off, you can still clean up a little more low frequency leakage by turning this eq knob down.
What you do here is similar to the variable folks. Listen to your instrument, and have them play some of their lower notes. Turn down your low eq until you hear a substantial change in the sound of the low notes. Then turn it back up just a notch. You have now cleaned up any leakage from those mics similarly to how the folks with the variable HPFs were able to.
Why It Cleans Up The Sound.
It’s actually not just about the leakage. It is also about finding the holes in the mix for instruments. If an acoustic guitar is to be placed in a contemporary mix with electric guitar, bass, and keys, then the low frequencies of the acoustic are really not necessary. That is not to say you should make it sound like a swarm of bees, but the bass and electric guitars are certainly more capable of providing low frequencies. So if the acoustic is also taking up that range in the mix, it is very likely that section of the frequency spectrum will easily get clogged up.
How Does It Help The Amplifier And Loudspeakers?
Eliminating any unnecessary low frequency content also helps our amplifiers and speakers. Amps and speakers pretty much do what we tell them. So if we have a sloppy low and low mid section of our mix, they will reproduce it just as we mix it, but if we clean up our mix by eliminating conflicting and extraneous low and even low mid frequency content, we not only get a cleaner mix, we actually allow our speakers and amplifiers to run more efficiently since we are not asking them to reproduce content that is not necessary.
A 20-year veteran of working sound on the road, John Mills is the Education & Development manager for Morris Integration.
Friday, April 15, 2016
Barry Weir Jr. Streamlines Workflow With Blue Sky’s Audio Management Controller
New AMC monitor controller supports up to 7.1 channel configuration and handles monitor measurement and calibration
Barry S. Weir Jr., a re-recording mixer and sound designer with Hollywood-based post production company Levels Audio, works on a wide variety of projects in any given week, requiring him to frequently switch between nearfield monitor configurations and calibration modes.
In late 2015, Weir installed Blue Sky‘s Audio Management Controller (AMC) to handle those monitor switching and calibration tasks.
“I decided to go with the AMC because it is one of the only monitor controllers on the market in the $2500 price range that provides so many features,” explains Weir. “Because the AMC is a monitor controller that supports up to 7.1 channel configuration and also handles speaker measurement and calibration, it was the best monitor controller for my workflow and studio setup.”
Weir has been with Levels Audio since August 2009, working on the Emmy Award-winning audio post team for HBO’s The 25th Anniversary Rock & Roll Hall of Fame Concert and taking a variety of roles on a wide range of projects, including feature films and television series and specials, promos, trailers and music shows. His credits include Emmy-nominated primetime TV series such as American Idol, America’s Got Talent, Cake Wars, So You Think You Can Dance, The Amazing Race and The Voice, on which he has worked variously as a re-recording mixer, sound designer, sound editor, sound effects editor, and voiceover recordist.
“Because I work on a wide variety of projects, I really need the flexibility to switch quickly between different monitoring setups, going from 5.1 to stereo, not to mention various room EQ calibration, from X-Curve to flat response, and bass management on or off,” Weir elaborates. “With the AMC Remote located directly in front of me within arm’s reach, it has really enabled me to streamline my workflow. I’m able to quickly and seamlessly switch between different monitor configurations and speaker calibration modes, bass management on/off, individual channel delays, levels and input source selections.”
Weir’s room setup is compact yet powerful, and places all of the important audio tools immediately under his hands. “My studio is only 100 square feet and I don’t have too much room for large equipment, so I require gear that has a small footprint but large firepower,” he says. Immediately in front of him is a Native Instruments Komplete Kontrol S49 keyboard, Avid Artist Mix and Control panels and RTW Primus meters, all centered around the Blue Sky AMC. His reference monitor setup comprises a 5.1-channel ADAM Audio system (three S3As for LCR plus two S3X surrounds) with a Genelec subwoofer, plus a pair of Behringer Behritone C50A mini speakers.
“I’m running all of my audio from my Pro Tools 10/11 rig out of an AVID OMNI interface via AES directly into the AMC. I’m using an ART TRS 48-point patchbay and my AMC is normalled to my ADAM Audio 5.1 monitor setup. Channels 1 to 3 are my ADAM LCRs, channel 4 is LFE, channels 5 and 6 are my surrounds and I use channels 7 and 8 for stereo playback through my mini speakers. Being able to use any speakers with the AMC is really convenient,” he comments.
The AMC ships with Blue Sky’s Speaker-Room Optimization (SRO) software, which combines precision room measurement and calibration tools with powerful corrective equalization capabilities. “I really like the SRO software,” Weir reports. “It was pretty easy to understand out of the box. Anyone with common professional studio knowledge can operate it. There are a lot of advanced features that I have been digging into more, which will enable more precise room calibration.”
Indeed, Weir has already immersed himself in the software and has had some suggestions for enhancements. “The customer support from Rich Walborn [Blue Sky’s chief technical officer] has been amazing,” he says. “Anything that I had a question about, or any bugs that I found, or any features that I suggested, he quickly addressed and implemented as new software and firmware updates. Not too many people or companies respond that quickly.”
Lightning Boy Audio Returns With New Op-2 Comp Tube Compressor Pedal
Features hand wired point-to-point construction, NOS paper in oil capacitors, and a pair of new-old-stock 12AU7 vacuum tubes.
Lightning Boy Audio (LBA) returns to the FX pedal business with the release of the new Op-2 Comp vacuum tube powered optical compressor pedal.
Designed from the ground up to deliver studio quality compression in a stomp box format. Op-2 Comp offers a wide, well balanced frequency response, which makes it suitable for both electric guitar and bass. Unlike its predecessor, Opti-Mu Prime, Op-2 Comp dishes out a very clean sound with low noise. It manages to do this while providing more gain and a wider range of compression than the former.
