Tuesday, October 25, 2016
What’s Old Is New: A Vintage Recording Gear Tour De Force
In the early days of our industry, equipment was made to serve the user’s needs. Many of the original studio owners came from a radio background. They built their own equipment, from consoles to compressors to equalizers, etc.
They didn’t have cost-effective manufacturing in mind when they created this equipment. They built it in order to make superior-sounding recordings.
At some point, other people in the audio community asked these pioneers to build equipment for their studios. The world of professional recording equipment manufacturing was born.
While much of the original research and development was carried out by Bell Labs, many of the early studio owners took these designs to the next level for their own purposes.
For many working in recording today, vintage equipment helps them to realize their goals. But the term “vintage” is seriously overused - it applies very well to wine and guitars, but not so well to pro audio hardware.
Whereas a ‘62 Château Lafitte-Rothschild is a dream to behold, and a ‘60 Stratocaster (the first year they made the fret boards in rosewood) can very well change your life, I ‘ve never heard of anyone who would consider a ‘74 Neve 8014 desk superior to the ‘72 version (or vice versa).
Although a Neve desk may be identified as “vintage” by its owner, “old” is often the more descriptive (though arguably less flattering) term.
The mere fact that something is old and has tubes doesn’t necessarily make it good. Transformers, Class A amplifiers, big knobs, faded paint, inability to pass a square wave, excessive second harmonic distortion or the need of a forklift for installation - none of these features necessarily makes a piece of equipment good.
What does make it good is its usefulness in a given application.
The humble tube, object of much discussion and debate.
Though I am a sales weasel by day, I am an engineer on nights, weekends and other days off. I’ve done major label work and have a few Gold/Platinum records on the wall, and so have chosen to approach this overview from the point of view of my recording engineering practice.
What follows is a list of a few of my favorite things, and why.
A caveat. The history of the earlier days of the recording industry is under-documented. In putting together this article, I went through piles of manufacturers’ original spec sheets, old advertisements, etc., and found them to be almost completely useless. Most historical knowledge is passed on verbally and may be distorted along the line.
With that in mind, I recommend that you take the following with great big handfuls of salt. Articles are no replacement for experience.
There are two major schools of mic preamp design - tube and solid-state.
We’ve grown to love many of the solid-state sounds and designs, and in many cases they are superior to their tube counterparts. The tube models generally have a more distinct character.
Let’s look at some tube goodies first:
Among the most popular are the German brown book standard units. They were built by a variety of manufacturers for German broadcast and were used by other manufacturers (most notably EMI) for their consoles. The V-72, V-72A, V-76 and V-78 are the most notable. (The V-74 is a line amplifier, V-73 a comp/limiter.)
This is especially good for high-output microphones like the U47, U67s (pad removed), M49s, C-12s, etc. The V-72 offers a very musical shimmer at the top end, as well as a full tone and thickness.
The V-72 is a dual-tube unit employing two Telefunken EF-804S tubes. It has 34 dB of gain and a maximum output level of +15.
The V-72A used one E180F and one 5654 tube; it has 42 dB of gain and also a +15 output. This will work better with FET condensers (414s, U87s, etc.).
V-72S amplifiers were found in the EMI REDD Abbey Road consoles that were used on Beatles recordings up to Sgt. Pepper. These have 38 dB of gain, but you’ll never find one anyway so don’t worry about it.
The V-76 employs four EF-804S tubes. It has a variable gain for 0 dB to 76 dB. The V-76/80 has an equalizer (useless!) and a high-pass filter (nearly useless).
The V-76/M doesn’t have these features and to my ears sounds a bit better. It’s double the width of the V-72 and, coincidentally, about double the cost.
EMI REDD console
The V-78 is perhaps my favorite of the group. It has a variable input gain from 50 dB to 72 dB, excessive in most applications but controllable with a Shure A15AS variable pad. Its maximum output level is +24, so it will net you headroom for days.
The A15AS variable pad is about as sonically transparent as I’ve ever encountered. It is an inexpensive and indispensable tool, and they’re needed when using V-78 modules because their input gain is so high.
While we’re on the subject of mic preamps that require pads in most applications, let’s look at some of the stand-alone tube units from U.S. manufacturers.
The Langevin AM-16, Altec Lansing 458, a couple of models from RCA, and my favorite, Western Electric (WE) units run on 300 volts of DC.
They have enormous potential to sound really good and to make you really sorry you didn’t pay more attention before sticking your hand into one.
The Langevin has a very “sweet” musical tone. It doesn’t seem to distort quite as much as the Altecs and WEs but is very full and large sounding. It will give you wonderful clarity without sacrificing any of the bottom you are trying to achieve.
The Altec 458 and RCAs I find very similar in tone; they seem to be slightly clearer versions of the Langevin. No less solid in their tone, they have the ability to make any track incredibly present and pleasing to the ear.
In solid-state land, there are innumerable great mic preamps. The two most common are the Neve 1272 and API 512. Both are very solid and thick-sounding. The API is a bit clearer than the Neve, but the Neve accomplishes the “Neve” tone that nothing else can achieve.
I have found that using a combination of the two makes it very easy to separate my guitars and vocal when it comes time to balance the song. If I use the Langevin on drums, Altec on the piano, Neve on guitars, I will use API on vocals, or other combinations that complement the music.
The next step in my signal path is usually compression. As we are limiting (no pun intended) the scope of this article to things that are old, I will omit the half dozen or so new products that in many cases do what these do, only better.
Again, I’ll emphasize that not all new things claiming to be as cool as old things are cool at all; some are real dogs and should be avoided like the plague. Try as many things as you can before deciding what will work best for your style of engineering.
Fairchild 670 (click to enlarge)
The world leader in cost and performance is the exalted Fairchild 670. Testing for this 2-channel, 70-pound units was performed in the late Les Paul’s living room. Serial numbers 1-6 were production prototypes and sound slightly better than the subsequent production models.
Those subsequent production models (fetching thousands of dollars these days) must be heard to be believed! It’s like adding the “in-your-face fatness” you’ve always craved with the thickest, most controlled bottom you’ve ever experienced, and with a high end that just shimmers and dances to your delight.
When I don’t have one around, I go to Georgetown Masters in Nashville to have Denny Purcell master my record. Besides being one of the finest mastering engineers on the planet, he has his “Fairchildren” (a pair of 660s, the mono version of the 670) that always make my recordings sound like I almost have a clue.
It seems that many of the West Coast engineers prefer Attack/Release constant #4. I prefer #2 and #3, faster release times than preset 4.
According to the manual, numbers 5 and 6 are user presets, but you have to go inside with a schematic and soldering iron to change these. (By the way, #1 isn’t bad, it’s just really fast).
The 670 was originally built for disc cutting and has Lateral/Vertical controls to limit the movements of disc cutter heads.
I’ve found that on stereo percussion tracks (congas and the like), you can use this setting as an almost psychoacoustic device. It seems to push the congas out about a foot to either side of the speakers. Way cool effect.
The RCA BA-6A is another serious favorite. I have no idea how it accomplishes this, but anything run through it gets 10 times larger than when it went in.
They are also amazing mic amps, with enough gain so you can plug a mic directly into the input and go straight to tape.
They can put out as much as 95 volts at the output, so when using it on line-level sources I generally find the need to pad the input and output 20 dB each. This seems to get you closer to the optimal operating range for the unit.
RCA BA-6A (click to enlarge)
These are not low-maintenance units; they require careful care and feeding, and it should not be performed by anyone not intimately familiar with the unit. The folks at RCA seemed to be aware of this and incorporated a tube tester in the unit.
The setup is critical, or you might find yourself spending many hours dealing with a loud hum.
The Teletronix LA-2A is perhaps the most popular (and badly copied) of all tube limiters. The original LA-2s were made in Sunnyvale, Calif., and are identifiable by their gray painted face plate.
The most commonly seen models were made in North Hollywood and feature a brushed aluminum face. These later models (post-serial #383) featured a switch on the back to give you limiting as well as compression functions. In the earlier models, this was accomplished through jumpers on an octal socket.
Through their evolution, they went from a T4-A opto attenuator to the T4-B. While several folks swear they can tell the difference, I cannot.
In my travels, I’ve run across an ITA LA-1 and and LA-1B. They seem to be rarer than hen’s teeth. Ambient Recording near Stamford, Conn., has an LA-1B, and it sounds amazing.
Tubes have become all the rage due to the inadequacy of digital storage devices. There are numerous solid-state units that will sound better in many applications than tube units. One of the unfortunate by-products of tube units is the fact that their attack times can be measured with a calendar.
Many solid-state units, especially those employing opto-attenuators, will have a similar response. But some will not.
I believe it was during the late ‘60s that Teletronix was purchased by Universal Audio (all hail Bill Putnam!) and the LA-3A was born. For many applications, the LA-3A is a solid-state LA-2A. It does have a different tone.
While the LA-2A has an airy quality to its distortion artifacts, the LA-3A has a more solid midrange. It is a tougher-sounding unit, fat as you could ever want, and it has the ability to take a sound and move it right to the front of the speakers.
Universal Audio grew to become UREI, and the LA-3A was improved to become the LA-4. The LA-4 has much clearer audio than the LA-3A, and greater function control.
Now instead of having input-dependent ratios, we could select between 2:1, 4:1, 8:1, 10:1 or 20:1.
We were also trusted with input and output level controls, instead of the threshold and gain make-up controls given to us on the LA-2As and 3As.
This gave us the ability to tailor our compression needs, rather than being at the mercy of the designer’s idea of what we needed. Unfortunately, the designer seemed to know a helluva lot more than many of the unit’s users, so this was not necessarily a good thing. It was the beginning for the potential misuse of compression.
Universal Audio also built a tube compressor known as the UA 175 (175b), which with added control became the UA 176.
UREI came out with a solid-state version of these and called it the 1176. The earliest models were silver-faced with a blue stripe around the meter.
They featured push-button ratio selection 4:1, 8:1, 10:1, 20:1, as well as attack and release controls. These blue-stripe ones are still in favor with many well-respected engineers.
Perhaps the best use I have heard lately of these units is by Ray Kennedy (Room and Board, Nashville), who recorded the Steve Earle & The Dukes’ “I Feel Alright album.” I ran into Steve when he was playing in Boston, and he joked that they almost called the album “1176”.
The original blue stripes were replaced by the black-face 1176 LN. LN allegedly stands for low noise; I think it stands for less noise.
It’s a great box with a unique, very present character to the sound. It’s very easy to use, and it’s a no-brainer to see why the unit has soared in popularity.
Like its predecessor, it has four ratio buttons. Mixer Michael Brauer told me about pushing in all four buttons simultaneously. Wow! It’s the most aggressive sound I’ve ever heard from any piece, any time, anywhere. It’s so cool, you easily want to overuse use the effect, though I strongly caution against it.
If you’re brave enough to try the four-button trick, do not look at the meter without a healthy dose of Dramamine. It ain’t a pretty sight. There were about four incarnations of the black face, but I’m not clear on the differences. I’ve found that the lower serial number models seem thicker in tone, while later serial numbers are a bit brighter and faster-sounding.
The 1178 is a stereo/dual mono version of the 1176 with single controls, and is a very useful item with its own distinct character. I’m told that there was a black-faced version, but have never seen one.
UREI 1176, black version (click to enlarge)
On the 1176 LN silver face, the bean counters made them take out the input transformer (my conjecture), and the unit never had the same rich tone. The four-button trick doesn’t work as well, either. It’s still better than most new limiters on the market today, but not as cool as the black face.
UREI made a similar error in judgment with the LA-4, although I don’t find the silver LA-4 to be as bad as the silver 1176 LN. When it comes to UREI compressor/limiters, black is beautiful!
About the same time all that was happening, a small Massachusetts firm called dbx was making a comp/limiter called the 160. Like the LA-3As and LA-4s, it was two rack units (RU) high, and half of the standard 19-inch rack width wide, allowing for two units to be strapped together and rack-mounted.
This is one of my favorite limiters for percussive instruments. We’ve all had to suffer through the drummer who gets excited at the beginning of each new section of the song - you know, the genius who hits the kick drum 2 dB harder at the head of each chorus. This is my favorite box for controlling that excitement.
The dbx 160 has a tremendous amount of “grab,” and when used sparingly, it can erase the dynamic range from almost anything. You have to be careful not to overuse it, but if used well, it will fix a lot of problems. The 161 is an unbalanced version of the 160, and it works equally well.
dbx 160 (click to enlarge)
It can be balanced with a transformer; doing that will net you a slightly fatter tone than the differential balancing circuit in the 160. Use a good transformer! For best results, consult a tech who really knows analog circuits.
The 162 is a stereo version of the 160, operating on one set of controls. I have found its best use is across a stereo drum bus. It’s not a favorite for the 2-channel mix bus, but your results may vary.
Units that are considered vintage, or are at least rarer in the world of solid-state, are Neve compressor/limiters. The original units were approximately 5.25 inches square and were delivered in the consoles, typically the meter bridge. They were not intended to be rack-mounted or moved for that matter. The most commonly found are the 2254/A and 2254/E.
The 2254/A and 2254/E are almost identical, the difference being in the limiter function. The 2254/A has a fixed attack time, and the 2254/E has a selection between slow and fast attack. It you’re at all handy and can read a schematic, it is not difficult to alter the attack time of the 2254/A to the slow attack time of the 2254/E (or so I’ve been told).
I find the slower attack time more musical. A variety of compression ratios are available on both units. The 3:1 ratio is my favorite, but experimentation may lead you to a different conclusion.
The 2254s were found in the older (dark gray) 80 Series Neve consoles. When they changed the color of the desks to a lighter gray and began to employ black plastic knobs with various shades of blue in the knob insert caps, they added an extra “3” at the beginning of the model number.
So a 32254/E is the same as a 2254/E except for the paint job.
The next model in the progression was the 2264/A, most commonly found as the 32264/A. Whereas the 2254s are nearly square, the 32264/A is 1.75 inches wide by about 8 inches tall. The functional differences between them have as much to do with tonal differences as anything else.
The fastest release time on the 2254 is 400 ms on the compressor and 100 ms on the limiter. On the 32264/A, the fastest release times are 100 ms on the compressor and 50 ms on the limiter. This gives you a whole new world of possibilities.
Also, the stereo link facilities are right on the module instead of being an outboard afterthought as on the 2254s. The “A” or “B” link buses accommodate tying multiple units in a console.
The Neve 1073 is probably the most famous of all Neve input modules. It features a wonderful mic pre, line input and an equalizer.
There are two other modules that could have been ordered as alternates for the same console - the 1066 and its Cadillac sister, the 1084. The 1073 has a 3-band equalizer with a high-pass filter.
Neve 1073 module
The EQ points are: 12kHz shelving on the high band, six points in the mid band (7.2k, 4.8k, 3.2k, 1.6k, .7k, .36k), and four available frequencies on the low band (220, 110, 60 and 35 Hz).
The 1066 has a 10kHz shelf on the high band, five available points on the mid (7k, 3.6k, 2.4k, 1.2k, .7k) and the same four on the low. It also has a high-pass filter.
Needless to say, the 1066 and the 1073 complement each other very nicely due to the variations in frequency points.
The 1084 has 10/12/15 kHz selectable shelving frequencies on the high band, the same six points on the mid band as the 1073, and indeed the same four on the low band. It also features high- and low-pass filters, allowing you to have a bit more control over your high-frequency boost.
The coolest part of the 1084 is the high “Q” switch available on the midrange band. “Q” refers to bandwidth—the higher the Q, the tighter the bandwidth. This lets you get a bit more specific with your midrange equalization.
A 1084 without a line input control, black plastic knobs/switches with light blue caps, is a 31102. This is the little fella found in the 8066, 8058, 8068, 8088, etc. consoles. (Yeah, there is a line input, but it’s unbalanced and lacks control function; if you know that much this article isn’t for you anyway.)
These are the primary 3-band modules of the early 80 Series desks. (I could write another thousand words on the subtle differences of the other models in this range, but I’ll spare you.) They are also of the 1.75-inch x 8.75-inch frame size. The other frame size is 1.75-inch x 12-inch.
The most common 3-band module in that size is the 1064. It has the same function as a 1066 (EQ points) except instead of a dual concentric frequency select/boost cut function, they are laid out on two separate switch assemblies. It is important to note that all of these models are Class-A designed throughout.
In the same frame size as the 1064 is the 1081. This is the powerhouse of Neve modules from a functional point of view. It has a 4-band equalizer that features multiple frequencies, selectable on the high and low frequencies, and a switch enabling both shelving and peak/dip use. The two midrange bands also have hi-Q functions, allowing remarkably specific equalization.
The beauty of most Neve modules is that Rupert Neve is so much smarter than the rest of us; he built modules that really couldn’t be used to make things sound bad. There are generations of engineers who look like incredible geniuses because Neve wouldn’t allow us the tools to screw up our audio.
On the 1081 he gave us the tools, so I implore you to use the power wisely.
Neve 1081 module
On a kind of technical note, the 1081 employed a Class-B output stage. There is nothing bad about the models with the “push/pull” output stages - they will not achieve the same rich, flowing low-end characteristics of their 3-band Class-A brothers but give a better low-end punch and a slightly “airier” top.
Most of the “broadcast” series modules I have heard have the same output stage. The 3115 has an equalizer comparable to that of a 1066. The 3114 has functions comparable to the 1084 (sans hi-Q switch).
There are many more variations in this class; these are the two we see most often. Be very careful when purchasing broadcast series modules, because many of them have the dreaded 5534 IC chip. Basic rule of thumb: If the module runs on 15 VDC, it’s a 5534 model and is to be avoided.
Neve 8078 console with EQ module 31105
The equalizer in the 8078 console is called a 31105. For all intents and purposes it is the same as a 1081, except it has logic functions so you may put the entire console in mic or line input at the flick of a switch, instead of turning the switch on every individual input module. In a 40-input desk, this will save you a bunch of time. This is a good thing.
Referencing the 10 Series input modules to the 1272 mic amp for a moment, you will find the same input, same output transformer and the same B283 gain card in the 1272 as you will in the mic section of a 1073 (etc.), thus it sounds the same.
On consoles like the 8014, you will actually find the 1272 used as the talkback microphone amplifier.
Most of the routing modules (1883, etc.) also have the same input and output transformers and a half-filled B283 card.
Personally, I don’t like equalizers much. I’ve always felt that if you’re a really good engineer, and you choose your microphones and their positions wisely, equalization is unnecessary.
Granted, when you need to work too fast, they are a very handy tool and, when used sparingly, will enhance your project.
Most of the modern console manufacturers seem to agree with me - otherwise their equalization sections wouldn’t sound as terrible as they do.
Well, as long as we’ve opened up that EQ can of worms, let’s spend a couple of minutes on some of the cooler old ones. Older equalizers tended to be closer to tone controls. They were regional devices.
The Trident A range module is one of the input modules that combined four-band function with wonderfully musical characteristics.
Trident later came out with a single-rack-space, 3-band fully parametric, which gives unknowing users the opportunity to make something sound bad. Caution is greatly advised, grasshopper.
Trident consoles were, of course, originally built for Trident Studios in London. The owners of Trident Studios allowed their staff technical department the freedom to go off on their own and start a console company. The world is a better place because of this decision.
During the late 1960s, the folks over at Olympic Studios (also in London, and just closed earlier this year) had a genius named Dick Swettenham on their staff. He invented the Helios console. The original desk from their studio now lives at Keith Grant’s house, and the important bits were built into his custom Raindirk console.