Op-2 Comp has the same simple feature set as its predecessor, supplying the user with a compression knob, volume knob, knee switch, and power switch. Inside the pedal is the same photo resistor found in the classic studio compressor, the Teletronix LA-2A. This device provides the auto-release characteristics of that much loved studio staple, while having a faster attack time from its LED light source. The heart of the pedal’s tone comes from its pair of new-old-stock 12AU7 vacuum tubes wired up with a healthy dose of Lightning Boy’s own secret sauce.
Op-2 Comp is 100% vacuum tube powered and runs off a standard 9V DC power supply. However, it is power hungry so make sure you have at least 1270mA available from your supply. The pedal is hand wired point-to-point and is made with NOS paper in oil capacitors (just like a quality vintage amp). Made in the USA and ready to ship when you order. Op-2 Comp is available online direct from Lightning Boy Audio for $399.99 USD.
Lightning Boy Audio
Posted by House Editor on 04/15 at 09:47 AM
Thursday, April 14, 2016
Audient Console Selected For The Laundry Foley Studio
Analog ASP4816 console chosen to complete brand new studio facility in Essex.
“An incredibly quiet recording chain is achieved with an Audient ASP4816 console,” says The Laundry’s website, when detailing the equipment used in the brand new dedicated Foley studio, which opened in Essex at the beginning of the year.
Combining the compact, analog desk with a 3m x 1.7m projector screen, a NEC 4k projector, JBL 3678 screen channel speakers and two Crown DSI amps, they’re all set for high quality playback, while creating all manner of interesting sounds.
Owner Barnaby Smyth cites, “Patience, musicality and articulate direction,” as prerequisites for the job – he clearly has all of these in spades. “It’s hard work, but hugely fun,” he says. “If you want to good Foley you have to immerse yourself in the scene or character you are performing. Half-hearted Foley is all too easy to spot.
“The challenge is trying to achieve good recordings for a huge dynamic range of sounds: from face touches to car crashes. I have a lot of ‘go-to’ objects, but each project usually throws up new situations in which we have to create new sounds. This often involves sourcing new props – something that I’m now in a better position to do, having my own studio,” he explains.
“We tend to perform with the scene on screen, but we record a lot of wild tracks – not to picture – which are useful in the edit.” Audient wonders if he can describe some of the more peculiar ways of creating sounds. “When I’m rubbing my chest for lovemaking scenes, and kissing my hand and my arm,” he suggests.
The edit suite is located upstairs from the main studio, which where the Audient desk has fitted in a treat. “It’s been fantastic. Great, clean mic pres and the EQ is very versatile. It’s small but very flexible with a great number of returns from ProTools available.”
Surrounding the desk is an array of seemingly disparate items all of which help create sound effects. A glance at the company website is enough to confound visitors: row after row of shoes, a butler sink, an alarm clock, a chest of drawers and numerous floor coverings, ensuring The Laundry has the capability to produce “…a range of sounds from rickety old house, to plush stately home.”
With his impressive credit list, including The Night Manager; Tinker, Tailor, Soldier, Spy and Downton Abbey, Smyth and his colleague, Emmy nominated Foley engineer Keith Partridge are a formidable team in this new venture. Between them they have a broad professional network to draw on, especially for TV work. “Clients, editors and mixers I have worked with and have a good relationship with, bring return work. Films are slightly harder to come by, but I usually do a few a year.
“Now that I have my own place, the best thing about my work is that I can do what I like with the studio and develop and improve as I need to,” he adds, definitely happy in his work. “We are being creative all day.” What’s not to love about that?
Posted by House Editor on 04/14 at 01:15 PM
Wednesday, April 13, 2016
SALZBRENNER media Introduces NIO xcel Series Dante-Based Audio Interfaces
New “plug & play” NIO xcel (NIO = Networked Input/Output) series of compact devices provide “Audio-over-IP” network for mobile or installation use.
At this year’s Prolight + Sound in Frankfurt, SALZBRENNER media presents its brand-new NIO xcel series of Dante based audio interfaces.
With their sturdy housing and professional connectors, these interfaces are perfect for any live scenario. The four models in this series are flexible and robust solutions for decentralized distribution of the required inputs and outputs exactly where they are needed.
While users in professional sound reinforcement, installation and studio settings rely increasingly on a network infrastructure, most existing solutions are both expensive and rather time-consuming to configure. SALZBRENNER counters this with a “plug & play” solution called NIO xcel (NIO = Networked Input/Output), a series of compact devices based on the Dante protocol for an inexpensive and reliable “Audio-over-IP” network.
With the NIO xcel series, SALZBRENNER media expands its product portfolio with four devices for both mobile and installed pro audio applications on stage. Two of the new devices (NIO xcel 1101 and 1102) provide inputs and outputs in the AES 3 format, another (NIO xcel 1201) offers 8 microphone inputs with 4 splits, phantom power and 4 line outputs, and the fourth (NIO xcel 1202) is equipped with 8 line inputs and 4 line outputs.
With these interfaces, pro audio users have all the relevant applications covered: feeding digital power amplifiers, inserting side-rack effects processors, providing microphone and line inputs for musicians and artists, and line outputs for monitoring purposes, and feeding analog power amps, press distribution systems, etc. The big advantage of these devices is that they can be positioned close to the sources and destinations and do away with need for separate DI boxes.
To ensure reliability and signal quality in rough and tumble environments, these format converters are fitted with professional connectors throughout. Dante and power supply redundancy are a given, and the interfaces can be stacked almost anywhere or mounted into 19-inch racks. Using conventional Cat 5 or Cat 6 Ethernet cables, they connect to an IP-based Dante audio network (100Mbps, 1Gbps, 10Gbps) that covers the entire stage, production studio, venue or installation.