Grant is making some of the most exciting recordings (from an audio perspective) with these modules to this day. The original modules were used on Jimi Hendrix’s “Are You Experienced?” album.
A Helios console at Olympic Studios (click to enlarge)
Olympic console number 2 was also built for Olympic. It can be heard on the Rolling Stones’ “Let It Bleed” and “Beggars Banquet,” and a copy of the desk was made for the Stones’ mobile truck for the recording of “Exile on Main Street.”
Folklore says that Chris Blackwell of Island Records wanted his artists to record in his studios, but they didn’t want to because he didn’t have one of these cool Helios desks.
Chris set up Mr. Swettenham with his own company and ordered the first five units. Helios modules are still available in loose form and are well worth investigating for the serious audio professional.
The kings of equalizers for equalization’s sake are now made in Virginia by a firm called API. They make an outstanding mic pre and, dollar for dollar, the best-sounding console under current manufacture, in terms of form without overkill function. (The Amek, Rupert Neve-designed 9098 is, in my opinion, the best-sounding desk that does everything but wash your car.)
API’s equalizer design is as cool as a Neve, with its own tone. Different, yes, but neither superior nor inferior. It can be chosen as the right tool for the job. When used wisely, the 550, 550A (a 550 with four additional frequencies) or the 550B (the new 4-band version—same design principle, equally cool sounding with greater flexibility) are very powerful tools. The 560 (10-band graphic) just rules.
API 550B (left) and 560B module
Kooster McAllister’s Record Plant Remote truck out of New Jersey taught me how great audio can be. His 48-input API console with 560 equalizers changed my life. The rest of my days on earth will be spent trying to re-create what I heard in that truck.
It wasn’t until the ‘70s that a couple of guys in Maryland added a bandwidth control. These smart fellas were Burgess MacNeal and engineer/producer George Massenburg. The box they built was called the ITI MEP-230. It featured three bands of fully parametric EQ, plus a 10 kHz shelf and a selectable 50 Hz/100 Hz low shelf.
Even though it was parametric, it still seemed that no matter how hard you tried, you couldn’t make anything sound bad. They made a console model as well, called the ITI MEP-130 - same function without the shelving band, and amazingly musical for a parametric.
Parametric EQ is, for the most part, used badly. It gives the user the ability to phase-distort a signal into complete submission. Very few in our profession should be granted a parametric license. This is one of the reasons that older EQ designs are so sought-after.
Recently, the term “British EQ” has popped up in our vocabulary. The British equalizers that I’ve used are all so radically different-sounding that this is, at best, an erroneous term. Let’s clue the marketing departments that there is as much of a British EQ sound as there is a British compression sound, as there is a British mic placement sound… ad nauseum.
And so it goes…
Fletcher moderates a popular REP forum here on ProSoundWeb.
Thursday, October 20, 2016
The Viper Room In Los Angeles Adds BAE Audio
Venue selects 500 series racks fitted with 1073 MPL and 312A preamp modules for weekly Sunset Jams.
BAE Audio announces that it has partnered with Los Angeles club The Viper Room to enhance the club’s signal chain with analog outboard gear.
BAE Audio’s Colin Liebich assembled a package of gear to provide a recording-studio-quality input section for The Viper Room’s mixing board. BAE has an established relationship with The Viper Room and owner Darin Feinstein, which has been extended throughout the year with the popularity of its Monday Night Sunset Jams.
“BAE Audio gear is well known in the recording studio circuit, but having a quality signal path at input is equally important for live environments,” Liebich says.
BAE Audio outfitted the venue with six 500 series racks, fitted with three 1073 MPL and three 312A preamp modules. “The 1073 MPL is loved by many for its ability to imbue vocals with a very desirable sonic color, making them sound both present and warm,” he says. “Meantime, the 312A excels at adding punch to snares and other drums, helping them cut through a busy live mix.”
“I’ve used BAE Audio microphone preamps and EQs extensively in the studio and on stage, so I knew they would be the perfect partner to elevate the Viper Room’s outboard rack,” says guitarist Erik Himel, architect of the Sunset Jams.
The sonic enhancements resulting from the BAE Audio gear were immediately evident to The Viper Room’s production manager Tyler Kunze.
“Instantly, the sound of The Viper Room was improved,” he says. “The drums were punchier, the guitars were more present, and the vocals were warmer.”
Kunze found himself having to apply far less equalization to individual tracks because of how good they sounded right off of the input stage. “On drums I used to need to apply drastic EQ to get a usable sound. Once we started using the 312A preamps, my drum tones were almost perfect right off the bat without any processing. The difference was just night and day.”
Meantime, BAE Audio and Himel have worked together to ensure that The Sunset Jam, hosted at The Viper Room, continues well into the future. Taking place every Monday night, The Sunset Jam allows local musicians the chance to sit in with top touring and recording musicians, including Dave “Chili” Moreno (Puddle of Mudd), Basil Fung (Alanis Morisette), Chas West (Foreigner), and Stephen McGrath (Billy Idol) while playing through The Viper Room’s sound system.
When Himel first came up with the idea for The Sunset Jam last year, his goal was not only to give musicians a place to network and connect with each other while honing their chops, but also to provide non-musicians with a lively, entertaining show featuring premium talent. “There are so many top session and touring players living here in LA, and I liked the idea of giving up-and-comers a chance to see what it’s like to play with people who are at the top of their game while giving the pros a chance to stay sharp and have fun in a relaxed environment.”
He knew he would need a high-quality space with top-shelf gear to attract the kind of talent he desired, so he approached the vaunted Viper Room, long known for being one of the best-sounding small rooms in LA.
“The Sunset Jam has made Mondays the highlight of my week,” says vocalist and regular Sunset Jam attendee Jayme Palmer. “The jams have given me an opportunity to expand my repertoire, gain experience in genres outside of my normal wheelhouse, and network with other serious musicians, not to mention the unbelievable sound. It’s invaluable.”
Electric violinist Koi Anunta agrees. “I’m so grateful to Erik, The Viper Room, and BAE Audio for making The Sunset Jam possible. To play through this sound system with these players has been incredible. I try to keep my Monday nights free every week so I can sit in.”
For Kunze, the gains of this partnership filter out into all of the shows he works on at The Viper Room. “The BAE Audio gear has been an absolute game changer for our live environment,” he says. “I couldn’t be happier.”
Thursday, October 13, 2016
Subjective Versus Objective: If It Sounds Good, Is It?
As with the ever-ongoing debates about “tubes versus transistors,” “analog versus digital” and “Mac versus PC,” there’s not likely to be agreement any time soon about “objective versus subjective” when it comes to sound quality.
Extremists in the “Objectivist” camp argue that, “if it can’t be measured, it doesn’t exist” while on the other hand, the “Subjectivist” side firmly backs the idea that “human beings can hear things that can’t be measured.”
How often has it been suggested, “use your ears as the final determinant” in making a decision about sound? At the same time, most would agree that a fundamental understanding of audio systems, including the basics of how each component works, how to set gain structure, and so on, logically can lead to “better” sound quality.
Does science (objective) or art (subjective) play the more important role?
ABX Or Death
Since its development as a scientific testing method, ABX has gained ground as a clear way to determine the threshold of perceptibility in a group of test subjects.
The basics of ABX: two different sources are compared - source “A” and source “B” - and the subject must make the decision as to whether choice “X” represents either A or B. If the subject can reliably (i.e. in a statistically significant manner) identify the sources, then it is concluded that there is a perceptible difference between the sources. Otherwise, the differences are deemed insignificant.
There are some good things to be learned with ABX, and it’s proven to confound many the “golden ears” in tests involving things like 44.1 kHz versus 96 kHz sampling rates, 16-bit versus 24-bit quantization, and others. And it turns out that it’s not common for subjects to be able to reliably identify these sources.
However, I contend that there’s a vast difference between a short-term test like ABX and a longer-term experience with a product, system and the subject itself. Humans have demonstrated a truly amazing ability to learn just about anything.
Take a person who’s never spoken anything but the English language, and stick him/her in Japan for a couple of years. This person will most likely learn to speak Japanese, engaging a new part of the brain.
Or take a person who’s only tasted wine costing less than $10 a bottle. A few months after being introduced to $150 bottles of wine (let alone $3,500 bottles!) and learning about the different varietals, harvest timing, and other specifics, he/she will balk at the cheap stuff.
Even more importantly, this fledgling student of wine will have picked up the ability to discern much finer differences between all types of wines.
In both cases, what changed these people? Exposure, mostly. We all have what some call “paradigms,” meaning that we each filter outside stimuli through our own various levels of experiences and beliefs.
Fixed Level Of Bandwidth
I call these changes through exposure successive thresholds of awareness, and contend that part of this is that human perception is scalable in terms of resolution. With computers and test equipment, there is a fixed level of bandwidth and resolution available.
Not so with people - the longer someone spends being exposed to an experience, the more resolution that person is able to impart to that experience. An analogy closer to home for us audio geeks: the person that has only used a cheap dynamic microphone for years will likely find that even the lowest-grade condenser mic sounds amazing. He will hear tons more resolution, less distortion, and better transient response.
This same person will also wonder how a Neumann mic costs much more, and whether or not it would be possible to sound that much better. And in fact, upon hearing the Neumann in comparison to the cheap condenser, he will conclude that indeed, there is not really that much difference between the two.
Now take that same person five years later, after he’s made several records and used a plethora of top mics of various makes. Now he should clearly be able to identify the differences between the cheap imitation and the real thing, having reached a much higher threshold of awareness between the different mics.
Only One Problem
A few years ago, I read an interesting article about how Dunkin’ Donuts intended to update its marketing plan to target Starbucks customers, based on a very simple idea: offer the same quality of coffee, but more quickly and at a lower price. There was only one problem. These weren’t the reasons that Starbucks customers were buying coffee from Starbucks. They didn’t want it cheaper or more quickly.
What they did want was the Starbucks experience—the club chairs, the subdued lighting, the fancy woodwork, the ridiculously overpriced accessory products, and whatever else they’re seeking. For this, they’re willing to wait (part of the experience) and pay more (another part of the experience).
Although it could be argued that they would appreciate the coffee being less expensive, it’s been proven over and over that there is usually a “right price” associated with a brand experience, and if the price is either too high or too low, the brand will lose credibility.
So what does all of this mean in terms of audio and the Subjectivists versus Objectivists? For one thing, different people perceive things differently, period. What’s important to some is not important to others, and visa versa.
For some, a slightly lower noise floor in a mic is not worth either the extra cost or the resulting lack of perceived resolution, while for others, it might be just the ticket for their application. Thus there can be no consensus on whether or not a lower noise floor is always “better.”
One thing I firmly believe is that both approaches are important for the improvement of audio (or anything else that is part of someone’s experience).
The Accidental Designer
Sure, there are stories where accidental discoveries made improvements in design. For instance, the story of the German broadcast engineer in the late 1930s that inadvertently left a high-frequency oscillator “on” while recording an orchestra.
The result? For the first time, there was playback fidelity beyond 10 kHz. This accidental discovery lead to the implementation of an AC bias for analog tape recorders, and it also pushed the envelope of what was possible with this type of system.
However, despite the muddled beginnings of AC bias, a scientific approach was required to produce repeatable, reliable and predictable results. The required circuitry had to be thoroughly understood by analog design engineers, and the right frequency and right amplitude had to be identified.
Then the right combination of these factors for each different tape formulation had to be developed in order to realize the full potential of the bias signal. It took until the 1950s before this was well understood, resulting in improvement of both subjective and objective experiences for the listeners of tape recordings.
One real problem with measuring various changes in audio quality and attempting to both attribute them to specific causes and simultaneously predict how they will be perceived is that – in the first place - we often don’t know exactly what to measure. Of course, we know the basics such as amplitude response versus frequency, phase response, distortion in its various forms and the like.
But it’s exceedingly difficult to get detailed measurements with real source material in place of standard testing signals. (Meyer Sound) SIM and (Rational Acoustics) Smaart are measurement tools in this direction, and they’ve greatly benefited sound reinforcement.
At the same time, there is no solid standard for transient response measurements and the resulting perceived effects. Several manufacturers claim that by extending frequency response of a system well past the “audible” limit (say, to 50 kHz) and maintaining phase accuracy through that range, that transient response and distortion will be improved in the audible band.
But even so, is this necessarily the way to predict that the system will sound good? Perhaps it could be argued that all other things being equal between two systems, the one with the lower distortion will “sound better.”
But then again, an interesting experiment done long ago by Bell Labs resulted in the conclusion that for a limited-bandwidth system, the one with more distortion was perceived as sounding “better.”
Perhaps this is one way to explain why low-power, all-tube, all-Class-A amplifiers are often perceived to sound more “musical” than huge, solid-state, “mega-kilowatt,” machined-aluminum monsters that are competing for the same piles of money.
Or maybe it’s other, psychological factors, such as the idea that tube amplifiers replaced the hearth in the home as a centerpiece around which to congregate…
Or perhaps it’s a result of something that is more easily quantified.
Class-A amplifiers distort differently from other designs. Not only this, but by running “wide open” in some cases, there’s more power available for short-term small-scale dynamic changes such as transient information.
It can be easily shown that although two systems may have the same signal-to-noise ratio and the same distortion figures on an analyzer, they sound radically different. The spectra of the noise, and the character of the distortion, play huge roles in perceived sound quality.
So again, the challenging question about quantifying performance in audio systems is what to measure in the first place, and how to measure it.
The bottom line is that both camps have something very important to offer. Without a scientific approach, we’d be stabbing in the dark trying to find solutions to problems about which we know very little.
But without a reliance on the subjective experience, even our most clever inventions would perhaps never reach the level of “art.” What good can come of setting fire to a silk-screened portrait of Andy Warhol in the middle of the woods if there’s no one present to snicker?
Designers and sound system users make decisions every day based on whatever they have at their disposal, including theory, available equipment, testing and measurement, intuition, and finally, critical listening. If there is not a balance among these resources, the results are likely to be unbalanced.
How would you like some power amps with “DC to light” response but producing crappy sound? Care for some loudspeakers that sound amazing but look like a “Dogs Playing Poker” on black velvet? How about mics that can pick up a gnat burping but make a Stradivarius sound like a banjo bowed with rosined fishing line?
Let’s leave it to the great Duke Ellington: “If it sounds good, it is good.”
Karl Winkler serves as vice president of sales/service at Lectrosonics and has worked in professional audio for more than 25 years.
Thursday, October 06, 2016
Audient Heritage Edition Console Chosen For Oslo Konserthus
Concert Hall in Norway’s capital outfitted with largest ASP8024-HE Heritage Edition analog console to date.
Just when we thought we’d seen the largest console to leave the Audient factory, the Concert Hall in Norway’s capital went and ordered an even bigger one.
Delivered in September, the 48 channel ASP8024-HE Heritage Edition comes with all the trimmings, including patchbay, Dual Layer Control (DLC) and producers’ desk. With nine bays, it’s officially the largest Heritage Edition ever made. Luckily, Oslo’s Concert Hall is a big place.
“We were looking for an analog console that doesn’t lie,” says production manager, Jan Olsen Skare explaining their decision. “Obviously the room has to ‘help’ the music, but we wanted the console to have a true dynamic and good preamps.” When introduced to the Audient brand by Norwegian distributor, Prolyd, he wasn’t disappointed. “The mic pres are so quiet and respond amazingly. The EQs are just wonderful and very accurate; I’ve never heard anything sound this good.”
Launched earlier this year, Heritage Edition is billed as the ‘definitive version’ of console designer, David Dearden’s classic ASP8024 design, which boasts a new look as well as an array of added enhancements ‘under the hood’. “We’ve done a few test recordings and we are very happy. Listening in the control room, we ‘hear’ the room as we know it.” Skare has surprised himself too, a self-confessed “analog guy of the 70s”, by admitting to “...rather liking the dual layer part.”
The brand new desk from the British manufacturer is located in a control room which will not only used to record shows, but also as a self-contained recording studio. “We’ll be recording classical orchestra so we’ll need all those channels,” says Skare. Oslo Concert Hall aims to be one of the premier music venues in Norway, presenting more than 300 events a year and receiving over 200,000 visitors.
Now it’s installed, the brand new Audient desk is ready to be put to work with some of the top Norwegian and international artists from a broad range of genres who are on the Concert Hall bill in the coming months.
Tuesday, October 04, 2016
BAE Audio Releases G10 500 Series 10-Band Graphic EQ
New 500 series module offers transformer-balanced signal path, 10-band graphic EQ configuration, and 2520-style op-amps.
BAE Audio has announced the launch of its new G10 equalizer.
The unit, first unveiled at the 139th AES Convention, adds to BAE Audios growing 500 series offerings with a punchy, transformer-balanced signal path, versatile 10-band graphic EQ configuration, and 2520-style op-amps. This versatile studio tool has applications from tracking to mixing to mastering, its tone shaping capabilities for tweaking drum or guitar sounds or sweetening an entire mix.
With 10 carefully selected bands offering up to 12 dB of boost or cut on tap, the G10 offers a level of tone sculpting that can help any audio sit perfectly in the mix. The easy-to-use slider-based interface helps users intuitively visualize the EQ curves they are applying. Switchable high-pass and low-pass filters, tuned at 80 Hz and 12 kHz respectively, help make the G10 a truly complete sound shaping solution.
“10-band graphic EQs are a popular studio tool because of their ability to sculpt your sound precisely so that it fits just right in your mix,” says BAE Audio CEO Mark Loughman. “With the G10 we’re providing that functionality with BAE’s trademark superior components and build quality plus an innovative implementation of input and output transformers in the path. As a result, the G10 is both versatile and great sounding and will shine on any source in any stage of the recording process.”
The G10 features input and output transformers that imbue any audio passing through it with a unique tonal character. “The transformers give the G10 an incredibly tight sound that has plenty of punch without harshness,” Loughman says. “It has a color and personality to it all its own and provides a nice counterpoint to the British-designed BAE EQs that you probably already have in your lunchbox.”
The G10 is shipping now and is available through authorized BAE Audio dealers.
Posted by House Editor on 10/04 at 10:23 AM
Wednesday, September 28, 2016
Church Sound: Using EQ The Right Way
Equalizers, by their nature and name, are supposed to redress imbalances in a sound system.
Unfortunately, equalizers, whether graphic, parametric, or shelving in nature, are only as good as the person using them. Knowing where to turn the knob or push the fader is usually a dark science, but there is a light at the end of the tunnel.
The first step in proper EQ technique is to realize the knobs go to the left better than they go to the right (or in the case of faders, down rather than up).
In other words, reduction of energy is more beneficial than an increase in energy, due to the nature of live sound. Live music in particular is awash in competing frequencies from melody and rhythm instruments in struggle with cascading voices emanating from the stage and the audience.
Since human hearing is most sensitive to frequencies in the mid-band and most instruments produce a majority of their fundamental and initial order harmonics in the same region, there is too much energy localized between 250 Hz and 2.5 kHz.
Again, though, without knowing where to cut the energy levels, the cure will be worse than the disease. Here, then, are several descriptive words and the corresponding frequency most attuned to the problem area:
Tubby – 125 Hz is a common resonant frequency in church auditoriums. Stage rumble can be reduced by eliminating (-9 dB) the frequencies below 125 Hz on the graphic EQ assigned to the main stage monitors.