While the underlying concept corresponds with that of proprietary network solutions from other manufacturers, thanks to adopting the Dante protocol this series is significantly more cost-effective, flexible, scalable, user friendly and future proof. Each device has its own web server for speedy configuration of many parameters and has 4-band EQs on all input and output channels and a freely configurable 16x16 mixer.
All NIO xcel interfaces that handle digital signals are equipped with sample rate converters that will accommodate all sample rates from 44.1 to 192kHz on every input. NIO xcel 1201’s microphone preamps provide 4-way splits, each with individual level adjustment, and a dynamic range of more than 152dB.
Other NIO xcel highlights include frame-accurate synchronization across several switches, negligible deterministic latency of the overall network, the flexible and scalable network topology with massive I/O counts, simple installation and intuitive operation, AVB support (TSN) and AES67 compatibility.
The new series will be on display at the SALZBRENNER media booth (E21/Hall 4.1) from 5th to 8th of April, 2016, where visitors of the Prolight+Sound exhibition will be able to get to know them hands-on.
Monday, April 11, 2016
Solid State Logic Releases Sigma δelta Version 2 Software Package
Update includes DAW-based analog automation, new Sigma Remote Control App, and MIDI-Over-Ethernet MCU control.
Solid State Logic has released a Version 2 software package for its Sigma remote controlled analog summing mixer.
Now Sigma owners can take advantage of three-way round-trip control and DAW-based analog automation with SSL’s δelta-Control (δ-Ctrl) plug-in technology, the new Sigma Remote Control App, and MIDI-Over-Ethernet MCU control.
This upgrade includes the V2 Sigma δelta firmware and the new Remote Control App (Windows or Mac OSX), and is available via Sigma Owners’ My SSL registered user profile on the SSL website.
δelta-Control technology uses DAW plugin architecture to enable automation of the Sigma δelta analog signal path. This streamlines Sigma as a standalone mix environment and also allows existing DAW automation to be transformed into SSL SuperAnalogue automation, as well as enabling easy transfer of Sigma δelta sessions to Duality or AWS-equipped studios.
The main interface for δelta-Control is a native AAX/RTAS/VST/VST3 plug-in inserted into DAW mix or aux channels. The plugin sends and receives Sigma level and mute control data via the SSL Logictivity Network (Ethernet), which can be recorded, viewed, edited, and played back as normal plugin automation. The ‘Paste Special’ command can be used to copy existing DAW fader automation data into the δelta-Control plugin.
Sigma δelta volume and mute control data can be entered using the plug-in GUI, or by direct control of the Sigma hardware via either an SSL MCU control surface (such as Nucleus) or the new Sigma Remote Control App.
Audio on the DAW track passes through the plug-in slot unprocessed so the δelta-Control plug-in can be combined with other DAW plug-ins.
The δelta-Control plug-in is available to purchase exclusively from the SSL online store and is compatible with AAX, RTAS, VST, and VST3 plugin platforms.
The new Sigma Remote Control Application offers full control of all Sigma functions, plus storage and recall of saved settings from Mac or PC. The user interface provides a series of intuitive pages that make controlling Sigma’s powerful feature set straightforward. Three main screens provide control and setup of the Master Section, Channel Control, and Global Settings parameters. Every parameter can be saved for future recall making swapping between projects simple and efficient.
MCU control for Sigma facilitates direct control over the Sigma analog mix path and monitor switcher from SSL’s Nucleus DAW controller or any other SSL, MCU enabled control surface. Sigma’s MCU control requires no DAW host to function, only an active Remote Control App.
Combined with the Nucleus, Sigma becomes a fully integrated 32 into 4 automated line mixer, with full monitoring and talkback capabilities.
δelta-Control is not currently available for the AU plugin platform (Apple Logic).
For Pro Tools, Cubase, Nuendo, and Ableton Live, δelta-Control plugin stores its volume data using exactly the same dB law as the DAW fader volume data and will translate to Sigma channels with an accuracy of better than 0.2dB.
Solid State Logic
Posted by House Editor on 04/11 at 06:57 AM
Are Headphones A Viable Tool For Monitoring Your Mix?
Editor’s note: This article originally ran in 2011, but the information is still just as useful and relevant today.
A question I’m often asked is “Do You Mix With Headphones”, which I think is something worth discussing.
To start, here are a few comments I’ve received in response to this very question:
I cater to the most popular form of listening. So far, I’ve found that mixing to headphones and then listening on speakers has worked. I’m not Rick Rubin, but neither is anyone else who is not Rick Rubin.
I do the main mix with Sony MDR-CD180 headphones, while checking with iPod buds, little Logitech laptop speakers and finally in my car. Between those, I can pretty much get it in the ballpark.
I must be doing something right – on my last CD, even my most pickiest of listeners actually commented on how good it sounded. (excuse while I break my arm trying to pat myself on the back)
Those both made me laugh out loud. So, Let’s talk about mixing with headphones.
1. Less-than-ideal Mixing Environment
Most people that use headphones do so out of necessity. If they had a properly treated acoustic environment with nice studio monitors, they would likely use those.
But since they don’t have a great mix room, they revert to headphones.
I’ve talked before about acoustic treatment in your studio. It’s absolutely a necessity for both recording and mixing. However, some people just can’t afford to properly treat their entire room.
They may only have enough money to treat a portion of the room to allow them to get a nice, clean recording.
When it comes to mixing, though, frequencies are flying all around the room. There are huge peaks and dips in the frequency response of the room itself. (My room, for example, has some serious issues in the 120-160 Hz range.)
All this craziness can make it very hard to get consistently good mixes. Acoustic treatment will help “flatten out” the frequency response of the room.