Boomy – 160 Hz is “fake bass,” a tone associated with a wimpy boom box trying to reproduce the solid fundamental tone of kick drum and bass guitar that occur at 80 Hz. Because an octave is a doubling or halving of frequency, boom-bass is an octave above real bass. Pulling out 5 dB at 160 Hz will clear up the rhythm section.
Muddy – 250 Hz is the most effective region to grab and reduce “masked” energy from the sound system. If you can do one thing, reduce 250 Hz by 3 dB.
Boxy – 400 Hz is the fundamental tone of most male and some female voices. Too much of it can trap the energy into a localized source without any ‘air” or ‘breadth.”
Honky – 800 Hz is the leading culprit behind exquisite grand pianos that sound like a cheap honky-tonk upright through the PA. The pianist will thank you for removing 4-6 dB in this region.
Nasal – 1000 Hz (1 kHz) is the nasty tone in some reproduced lyrical vocals as well as the tone that gives away a digital piano as an unreal instrument. Take out 3 dB and watch the reaction.
Vox Box – 2 kHz is the chosen pre-emphasized frequency for many stage grade vocal mics. While it allows the vocal to “cut through” the mix, too much of it can spoil a good thing. It’s already boosted in the mic; don’t add to it.
Fatigue – 3.5 kHz is the most sensitive region of the human hearing system. Unfortunately, many audio engineers have significant hearing loss around this zone, so they compensate with too much of too much. If your ears tire of the mix after 30 minutes, it’s time to pull back 4 dB here.
Sizzling – 5 kHz is a wonderfully definitive region, but too much on hi-hat and snare can reduce the audience to mush. Super cardioid condenser mics tend to emphasize 5 kHz and produce a “harsh” tonality. However, much more than a 3 dB cut risks losing the vocal articulation.
Zingers – 8 lHz is a necessary part of music, but it removes heads when it oscillates into feedback, so never add it, just subtract its decade differential at 800 Hz.
In fact, any needed boost can usually be found by reducing the frequency ten times less or more than its value. To increase clarity at 4 kHz, reduce 400 Hz by 4 dB. To get the vocals to cut through at 2 kHz, remove energy at 200 Hz. For pounding kick drum at 60 Hz, reduce 600 Hz.
As with everything in audio, these tips are just a guideline. However, they have proven effective in hundreds of live concerts and thousands of worship services, so feel free to start turning, as long as it’s to the left.
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.
Monday, September 26, 2016
Piper Payne Masters With Manley
Engineer from Coast Mastering selects Mini Massive stereo EQ and Manley SLAM! Mastering Edition limiter for a variety of projects.
Coast Mastering‘s Piper Payne was mentored by pros like Bob Katz, Thor Legvold, and Michael Romanowski, Payne has earned a notable reputation for superb ears, supported by her deep experience in many musical genres.
She works in a precision acoustical environment, with a meticulously selected signal chain of exotic audio gear including products from Manley Labs.
“With mastering, you cannot skimp on quality,” Payne insists. “You don’t get into this business to get rich; we’re in it to make quality music.”
Payne compares her philosophy to that of the manufacturer of some of her favorite audio gear. “It’s like the way Manley has worked for a long time to create the best quality products that they possibly can,” Payne muses. “They do it with the intention that somebody will use Manley gear to make a great record.”
Indeed, Payne has mastered many fine recordings with Manley gear, including the Madame Gandhi and Basement EPs, and the Headlander soundtrack. “The first piece of mastering outboard analog gear I ever bought was a Mini Massive stereo EQ from Manley, and it’s still the first thing in my signal chain. My new Manley SLAM! Mastering Edition limiter, which I totally love, is now the last thing in my chain.”
Coast Mastering also has a Massive Passive stereo EQ that Payne occasionally chooses instead of her beloved Mini Massive.
“I like the Massive Passive a lot,” she confirms. “It feels very flat in a good way; I know what I’m going to get out of it. It’s a very musical EQ, and I really like the shelves. The bottom end feels so accurate that I can easily figure out what’s going on, and the top end provides that ‘air.’ No other EQ does all that-except the Mini Massive.”
Payne is a confessed purist when it comes to audio quality. “I spent a lot of my early career doing classical recording,” she recalls. “My degrees are in sound, and my focus in school and right after school was entirely classical recording. I love classical but I also love Top 40 music, hip-hop, rock, jazz, and other genres. In the classical world, I loved the quality we could achieve by doing it right and not scrimping on any part of the process. I’m taking that ‘quality first’ approach with pop and rock.”
Even high-end gear sometimes falls short in Payne’s demanding environment. “I had a mastering chain that I was happy with,” she recalls. “I had a compressor/ limiter at the end of the chain and a good compressor in front of it. But when I pushed the compressor/limiter, it started breaking up; it couldn’t handle heavy guitar music and sometimes classical. I started looking for another option, and I wanted a couple of different types of limiters, so I called EveAnna Manley. She recommended the SLAM! because it gives you two different flavors of limiters. She sent me a SLAM! to check out, and it blew me away from the first second I listened to it.”
When she received the SLAM! Mastering Edition, Payne conducted two different mastering tests: one through the chain as she had it before and one with the SLAM! “It was like night and day,” she recalls. “I noticed the difference immediately in the lack of breakup or distortion. Then I A/B’d for a long time. I thought my chain was good, and my clients were happy, but there was always something I was working too hard for. In a way, I didn’t realize I had a problem until the SLAM! fixed the problem.”
The SLAM! Mastering Version also offers other advantages. “It allows me to make things sound big and impressive but without that nasty breakup you get when something is too loud,” specifies Payne. “For that, I need a versatile limiter to complement the rest of the chain, and I need something that’s going to deliver all that and make it feel big, impressive, awesome. And it has to be able to work on every kind of music. I’ve put classical records through the SLAM!, spoken word, massive pop punk, metal, jazz, solo guitar-everything works. I haven’t found anything that it doesn’t make better.”
Another consideration is predictability. “Some limiters sound cool but I have no idea what they’re really doing,” she admits. “I have to turn the knobs until it sounds good. I can’t think ahead, dial in a timing, and then replicate it; you have to be able to get back to that setting or darn close. The Manley SLAM! Mastering Edition’s stepped switches solve that. I like to get really fine but Manley makes intelligent musical decisions about what those notches are for and where they live, and the unit has been tuned properly. With a musically designed piece of gear like the SLAM! Mastering Edition, half a dB is enough of a change for me.”
Posted by House Editor on 09/26 at 07:03 AM
Friday, September 23, 2016
Recording The Masters Launches Kerwax Replica
New analog tube processor is a 2-channel excerpt of the 24-channel custom tube mixer installed at Kerwax recording studio
The Kerwax Replica analog tube processor has been designed at Kerwax Recording Studio, a residential facility located in Brittany, France. The Kerwax Replica is developed by Recording The Masters factory to distribute it as a worldwide product.
Its unique design features easy change and combination of tubes, to sculpt and color the sound artistically, with either vintage or modern tubes. The controls interact with each other to create a wide variety of tube harmonic distortion characteristics.
The Kerwax Replica is a 2-channel excerpt of the 24-channel custom tube mixer installed at Kerwax recording studio, a unique mixing desk conceived in the pure tradition of historical studios to meet the requirements of in-house sound engineers and producers. Its 2 independent channels are ideal for stem or individual instrument processing.
The Kerwax Replica creates a wide variety of tube harmonic distortion characteristics, with an easy tweaking combination of gain drive and output volume stages, bias adjustment and EQ knobs. It is the ideal companion as an insert or a front-end unit for processing and warming up your stems or mix.
Now, you can build your distinctive sound, and stand out in the digital crowd. You can easily warm-up and add distortion to any input audio signal: use trim and bias settings to adjust depth of sound and produce harmonics, and select drive function with gain setting to add progressive saturation to the sound.
Thanks to its gentle Baxandall curves, the integrated treble and bass EQs are very musical and smooth. In case you need to remove some harmful low frequencies, a high-pass filter allows you to choose between 80 Hz and 120 Hz frequencies.
The Kerwax Replica uses two vacuum tubes per channel: the first one for preamplification, the other for saturation. We selected 12AX7 reference because it offers the best audio results for a wide variety of sources. However, if you want to experiment and obtain different results, the tubes are replaceable. You can use all references from the 12AX7 family.
Available for pre-order from September 15th 2016, Kerwax Replica will be showcased and demonstrated in exclusivity by its designer Christophe Chavanon on Recording The Masters’ booth during next AES show in Los Angeles (September 29th - October 1st). Recording The Masters will be represented on booth #526 by experts and US representatives (RMGI USA) to discuss together about analog recording, tapes and machines.
Recording The Masters
Posted by House Editor on 09/23 at 08:01 AM
Wednesday, September 21, 2016
What To Look For When Purchasing A Digital Mixer
With the plethora of new (and affordable) digital mixers on the market, it’s easy to be overwhelmed with features, options and pricing.
So how do you determine the right board for your situation? I use the same criteria as purchasing an analog console with just a few additional twists.
When purchasing an analog console, there are five primary aspects that I suggest looking at. (Well, six if you include price.)
1) Quality. I’m a big fan of “touching and feeling” a console. Being a tactile person, the overall construction quality can be useful in terms of determining reliability and lifespan. I’m also a big fan of listening before purchase. Some of the apparently well-built consoles have sounded just as rugged as they look! Touch and listen carefully.
2) Reputation. With the internet, it’s pretty easy to find out about a company’s reputation, which is a good way to start further informing your decision. I also rely heavily on the peer network I’ve developed over the years—the opinion of my colleagues matters greatly to me. If you’re newer to the craft, tap into some seasoned veterans via product reviews and by posting questions for the community on the PSW Church Sound Forum.
3) I/O. How many inputs and outputs do you need today? And how many do you anticipate needing in the future? My rule is to add at least 25 percent more to what you are currently using. For example, say you’re currently using 15 inputs and 4 aux/monitors. Look at the next step up from a 16-channel board, which is most likely a 24-channel board with at least 6 aux/monitors. A word of caution: not all manufacturers count channels the same way. My own methodology is to count the number of channels with preamps and consider that the total.