Headphones, on the other hand, don’t need acoustic treatment. They sound the same every time.
2. Increased Detail
Most people would agree that you can hear more detail on headphones than on studio monitors.
I always use headphones for editing, for example. I want to make sure I don’t miss any pops or clicks in cross-fades, etc.
When mixing, headphones can give you an added amount of detail with things like EQ, compression, panning, effects (reverbs, delays), level balance, etc.
3. Keeping Things Quiet
Many of us work with other people, or in less than ideal conditions. Also, many of us have day jobs, which makes “studio time” synonymous with “late nights.”
When I first got the studio monitors I have now, I was living in an apartment, and I could only work on music at night. I was so bummed, because I never had a chance to try out the monitors, since they would wake the neighbors.
Obviously there are some definite reasons to use headphones, so lets examine the down-sides.
1. Limited or Exaggerated Frequency Response
One of the reasons we get big 6-inch or 8-inch studio monitors is so we can actually hear what’s happening in the low end.
Headphones typically cannot reproduce the lows the same way that studio monitors can. After all, they’re small little mini-speakers, so we can’t expect them to thump like a 12-inch subwoofer.
Sometimes headphone manufacturers make up for this by boosting the low end in their headphones. This isn’t necessarily wrong, but you need to keep this in mind when mixing on headphones.
Since they probably don’t have a super-flat response, you need to know what the headphones are doing to your mix as you make your mix decisions.
2. Altered Stereo Image
When you’re mixing on studio monitors, when you pan something hard right, you’re still going to hear it in your left ear. The sound will travel from the right monitor, past your face, and into your left ear.
With headphones, this doesn’t happen. If you pan something hard right, it’s only playing in your right ear. This isn’t necessarily bad, but you might choose to pan things differently when mixing on monitors vs. headphones.
Also, the “center” of your mix is very different on headphones. When listening on monitors, anything panned to the center sounds like it is in front of you, between the monitors.
With headphones, anything panned to the center sounds like it’s in the middle of your brain. This may seem like a non-issue, but it can effect how you handle things like lead vocals, bass, kick, snare…anything panned to the center.
The lead vocal might sound great on monitors but way too loud in the headphones. Gotta find a balance somehow.
3. Lack of the “Wow” Factor
There’s something awesome about listening to a mix blaring loudly on a nice set of studio monitors. No matter how good you are at mixing on headphones, you’re really missing out if you never listen to your mixes on monitors.
You need to be able to crank it up and enjoy. For one thing, it’s just fun. Secondly, it’s a great way to check for issues in your mix that you can’t hear at lower volumes with headphones.
A word of caution, don’t listen at super-loud volumes, whether you’re using monitors or headphones. Listen at a “reasonable” level and protect your hearing.
So, that’s my take on mixing with headphones. Frankly, I don’t think there’s a right or wrong answer here. You really need to evaluate your current gear, room, experience, and budget. If you can treat your room and get some nice monitors, great!
If you don’t have a dedicated studio room, or if you’re constantly mixing in different environments (especially live or remote situations), it might be worth your while to get some nice headphones.
What do I do? I’ve had the pleasure of moving several times over the last year, and each room I’ve used for my studio has sounded very different from the one before.
Since they’ve all been temporary, I haven’t been able to really dive in and treat them. So, I use a decent amount of acoustic treatment. However, the room still has some issues.
So I use headphones to give me a “room-less” mixing environment. I mix for a while on headphones, then I mix for a while on monitors. I’ve found that if I make major decisions with headphones and then “check” them on the monitors, it tends to translate much better than the other way around.
The biggest take-away point here is that you need to learn how your system sounds. Even in a less-than-ideal mixing environment, you can get good mixes. You just need to know how to compensate for any problems your system brings to the table.
This is a long learning process, but it’s well worth it.
Joe Gilder is a Nashville based engineer, musician, and producer who also provides training and advice at the Home Studio Corner.
Friday, April 08, 2016
We live in the present and plan for the future. Every once in a while it can be interesting to look back at our history and see he we got here. This puts things in perspective and helps us see what’s coming.
The essential challenges to tuning a sound reinforcement system, a process we now call optimization, haven’t changed in 40 years, and neither have the laws of physics.
But the tools and techniques of the trade have changed dramatically, albeit incrementally, over time. As a person who has been in professional audio for 40 years, here’s my perspective of how it was and the journey to the present.
Let’s take a walk to front of house and meet our modern sound system. We see an engineered loudspeaker system and a multichannel digital signal processor loaded with all types of filters, delays, and more (maybe too much more). There’s an acoustical analyzer ready to guide the optimization process and an experienced operator with a step-by-step plan for system tuning.
Not too long in the past, none of these things would be found there. Just 35 years ago we would likely find a collection of various custom loudspeaker cabinets built in the rental company’s shop. The extent of the “signal processing” would be an analog crossover with fixed filter slopes and a graphic equalizer.
At best we would find a primitive analyzer that no one trusted and little, if any, methodology or scientific process. All these aspects of our sound system have evolved in the period since, each in their own way and together as a whole.
I’ve been an eyewitness to the evolution of these tools and techniques into their present mainstream forms, as were countless other engineers. We can explore this history together by following five primary threads: the evolution of loudspeaker systems, signal processing, acoustical analyzers, analysis systems, and the methodology of optimization.
This is a personal perspective, not an authoritative history (which would be an entire book in itself). Every veteran of the pro audio industry has played some part in our collective progress, for which I am grateful. I encourage any and all to share their journeys with us and enrich our knowledge and respect for our past.
The central thread in my story is a 32-year relationship with Meyer Sound’s Source Independent Measurement (SIM). This was the first analysis system capable of measuring the acoustic response of sound systems during a concert using the music as the test signal. The analysis system approach originated in SIM has helped lead to the analysis programs that are now so common that most live sound systems use some version of it today.