4) Busing. This ties closely with I/O when talking about the number of aux sends available. There should also be focus on what other outputs a console offers (i.e., matrix, control room), and what type of routing is available. Are there sub groups? Can I use an insert on a sub group? And so on…
5) EQ. How robust is the EQ section? Are there sweepable EQs? Is there a Q knob? Is there high-pass filtering? Also, listen to the EQ when making changes: does it sound responsive? Does it seem to overly color the sound?
I/O configuration. Digital consoles usually offer a digital snake and more I/O than analog models. The challenge is to figure out, physically, where you need the I/O. Does the console itself have limited I/O? Will you need to add a local input rack? How does the digital snake connect to the console? What configuration do the stage boxes come in, and where do you need to place them?
iPad/remote mixing integration. The majority of digital consoles now offer iPad remote mixing integration. This is an incredible tool! However, look carefully at how it integrates, taking into consideration things like the need to add a host computer or a router. This will increase cost and complexity.
I also wouldn’t get overly hung up on the actual app (unless you’re going to mix exclusively from an iPad). There are many scenarios where the iPad option is really helpful, such as walking around the room and checking the mix, as well as setting monitors while standing on stage.
But I don’t see many scenarios with a live band where I’d want to do the entire mix on an iPad because there’s just not enough surface area to feel comfortable (at least for me). That said, I’ve been mixing exclusively on consoles for decades, and with that experience comes the comfort factor of doing what you know.
Personal monitoring options. With the popularity of personal monitor mixing, it’s a good idea to find out options offered by digital console manufacturers, and also, to look at integrating third-party solutions such as Aviom.
Some console makers have chosen to use their own proprietary digital bus, so be aware that your existing personal monitoring system might not interface with your new digital console in a very elegant way. And if you don’t have a personal monitoring system, it’s still a good idea to understand future options.
Storage/recall/presets. Evaluate how the presets/snapshots work. Are they global, or can you recall individual channel settings separate from a global preset? There are several options of how this is done.
A friend who is new to the world of mixing purchased a board that gave him suggested settings for different types of inputs. He starts by simply recalling (per channel) the suggested setting for a particular instrument or vocalist. He loves this feature because it gives him a great starting point.
Recording. What type of recording software is available? Can you record multi-track? Is there a USB option that allows recording stereo right to a USB drive?
Onboard effects. I suggest not only evaluating the available effects, but also the EQ and dynamics sections (compression/gate) sections. Find out exactly how many things can be used at the same time. Some boards have limited processing.
Best wishes on selecting your new digital mixing console—mixing has never been more fun and never have so many tools been available in a console that most can afford.
Gary Zandstra has worked in church production and as an AV systems integrator for more than 35 years. He’s also contributed numerous articles to ProSoundWeb over the past decade.
Tuesday, September 13, 2016
RE/P Files: An Interview With The 1971 NARAS Engineer Of The Year Roy Halee
From the November/December 1971 issue of the late, great Recording Engineer/Producer (RE/P) magazine, this feature is an in depth look back at the career of a legendary engineer.
When a child is learning to walk, he is able to do no more than put one foot in front of the other and shift his weight.
He learns quickly (after a couple of falls), that he must master these basics before he can advance to running, skipping, or dancing.
His thoughts may not be quite so complex as we seem to imply, but the fact remains that for the time, he can only do one thing, and that without much skill: he can walk.
After several years, he becomes adept to any number of methods of getting himself from place to place.
He may walk, skip, run, dance, or whatever. It all depends on the situation he finds himself in. Further, if he’s been aware of his learning process, he knows he’s gotten beyond his walking stage by experimenting, by playing around with his balance and coordination.
Great dancers are great experimenters. They’ve discovered that they have to go beyond just walking in order to fully express themselves, and to respond artfully to the music to which they dance.
We think the analogy is not too strained when we compare the artistry of a great dancer to the artistry of a great recording engineer.
Such an engineer is beyond the elementary repetition of, “It worked then, and it’ll work now. Why take chances?”, just as the dancer is beyond carefully putting one foot in front of the other and merely walking.
The techniques of the engineer and dancer are always growing, changing, expanding, in order to better express the music and feeling they deal with daily.
Roy Halee dances with his fingers. For his artistry and technique as an engineer, he was awarded a Grammy and the title Engineer of the Year for 1971 by the National Association of Recording Arts and Sciences.
He is the man-behind-the-scenes, engineer, friend, and cohort of Simon and Garfunkel. The classic album “Bridge Over Troubled Water” is the product of his engineering skill combined with the musical genius of Paul Simon and Art Garfunkel. A quick listen to the album will show that its greatness is not music alone.
Columbia has been giving out gold records for only two years, but Halee already has 16 years of them.
His association with Simon and Garfunkel began just as he graduated from Columbia’s editing room to the studio. He was starting as a recording engineer when Simon and Garfunkel arrived for their first audition.
He engineered, they played, and that first audition became their first album, “Wednesday Morning, 3 A.M.” He’s been with them ever since.
Just as the dancer must obey the law of gravity, there are certain limits that Halee must work within. But those limits are becoming frayed and dented by his insistent forays against them.
Halee says it succinctly: “I don’t like to make hard and fast rules. When experimentation goes out the window, new sounds go out the window.”
It’s hard to pin the man down on exact and repeat- able techniques; he’s always changing. We managed however, to get some insight on him, and the contribution he’s made to engineering artistry.
It is well known that Halee gets tasteful though unusual sounds from his drums.
He normally mikes them with a U-47 overall, snare top and bottom with a salt shaker or RE-20, floor torn often top and bottom, a mike over the sock cymbal (high hat), high enough to get some splash from the snare, and the bass drum front and back.
The second mike on the snare allows him to get a bit of crack without over eq’ing, a technique that is especially effective on the louder rock dates. Double miking of the other drums allows similar effects to be employed.
Phasing seems generally not to be a problem, but readers are cautioned when employing such a technique to be certain that any double miking done is constructive, both electrically and acoustically. Halee is unhappy with copying for the sake of copying. As he puts it:
“There was the day Halee put a drummer next to the elevator shaft.”
“Some people think, ‘That record was successful, so I’ll always mike drums the way I did [or Halee did] on that date.’ But that was that day, that temperature, that studio, and it was Hal Blaine’s set of drums. Tomorrow you’ve got another drummer coming in with his set of drums and the humidity is 80%.”
The point is well-taken. Creativity is most productive when one first recognizes exactly what he is dealing with, and then builds from there.
The creative building of tracks takes strange forms on occasion. There was the day Halee put a drummer next to an elevator shaft,
“We wanted to get an explosion effect, so I put the guy out in the hall next to an elevator at 49 East 52nd street in New York. The hallway itself was extremely live, so I put mikes in the shaft and in the hall, and limited the hell out of them. And we got an explosion sound. It’s in ‘The Boxer’.”
In “Bridge Over Troubled Water” there is a snapping sound, like a whip in the distance. It was created by physically placing the drummer inside an echo chamber.
The willingness to seek unusual methods and sounds is certainly worthwhile, but to be effective, it must be coupled with a more gut feeling for the effect of music on a listener.
When we asked Halee about his use of stereo spread on drum and piano mikes, his response was typically non-commital, but at the same time clear.
“The degree of stereo spread I use depends on the piece. If it’s disconcerting to the song and music, I won’t do it. A lot of times I put drums on one track. It depends. If you’re doing a thing like “Cecilia”, where you want it to dance around, it adds to the arrangement.”
“And generally, when I mike the drums and split them into stereo, I won’t split them extreme left and right. I’ll split them left-center, right-center, and center.”
“If ... the tune elevates itself, or picks up, I will sometimes pan the drums extreme left and right to give it more motion.”
“If a tune calls for a lift, as when it goes into the waltz section, or some such thing where the tune elevates itself or picks up, I will sometimes pan the drums extreme left and right, to give it more motion.
You won’t even be aware that it happened unless you have headphones on. But it does create the effect of lifting that particular section of the tune. Then I’ll bring it back again.”
Simon and Garfunkel’s first album was guitar and voice. To this day, though other instruments overwhelm the guitar in much of their music, it is still given as much attention as ever, particularly in terms of how it is miked.
The exact configuration is dependent on what kind of guitar it is, how it is being played, and whether it is being finger picked, strummed, or flat picked. When a flat pick is used, Halee likes to stay away from a condenser mike, and uses a dynamic mike instead. Otherwise, one, two, or three condensers are often employed.
Normally, two is the number, one at an angle over the hole (to prevent the hand coming between the hole and the mike), and one down over the guitarist’s right side, behind the hand.
When Halee does other stringed instruments , he invariably uses condensers, notably a U-87, 67, or M-49. He’s made the comment that he likes a “wall” of strings. How does he accomplish this?
“I try to use more than one track, like for violins, so I can spread, and the same for low strings. Instead of putting all the violins on one track, and viola and eel I i on another, I try to use a lot of tracks.”
“For eight violins I’d use two mikes. Again it depends on what you’re doing. If it’s a hard rock date, where you can’t get far away because of leakage problems, I’ll mike every two players. But on overdubs, I use an average of two mikes; with eight violins, one on the front four and one on the back four. If there are two violas and two celli, I’ll put a mike on the violas and a mike on the celli.”
“That’s why I got them to put this mixer in this console [A small mixer, independent of the standard console inputs, is mounted on the right hand side of the console].
If I had strings on a hard rock date, I might mike every two violinists, as I said, to get a lot of presence on the string-., bring them up and mix them on this mixer, and take thorn all in on one channel. All eight mikes.”
“Then again, there’ve been occasions where I’ve used one mike on twelve violins, and I’d put it far away. But in a room where the air conditioning and rumble weren’t ridiculous.”
“I like to create a lot of crazy rhythms.”
“I mult the strings, and I usually use a little tape reverb. Just a little bit. Then afterwards in the mix. It depends, again, on what the mix is, what the tune is, what the arrangement is, whether they’re going to have echo or be dry, whether they’re going to have delayed echo. But I always put a little tape reverb on strings.”
Piano is approached according to the nature of the piece. In “Bridge Over Troubled Water” (the song), the piano was miked as would be a classical piano solo.
Three mikes were placed high and back, about six feet horizontally and six feet vertically. Condenser microphones were employed.
On a rock date however, the dynamics and presence of the piano are altered considerably, and dynamic microphones, with an occasional condenser, are placed in tight.
Listening to tracks Halee has done immediately impresses the listener that considerable innovation and work have gone into their making. Technique follows technique, building a complex stereo matte of interweaving and overlaid effects.
“Through the years I’ve put guitars in bathrooms, drums in bathrooms. Sometimes I phase the echo. You don’t know it’s being phased, but it is. I did that on “At The Zoo.”
I’ve put a couple of choruses of voices inside a bathroom or an echo chamber. Sometimes I put Dolbies in when I record and then take them out when I mix. I use a lot of tape reverb. Sometimes we use it to create our own rhythms. When I say “our”, I mean Simon and Garfunkel. We create our own rhythms in the mix.”
“That happened, for instance, on “Bridge Over Troubled Water” Hal Blaine is playing the bass drum, but he’s not playing the part you hear. There’s tape reverb on that bass drum.
Oddly enough, when we did it, we found that it would only work with a Scully four track. If we used another machine, because of the head distance, it was out of rhythm.”
“When you hear ‘Ba da Ba da da da da’, that’s not what he’s playing. He’s playing something like ‘Ba . . . da da’. Something like that. I’d have to listen to the original tape to get the exact figure he plays, but what you hear is different.”
“I do an awfully lot of that. I like to create a lot of crazy rhythms. A lot of times it’s out of rhythm, and you strike out. But I like to fool around and find out when it’s in rhythm, and then if it’s good, I’ll use it.
I’ll flip it in for a couple of bars, and take it out. Sometimes I’ll program it to another track, so it’ll answer itself.”
Halee has been asked to remix “Bridge Over Troubled Water” for quad, but so far he has been reluctant to do so. This might seem out of character for a man so willing to experiment and try new techniques, but Halee feels the quality of quad is not up to par yet.
Further, he says he’s like to get into miking and recording in quad, rather than just remixing.
“I don’t care for drums in back of me, or swishing around. There are phasing problems.”
It seems to me the direction everybody is going now is to have completely isolated tracks, so you can place things in definite positions. I’d like to get into room sound—dimension rather than direction. I feel a lot more experimentation is in order. Until I’ve done it, I’m not going to get into remixing old Simon and Garfunkel tunes. I want it to be good.”
What about the controversy over more tracks, 24, 32, or more?
“The more the merrier,” says Halee.
The technology doesn’t seem to frighten him. He plays with it until he knows it well enough to master it. Like a dancer, he finds new movements through experimentation, and learns them well before performing them for the public.
The spirit, coordination, and balance are all there. The engineer is an artist. Roy Halee dances with his fingers.
Take the PSW Photo Gallery Tour of audio equipment ads appearing in RE/P magazine, circa 1970
Editor’s Note: This is a series of articles from Recording Engineer/Producer (RE/P) magazine, which began publishing in 1970 under the direction of Publisher/Editor Martin Gallay. After a great run, RE/P ceased publishing in the early 1990s, yet its content is still much revered in the professional audio community. RE/P also published the first issues of Live Sound International magazine as a quarterly supplement, beginning in the late 1980s, and LSI has grown to a monthly publication that continues to thrive to this day.
Our sincere thanks to Mark Gander of JBL Professional for his considerable support on this archive project.
Monday, September 12, 2016
In The Studio: Control Room Techniques To Foster Great Vocals
During a session, I remember when an artist was on mic, out in the studio ready to start vocal overdubs, and the producer asked: “How do we look in here from out there?”
Interesting, because he knew the appearance of the control room to the artist might affect the vocal performance. The control room (from the studio) does look like an aquarium with the huge window and the silent action of the animals encased within it.
Reactions to performances reflected in facial expressions and body language are everything to singers and musicians isolated out in the studio. The concern is that the working in the studio does not feel like being in a Petri dish under the microscopic scrutiny of the control room.
A great vocal sound starts with a good singer who has the artistic goal to perform the best vocal possible. Control room personnel—producer, engineer, assistants, gofers, etc., all have a professional responsibility to work towards the pursuit of the artist’s goals.
For the first hour of a new vocal session, everyone in the control room is on a kind of “audition” until the artist feels comfortable and performs well. It is the producer’s job to create the studio setting—the whole “vibe” to help get a good vocal performance from the artist and to make everyone else produce their best work.
During a vocal session, the producer is the arbiter of the feeling and quality of the vocal performance. The producer is the artist’s confidant, coach, good friend, creative partner, mentor, and most importantly the de facto proto audience - the first public ears on the artist and their music.
Because a well-prepared singer might give immediately the best and freshest performance in the beginning of the day, even during the microphone audition process, the engineer should be prepared and ready to record and capture a great vocal sound. The producer may require those mic audition recordings later and will be thankful for their useable fidelity.
The engineer’s vocal signal chain should be powered up, working and adjusted somewhere in the “ballpark”, the song booted up in the DAW with a new vocal track(s) ready to record, and a suitable monitor mix made and a usable cue mix done and checked in the singer’s headphones.
How Pro Can You Go?
Getting to know your favorite signal chain intimately is very useful for getting good vocal sounds quickly—especially in the case of the aforementioned first takes/microphone audition.
You need to know what different combinations of mic pre-amps, EQs and compressors produce in terms of vocal sound possibilities. Experiment often if time and your clientele’s interest permits.
I find the overarching difference between true pro gear and lower-end products is that professional gear is much more forgiving in it’s operational requirements than cheaper gear.
And that is not to say you cannot record usable sound using a $300 mic pre-amp versus a multi-thousand dollar boutique piece. You’ll have to work a lot harder to get a good sound with the low dollar gear and you can make just as crappy of a recording with either it or with the high-end boxes!
For example, pro gear usually has much more headroom, a lower noise floor and higher dynamic range. You’re less likely to overload the front end of pro mic pre-amps with a signal from a hot mic and a loud singer.
High-end pro gear is also smoother with less harmonic distortion at any operating level so the sound is automatically purer. Finally, pro gear will more often interface well, i.e., drive any subsequent processor you’d like in your vocal recording chain—from pro to junk.
I try to start with the best gear possible and I have my own collection to use when I’m “camping out” in a studio that does not offer my fave pieces. As an independent engineer, it’s a smart investment to own a high quality professional vocal recording chain you can use anywhere.
For more reasons that are not necessary to cover here, there are two schools of thought about the design philosophies and sound of mic pre-amps:
—“Super pristine” and transparent to convey accurately the microphone’s signal
—“Enhancement” - sonic embellishment through harmonic coloration and/or the inherent characteristics of non-linear, small signal amplifiers
Both types have their place in today’s recording but my preference for vocals (unless directed otherwise by the artist and producer) is for a transparent and clean signal chain. Some of my choices for clean, discrete transistorized mic pre-amps include the George Massenburg Labs GML 8302, Audio Engineering Associates Audio Engineering Associates RPQ, Avalon Design M5, and Millennia Media HV-3 and STT-1.
For tube-based pre-amps that can be operated in clean modes, I’ve always liked the Manley Mic/EQ-500 Combo, Groove Tube ViPre, DW Fearn VT-1, and old Telefunken V72 units. Tube mic pre-amps, by virtue of the tubes, have a built-in “personality.” They can be very clean but, when overdriven, get into coloration zones unique to each of them.
“Colorful” transistor microphone pre-amps I like are the old British Neve 1066, 1073 and 1084 modules for their thick-sounding Class-A design and line input, mic input and output transformers; for more punch and purity, I like the API (Automated Processes Inc.) model 512C amplifier for its Class-AB amp that gives a harder and “in your face” presence; the Helios Type 69 mic/EQ unit is ‘60s-era technology and a very “vibey” sounding unit also from England; and the Chandler Germanium, which uses esoteric transistors to produce its unique sound.
Mic and Signal Chain System
When deciding on a vocal recording chain, consider both the mic and mic preamp as a system. If you’re looking for a super warm and “tubey” sound, try using a tube condenser and a tube mic pre-amp.
Such was my choice for recording Rod Stewart on a couple of albums. He sounded best on a completely stock and original Neumann U67 tube condenser (no pad and no roll-off) into a tube-based Manley EQ500 Mic/EQ Combo that I followed with a TubeTech CL1-B compressor—a slightly colorful and all tube signal chain. This chain did not accentuate Rod’s raspy vocal quality we all love, yet it kept enough mid-range cut to compete with the track.
A much cleaner and more pristine path might be a Brauner Phanthera FET-based condenser microphone into a GML 8302 mic pre-amp followed by a dbx 160SL compressor that uses a VCA (voltage controlled amplifier) for nearly transparent gain control.
This signal chain would produce a more neutral or uncolored sound that is completely faithful to the source. I’ve found recording choirs with a super-clean chain like this reproduces the rich harmonic content in the best way.
I liked the Phanthera into a Neve 1073 module followed by an UA 1176LN (Rev D) limiter for recording Pat Benatar’s vocal. The Phanthera will handle all the loud level Patty can produce right on top of it without clipping.
The Neve 1073, like all old Neve modules, doesn’t sound good in clip and is a little unforgiving with regard to getting an exact gain setting so I set it a little low for the additional headroom.