Stone Knives & Bear Skins
It may be difficult for the modern engineer to visualize the crude tools of the 1970s. The only commercially available professional loudspeaker systems were targeted for fixed installations such as cinemas and “public address” systems, and weren’t the least bit ready for trucking, stacking or flying. The enclosures were not ruggedized with internal (or external) steel framing or integral rigging.
An “old school” column loudspeaker, the Shure Vocalmaster, which was introduced in 1967 as the first complete system (mixer, amplifier and loudspeaker). Legend has it that the Beatles used this at Shea Stadium concert (1965). It was not, but the system on the field was not much more powerful than the Vocalmaster shown here. We’re using it for our “Let it Be” concert on the roof of a dorm at Indiana University in 1975.
Popular music concerts needed portable power on a scale beyond the imagination of the loudspeaker manufacturers. Rental companies innovated in developing mobile touring boxes while several manufacturers transitioned into the role of supplying the driver components inside them.
At the time I entered the pro touring market, systems were described by which rental house made the boxes and the maker of the drivers, i.e., “a Showco system with JBLs.” Nobody (at least that I knew of) was touring arenas with off-the-shelf boxes from a manufacturer. That was for high schools, cinemas, and “Hair” on Broadway.
Loudspeakers did not fly. They came out of the truck and stacked on the stage.
The only ones that did fly were designed by consulting firms, usually comprised of a big pile of horns and a few woofers that we termed “flying junkyards.” You can still see a few of these gathering dust in the ceilings of old arenas.
It got loud in front of our ground-stacked systems. It wasn’t “hi-fi,” it was rock ‘n’ roll. It was supposed to be rough and edgy. We all knew that you could either have high power or high fidelity but not both. The tradeoff in favor of power was simply accepted as normal as long as it got loud. Really loud.
The arena rigs I toured with in the 70s while at Showco and FM Productions were 4-way systems with 15-inch woofers, 12-inch woofers, horns and tweeters all in separate boxes. We ground stacked them on the stage and pointed them in the general direction of the audience.
For bigger shows, the stacks were taller and wider. Frankly, one of the key sound design principals was making sure it didn’t fall over. Other companies, most notably Clair Brothers, made single boxes that were complete 4-way systems, but again, the same quantity and stacking principles applied.
I remember the first time we flew an arena system. We stacked the loudspeakers into a big steel basket with a plywood floor and up it went. For real. More “sound system in a freight elevator” than “flying system” but the progress toward level uniformity was amazing. It did not have to be insanely loud in front to be stupid loud in back!
A 1979 concert with FM Productions at the Greek Theater in Berkeley. We are struggling with a custom 3-way box that has an HF horn tacked on the top of it (a primitive attempt at time-alignment). The box had minimal handles cut into it and an “L-track” screwed onto the side for rigging. (Credit: Clayton Call)
The signal processing of the time was comprised of a 4-way crossover that drove the whole system, and simple limiters/compressors. Phase alignment was mostly just talk, since we had no tools to control it (no delay lines) or quantify it (no analyzer). This left us with the visual version: physically lining up the boxes as best we could. Again, the “don’t let the speakers fall over” principle came first.
Level adjustments were done at the crossover or at the amplifiers, the latter being a notoriously difficult method to obtain consistent results. Typical level settings for power amplifiers in that era were “2 clicks down” or “3 o’clock.” We had very little idea what the actual gain values were or how two different models related to each other. (By contrast, the modern amplifier can be software controlled in precise settings read in dB.)
Jorma Kaukonen at the 1980 Berkeley Street Fair. This era marks the beginning of the transition from custom and user settable loudspeakers to systems (processed). I was mixing monitors and had the chance to compare the new “processed” system, the Meyer Sound Ultramonitor (on the left) to a pair of conventional “do it yourself” loudspeakers. The single UM-1 was much clearer and more powerful than the old school pair. There was no going back. (Credit: Clayton Call)
The concept of the engineered system barely existed in the 1970s. Custom was king and users were expected to be able to add their preferred touches to the settings of the crossover and other electronics. Engineered systems, by contrast, fixed these parameters to precise settings selected for the particular combination of drivers, horns and enclosures.
This doesn’t sound radical now, but it sure was then. My first encounter with an engineered system was the rental company McCune Sound’s JM-10, designed by a young John Meyer. I stood there stunned as one of my long-held beliefs was shattered: yes, you could have high power and high fidelity at the same time in an arena.
The 1980s were a time of tremendous evolution in loudspeaker technology. For one thing, the dinosaur boxes with separate lows, mids, horns and tweeters gradually became extinct.
Basically the industry settled toward two box types: “full range,” i.e. 2-way or 3-way boxes that covered from 70 Hz on up, and subwoofers for 100 Hz on down.
Manufacturers also began to make ruggedized enclosures ready for touring. The engineered systems up to this time had been kept as exclusive and proprietary by the rental companies. We now began to see this approach become non-exclusive as the manufacturers embraced it.
The integration of multiple components into a single box with fixed physical characteristics set the stage for the next level: the introduction of “processed” systems.
Loudspeakers were sold as a “system” with dedicated loudspeaker controllers that provided fixed crossover settings, amplitude and phase response correction, and limiters. All of these parameters were pre-optimized for the specific driver/box combination.
It was the beginning of the “plug and play” paradigm that has become the industry standard today. It may be surprising to know that this was highly controversial at the time. Many engineers felt that the manufacturers were limiting their options for optimization by locking out parameters such as crossover frequency and slope, etc.
The main system shown here for the Grateful Dead at the Greek Theater in Berkeley (1984) had minimal subdivision, entirely driven by a single channel of equalization with no delays or relative level adjustment. The standard optimization options of a modern system (macro and micro delay, relative gain adjustment and separate EQ for the uppers/lowers/side fills/etc. were not available options at that time. (Credit: Clayton Call)
Part of the resistance was fear of the unknown, since what was going on inside the proprietary controllers was mysterious. Another part was the feeling that manufacturers were taking away a part of their tool set. Many engineers took pride in their personal crossover setting skills and did not want to surrender that control to others.
It took a long time for them to realize that a complete box with fixed, known parameters is better engineered in a manufacturer’s research lab than on the job site. The benefits of fully engineered systems over the custom recipe from Joe’s Garage became so apparent that the custom boxes of the Wild, Wild West era rode off into the sunset.
The processing and amplifiers for the system at Orchard Hall in Tokyo, Japan in 1988 or so. This system had an abundance of subdivision: left/right, upper/lower, inner/outer, and so on. The single-channel loudspeaker processors are in the rack on the left while the amplifiers are in the rack on the right.
Bringing It Together
Modern professional systems all have fixed pre-optimization. We buy “systems,” not just loudspeakers, and expect them to be fully engineered with optimized crossovers, pre-aligned frequency response, appropriately scaled power amplifiers, and dynamic protection.
They come in two varieties: self-powered and those with external amplifiers containing dedicated presets. End users expect to deliver a line level (or digital) signal to the system that in turn results in predictable output.
Each level of system evolution enables an evolution in optimization. How could we optimize a crossover in the old approach where horns and woofers were stacked up next to each other?
Even with the best analyzer and digital processing there’s no sensible solution to that challenge (except a dumpster).
The standardized linear loudspeaker system opens the door to optimization. The sensitive crossover points have a fixed solution, which creates a known element that can be utilized to build arrays.
We can learn lessons about the coverage pattern, aiming, splay angles, compatibility with other models and more. Results can be predicted in advance because the response of the loudspeaker is standardized.
It’s personally embarrassing to think of how little I knew about loudspeaker array behavior back in those early days, but then again, it’s a subject complicated enough for an entire book. Even a simple, small array of identical elements is very complex, but the outcomes are predictable.
Known elements with known spacing and angular orientation will yield predictable results. This holds true with standardized engineered loudspeaker systems, but good luck with a system built in your garage and/or with wild parameters such as unmatched loudspeakers and processing.
A highly subdivided center cluster in a symphony hall with extremely complex 360-degree seating (and only one hang point). The front mains have upper/middle/lower subsystems with inner/outer. There are three subsystems for the sides and another for the rear. A total of nine channels of signal processing (EQ, level and delay) were needed. Each cabinet has independent pan and tilt to help facilitate optimal vertical and horizontal aim and splay.
Evolved loudspeaker systems exist at all power scales, running from Bambi to Godzilla and everything between. We see similar coverage patterns from 2-way systems whether they incorporate 15-inch or 5-inch drivers. They cover nearly the same frequency range, with the “big boys” reaching maybe an octave lower (9 versus 8 octaves).
The difference in power scale is gigantic though, which allows us to use a size proportional approach to achieve similar results. Recall the old school design principle: bigger venues use bigger stacks of the same stuff. The new paradigm is proportional scale: the same quantity of boxes can be used for a small or large venue, but we proportionally scale the elements up in size/power.
We define loudspeaker systems as coverage shapes and power scales. We can get 80-degree by 50-degree dispersion in all sizes, which makes it a scalable building block for both coupled and uncoupled arrays. After all, one person’s main is another person’s front fill. The modern line array is engineered to couple in large quantities in the vertical plane, but again the scalar paradigm applies.
This provides a picture of loudspeaker system evolution. Next time I’ll focus on the signal processing that drives them.
A modern era line array system (and the author) at the Appel Room, Jazz at Lincoln Center in New York. The main system is subdivided into four channels of signal processing (sub +3), and the various fill systems have independent processing as well.
Bob McCarthy has been designing and tuning sound systems for over 30 years. The third edition of his book, Sound Systems: Design and Optimization, is available at Focal Press (www.focalpress.com). He lives in NYC and is the director of system optimization for Meyer Sound.
Thursday, April 07, 2016
Radial Introduces Headlight-Pro Direct Box Designed To Handle Multiple Instruments
Designed to make juggling multiple instruments on stage easier for front of house and monitor engineers
Radial Engineering has announced the Headlight-Pro, a unique direct box designed to handle multiple instruments on stage.
The Headlight-Pro begins with a standard 1/4-inch input for acoustic guitar, mandolin, fiddle or bass. (It will also work well with many other instruments.)
A muting footswitch is provided to silence the switcher when connecting or disconnecting an instrument. The signal is then routed to a selector footswitch that sequences across four outputs allowing each instrument to have its own dedicated channel on the PA.
Simply mute the Headlight-Pro, connect the desired instrument, and then use the footswitch to select the output. High visibility LED indicators provide clear visual feedback on which output is active.
Radial tech team associate Mike Bauer explains: °When you are on a busy stage, switching instruments can pose several challenges. When sharing the same direct box, switching between an acoustic guitar and mandolin requires both the FOH engineer and monitor engineer to mute the active mixer channel in order to avoid damaging plug-in transients and painful pops for the audience.
“You then need to alter the EQ and adjust the level to adapt it for each instrument. Alternatively, when using separate DI boxes, you end up with a bunch of wires connected to each instrument which often get tangled. Ask any stage tech and they will tell you that the preferred approach would be to use a single cable and manually switch between instruments and ideally, do so without need for concurrent action from the audio engineer team.
“WIth the Headlight-Pro the artist can mute the instrument on stage and avoid having to flag down the FOH and monitor engineers between each changeover. Since each instrument is now connected to a dedicated channel, the signal level, EQ and effects can be predetermined and optimized without having to make radical changes every time an instrument is changed. With so many auxiliary musicians onstage these days, there has never been a more appropriate time for this type of solution.”
The Headlight-Pro has four active balanced outputs, each of which is equipped with a ground lift switch to help eliminate hum and buzz caused by ground loops and a 180-degree polarity reverse switch to either help time-align the PA with the stage amp, or help eliminate acoustic hot-spots on stage that can cause certain frequencies to combine and create feedback.
All of the connectors are made from glass-filled nylon for exceptional durability and isolation, and have nickel-silver contacts that will not tarnish over time. Two “set and forget” switches allow the user to limit the range of the selector footswitch to two, three or four active outputs. There is also a dedicated tuner out that is always on for on-the-fly adjustments.
The Headlight-Pro is made from 14-gauge steel for added rigidity and greater immunity against stray magnetic fields. It employs a standard Boss-style 9-volt power supply (not included) and is equipped with a cable lock to prevent accidental disconnection.
The Headlight-Pro will be released soon with an estimated MAP price of $279.99 USD.
Wednesday, April 06, 2016
The Classic Sound Of The Moog Ladder Filter
Some speculate that “Dr. Robert” by The Beatles, recorded on April 17, 1966, is a reference to the inventor of the music filter and synthesizer Robert (Bob) Moog (pronounced like “rogue,” not “fugue”).
Alas, although such a tribute would be well deserved (Bob Moog studied for his Ph.D. at Cornell University in ’65), the song more likely refers either to Dr. Robert Freymann, who apparently supplied the band with amphetamine cures for a variety of “ailments,” or to Bob Dylan, who allegedly introduced the band to pot!
Whichever the case, the track itself proved to be a premonition. Later that same year, one of the greatest and most important inventions of modern music technology was made: the ladder filter!
On October 10, 1966 (coincidently, the day after John met Yoko at the Indica Gallery in London), inventor Robert Arthur Moog filed the innocuous-sounding patent #3,475,673, titled “Electronic High Pass and Low Pass Filters Employing the Base to Emitter Diode Resistance of Bipolar Transistors.”
As the patent application slogged slowly through the system, Bob Moog worked to develop his first commercial modular synthesizers to compete with the monolithic and expensive RCA synthesizers of the time, which cost over $100,000!
These early Moog modulars found homes with Paul Beaver and Wendy Carlos, leading to the first commercial recording of a Moog, attributed to The Zodiac’s November, 1967, album Cosmic Sounds.
Original Moog patent for the classic Moog filter.
This filter-design patent was finally granted on October 28, 1969. Earlier that year, George Harrison had used his personal Moog Modular on several tracks on the pivotal Abbey Road album, including “Here Comes The Sun.” This was the first widely heard sound of a Moog synthesizer.
Moog business records also confirm that Mick Jagger of the Rolling Stones purchased a system during the same period, but rumor has it that it was only used as a prop. It was later sold to those ‘70s space cadets from Germany, Tangerine Dream. These early, wealthy synthesizer pioneers all recognized one thing: This new technology was something innovative and sounded special.
In fact, to use a modern colloquialism, it sonically “kicked ass,” and went on to become the design reference and sonic benchmark for nearly every synthesizer that has followed. Bob Moog is thus considered the founding father of the music filter, the synthesizer, and the original inspiration/catalyst for so much of today’s electronic music.
Bob Moog at the helm of the Moog Modular System.
The Moog Modulars And The 904 Filters
The first Moog synthesizer prototype was designed in collaboration with Herb Deutsch (a composer and lecturer at Hofstra University) from July through September, 1964. Bob published his AES Journal article, “Voltage Controlled Electronic Music Modules,” in July, 1965 (Volume 13, Number 3), and demonstrated custom modular synths in the same year, with the production models 1, 2, and 3 coming to market in 1967.
The AES article featured British-born Composer Eric Siday, who commissioned Moog to build the first percussion synthesizer, and was the second-ever Moog customer (Alwin Nikolais, a composer and modern dance choreographer, was the first).
The Moog 904s, the original building blocks of electronic music.
The Moog 904 series is the historic base-design for the Moog filter series from the early modular synthesizers and influenced all those that followed. The 904A is the 4-Pole 24 dB/octave low-pass filter (LPF), the 904B is the 24 dB/octave high-pass filter (HPF), and the 904C is the filter coupler, which allows band-pass filtering (BPF).
The 900 series prices in 1969 ranged from $125 for an envelope generator to $1,225 for a sequencer, which are ironically around today’s-dollar prices for plugins and DAWs! The 901 ABB formed the oscillator section, and the 902 the Voltage Controlled Amplifier or VCA.
Original MiniMoog “Model D” and original Moog factory in Trumansburg, NY.
The Model D (a.k.a., the MiniMoog)
Studio musicians wanted a cut-down version of the modular to take on the road. In essence, they wanted the Moog sound with portability. Despite some radical design concepts with sculpted plastic and sci-fi looks, the classic wood cabinet was chosen by the musicians that Moog polled. The MiniMoog filter was adapted directly from the modular synthesizers and the temperature-compensated oscillators and contour generators designed from scratch.
The MiniMoog model D debuted in June, 1971, at the National Association of Music Merchants (NAMM) Convention (following 1970 prototypes knows as models A, B, and C). The first Minis were originally manufactured in Trumansburg, New York, and the MiniMoog quickly established itself as the all-time classic mono-synth and formed the sonic backdrop of many early ‘70s classic records such as Stevie Wonder’s Fulfillingness’ First Finale.
Another MiniMoog view.
The MoogerFooger MF-101 Low-Pass Filter
The MoogerFooger introduced a new analog, voltage-controlled low-pass filter and envelope follower design by Moog at an attractive price. This put a MiniMoog-style VCF filter in a stomp-box form factor with knobs to control filter, cutoff, resonance, envelope, and mix. The filter has a rocker switch for 2-pole and 4-pole modes, a smooth-fast rocker for envelope follower response, and the drive knob allows harmonic overdrive of the input.
The Moog Voyager
The Voyager was first publicly touted in 2001 (although it shipped in 2002) as a new MiniMoog for the modern MIDI era, and clearly referenced the classic design cues of the original MiniMoog. There were many new features in the Voyager, but for the purpose of this article we’ll focus on the updated filter section, since this is all about Moog analog filters!
The spacing control, which allows the cutoff frequency of the dual filters to be changed independently, was a new core Moog filter feature. In dual low-pass mode, the spacing control changes the cutoff of the right filter channel only, which allows for interesting stereo effects when the outputs are hard panned.
In high-pass/low-pass mode, the spacing control moves the cutoff frequency of the high-pass filter and the cutoff only affects the low-pass filter. In this mode, resonance only affects the low-pass filter. Riding the spacing and cutoff controls together allows the pass band width to be dynamically changed.
The extended modulation matrix also added considerably more filter control than the original Moog, including the essential ability to clock-sync LFOs for BPM-locked filter effects for timed sweeps/chops.
The Moog “Sound”
What makes the Moog filter sound special has been the subject of many academic studies and much speculation. The filter can certainly be overdriven in a musically pleasing way, as all the transistor stages clip gradually.
But it is also that famous 24 dB per octave filter slope resonant sound that is central to the essence of the Moog sound! The way that you can crank the resonance without losing too much of the low end and sweep the cutoff and hear those beautiful “transitor-y” harmonic peaks. When Jason Gross, interviewing Moog for Perfect Sound Forever in March, 1997, asked him to name his proudest creation related to synthesizers, Bob made the most humble of comments:
“I’m well-known for the low-pass filter that is the basis of ‘the Moog Sound.’ It’s a simple circuit, but it works really well.”
Early Moog Synthesizer Music
1. The Zodiac- Cosmic Sounds ( November 1967, Water/Elektra)
2. Beaver & Krause – The Nonesuch Guide To Electronic Music (1967, Nonesuch)
3. Monkees – Pisces, Aquarius, Capricorn & Jones Ltd (November 14, 1967, Colgems Records)
4. The Byrds- The Notorius Byrd Brothers ( January 3rd, 1968, Columbia/Legacy)
5. Wendy Carlos – Switched on Bach (1968, Columbia)
6. George Harrison – Electronic Sound (9 May, 1969, Zapple Records)
Books And Media
1. Trocco & Pinch – Analog Days. The Invention and Impact Of The Moog Synthesizer
2. Moog – A Documentary by Hans Fjellestad
This article is courtesy of Universal Audio.
Monday, April 04, 2016
Audient Unveils New ASP8024 Heritage Edition Console
Builds on the original ASP8024 platform while providing several enhancements, including a new variable mix bus technology called RETRO IRON
Audient has introduced the ASP8024: Heritage Edition console, which builds on the original ASP8024 platform while providing several enhancements and a new look. Attendees of this week’s Prolight+Sound show in Frankfurt can see it at Hall 9.1, C25/29.
The SP8024-HE incorporates a new variable mix bus technology called RETRO IRON, about which Audient tehnical director Tom Waterman explains, “By utilizing an all-discrete and warm sounding output amplifier with Carnhill output transformers, RETRO IRON adds punch and ‘vibe’ to the console’s mix output.
“We based this design on two custom consoles from (David Dearden’s) first job in 1970 at the now legendary Advision Studios in London,” he continues. “The circuits used are a homage to the finest in late 60s American & British console technology, but remain unique to the ASP8024-HE.”
When engaged, the RETRO IRON output card also provides subtle low bump and high lift mix EQ’s inspired by classic mastering equalizers of yesteryear. They can be switched individually, allowing for wider, more spacious mix bus tones. The main mix bus has also been upgraded with John Hardy 990C discrete amplifiers as standard.
The VCA mix bus compressor now has smoother make-up gain circuitry, and includes a gain reduction meter and a high0pass filter in the compressor’s sidechain called Bass Expand. “This little tool tightens up mixes and adds punch without destroying the low end, particularly useful for bass heavy productions,” Waterman notes.
The look of the Heritage Edition takes inspiration from the past, including “Midnight Raven” coloring, vintage knobs, chunky fader caps, walnut armrest and a British design theme.
Customer feedback has resulted in further improvements, including the addition of a monitoring grade headphone amplifier, latching footswitch triggers for remote hands-free talkback, ALPS Blue Velvet main monitor pot with custom aluminium knob for a solid “everyday” feel, an improved FaderLink plugin (now 64-bit AAX, AU & VST compatible) controlling eight channels of patchable DAW integrated VCA automation for the Dual Layer Control (DLC) module, and an improved quiet power supply.
The new console also retains all of the familiar features of its predecessor, including fully featured in-line architecture, flexible four-band EQ; Class-A mic pres (used across the entire product range), monitor control, and a range of modular options, including DLC and integrated 24/36/48-channel patchbays or producers’ desks.
“Much more than just a facelift, the re-imagining of the ASP8024-HE ensures that this console, which has been first choice for studios and educational facilities for the past 18 years, maintains its status as a ‘modern classic’ and cements David Dearden’s heritage alongside his timeless design,” Waterman concludes.
The ASP8024-HE is expected to ship shortly, with 24 channels available from 17,100 GBP (32,800 USD MAP).