After compression, I made up the record level within the very distinctive sounding 1176LN. Between the thickness of the Neve, the gritty edge of the 1176LN, and the pristine sound capture of the Brauner, this is a killer rock vocal sound signal chain.
EQ & Compressors
Generally adding equalization in the recording is to make up for what the microphone is not giving you. In some studios, there is not a big choice of mics so you have to add or carve out frequencies to try and mimic the sound you’d get automatically with the right mic. Along with a signal chain, owning a few classic vocal mics is an obvious asset for a recording engineer.
Again, unless requested by the producer or artist, I go very conservative when recording with EQ. For example, if you are adding a lot of low frequencies, there is something wrong with the microphone or the pre-amp or more likely the way the singer is addressing the mic.
If you’re finding that adding a lot of high frequencies sounds better then you’ve got the wrong mic, as if you were using an old RCA 77BX ribbon but really were looking for the ultra bright sound of a modern Sony C800G condenser.
The same goes for compression. There is a wealth of sonic possibilities using vocal compression especially with vintage classics like the Fairchild 670 limiter.
I love those sounds but when and how much depends very much on the “bigger picture” - the mix!
If you and/or the producer are unsure, compress only enough (at a low ratio) to get it recorded at a good level without distortion and errant peaks—and then back the compression down from there. For a “vibey” sound, go with a tube compressor like the TubeTech CL1-B or UA Teletronix LA-2 leveling amp.
Cleaner or more transparent compression comes from VCA-based units such as a dbx 165. You could also record the vocals on two tracks: one with compressor and the other without. I like to provide as many options for the mixer as possible.
(See Barry’s related article on this topic here.)
Barry Rudolph is a veteran LA-based recording engineer as well as a noted writer on recording topics. Be sure to visit his website.
Friday, September 09, 2016
Broadcast Devices Introduces 8/16 Series Passive Audio Switcher
New unit is designed to fill the requirement of preventing single points of failure in analog and digital audio systems.
Broadcast Devices (BDI) announces the releases a new product for the sound contractor and broadcast markets, the 8/16 Series passive audio switcher.
The 8/16 Series passive audio switcher is designed to fill the requirement of preventing single points of failure in analog and digital audio systems. This product can accept either 8 or 16 sets of (A/B) balanced pairs and route them to a common output.
Two models are available: the 8/16-8 8 input and the 8/16-16 16 input version.
Because the switching function is passive it can perform emergency switching for analog and digital audio signals and for other types of control pairs. The products intended uses are for failure bypass of routers, consoles, monitor systems or just about any application where multiple pairs need to be shifted from one input to another.
Interface is the standard Tascam 25 pin interface. Remote control interface is standard DB25 connection. Accessory DB25 to XLR and/or 75 Ohm BNC interfaces are available.
Four versions are available within the series. 8/16-8, 8/16-16 (8 and 16 channels respectively) and a set of dual power supply versions 8/16-8-D, 8/16-16-D.
Posted by House Editor on 09/09 at 09:51 AM
United Recording Launches Comprehensive Archiving Division
New head of archiving, Dan Johnson, brings extensive experience as a former recording engineer at Capitol Studios and Ocean Way.
United Recording of Hollywood, California has launched its new archiving division, it was announced by studio manager Robin Goodchild. United was founded in 1957 by recording engineer, studio designer and electronics inventor Bill Putnam and expert archiving is a key element of the studio’s heritage.
“United has a 60 year history of uncompromised audio excellence and innovation,” commented Goodchild. “We have assembled a vintage treasure trove of virtually all modern recording machine formats and the ancillary equipment crucial to accurate archiving to insure the new masters will be preserved for the ages.”
United’s new suite is a secure, climate-controlled suite that features such attention to detail as a specially built anti-static floor to prevent any electrical mishaps. A full-time dedicated maintenance staff means the gear is well cared for and running correctly at all times.
United’s new head of archiving, Dan Johnson, spent the past five years as a dedicated audio preservation engineer working with priceless masters by such artists as Led Zeppelin, Jimi Hendrix, The Doors, Eagles, Prince, Red Hot Chili Peppers, The Ramones, Van Halen, Rod Stewart and Otis Redding. Prior to that, Johnson was a recording engineer at Capitol Studios and Ocean Way (now United).
“I started my engineering career at United almost 20 years ago, and opening an audio archiving facility here is a timely decision,” commented Johnson. “The studio’s high standard of quality and excellence, as well as the commitment to an unparalleled legacy provided me with the foundation that I have built my career on. It’s good to be home.”
Archiving masters lowers insurance/storage costs by having digital back-ups, rescues audio from deteriorating tapes, and keeps assets viable after the tape becomes unplayable. Sending digital copies for mastering/mixing means no damage/mishaps to fragile tapes. Digital files can be electronically delivered, which saves on shipping and packing costs. Assets are always safe and available.
The archiving process involves tape condition being checked precisely and processed accordingly. Formats are correctly determined and documentation checked regarding speed, noise reduction, etc. All tape boxes, notes, and track sheets scanned at 300dpi. Tape preparation includes baking when necessary, as well as replacing damaged splices and bad leader tape. Multi-track tapes are transferred in real time and synchronized to Pro Tools. Final assembly of recorded assets are transferred to archival DVD or Blu-Ray discs and .wav files.
Thursday, September 08, 2016
Crane Song Debuts Updated Egret With Quantum DAC
Entire line of digital hardware products has been updated to take advantage of proprietary 5th generation digital to analog converter technology.
Crane Song (AES Booth #1123) announces that its entire line of digital hardware products has been updated to take advantage of its proprietary 5th generation digital to analog converter technology.
With its AES debut, the Egret 8 channel D/A converter / summing mixer joins the Avocet monitor controller, the HEDD 192 AD/DA converter, and Solaris stand alone digital to analog converter to complete the line up of Crane Song products equipped with Crane Song’s Quantum DAC.
The Quantum DAC uses a 32-bit converter and asynchronous sample rate conversion for jitter reduction with up sampling to 211 KHz. The reference clock uses a proprietary reconstruction filter for accurate time domain response; and with jitter less than 1pS.
“I have done several years of measurable analysis and subjective listening in the development of this technology; the Quantum series DAC is the most accurate that I have ever designed,” explains Crane Song founder and developer Dave Hill. “Typical jitter from 10Hz to 20 KHz from the internal clock is 0.055pS and from 1 Hz to 100KHz it is less than 1pS. The result is a very 3D sound that is exceptionally transparent and accurate.”
The Crane Song 5th generation Quantum DAC has been shipping in Avocet IIA since November, 2015, and in April, 2016 Crane Song quietly updated the HEDD 192. As of AES show the Egret will be shipping with the upgraded DAC. This completes the updating of the DACs in all Crane Song digital hardware.
Egret is a flexible digital audio workstation back end. It contains eight channels of Quantum D/A converters and a stereo line level mixer with color options to help bring analog summed digital mixes to life. Each channel of the stereo mixer has a level control, an aux send (which is post level control), a color control, and a pan control. Each channel also contains an analog / digital source button, and solo - mute buttons. The color function is adjustable from a transparent sound to a complex mix of second and third harmonic content, creating the possibility of having clean modern sounds mixed with vintage sounds.
DAC upgrades are available for previous generation Crane song digital hardware products.
See Crane Song at AES in Los Angeles at Booth # 1123 at the Los Angeles Convention Center, Sept 29-Oct 1, 2016.
Posted by House Editor on 09/08 at 09:31 AM
Recording Basics: Starting At The Source With Good Levels
There are a lot of things to focus on during a tracking session, especially when you’re recording a dozen or more inputs at once.
You want to make sure you’re getting a good sound from each microphone. That’s step one. Let’s be honest, you’ll spend the rest of your recording life perfecting step one…
For now, I want to focus on step two – getting good levels, both when you’re tracking and during mixing.
When I first started recording, I was taught that you want to get the level as close to peaking as humanly possible without going into the red.
I would keep cranking up the mic pre on the snare drum mic until it was pixels away from clipping.
What happened? Everything clipped, of course. Apparently musicians play louder during the actual take than they do during sound check.
So I would turn the preamps down a little bit. Everything looked good, then BAM! More clipping.
I would keep turning down the preamps little by little until the clipping stopped. By this time, the musicians are tired of me coming over the talkback and saying, “Whoops! Sorry guys, that clipped. Let’s start again.”
Not a good scenario.
The reason people tend to think that you need to really “peg” the meters is leftover from the analog days. The harder you hit tape, the better the recording would sound. If you had lower levels, the tape noise would become much too audible.
Today, however, just about everyone is recording a 24-bit digital signal. Digital signals don’t sound better when you turn them up, they simply get louder.
If you record the same track really close to the clip light and then again with plenty of headroom, you won’t notice a difference in the quality of the signal, only the volume.
Analog equipment tends to saturate and add color the harder you drive it. Digital systems do not.
What does this mean?
If you’re recording at 24-bit (and you should be), you’ve got a whopping 144 dB of signal to work with. What does that mean? The noise floor of your system is significantly lower than on an analog system. In fact, the noise floor of a decent digital system is virtually non-existent.
Let’s say you record a snare drum in Pro Tools, and its loudest part is 6 dB below clipping. So, you technically could have recorded it 6 dB louder, but even 6 dB down you still have 138 dB of signal left in your system. You’re still WAY above the noise floor.
My suggestion? Give yourself some room to breathe! Rather than trying to make the signal get as close to the top of the meter as possible, have it max out somewhere between one-half and three-fourths of the way up the meter.
This way the drummer can do an awesomely loud fill without clipping every track in the session, and you won’t end up smacking your forehead every time the clip light goes off during and awesome take.
So, what about mixing, anyway? What was your biggest issue when you mixed your first song? I bet you a nickel it was getting the levels right.
You probably got half-way through the mixing process, and suddenly several of your tracks are clipping, or your master fader is clipping.
So you turn the clipped tracks down a bit. Well now the mix doesn’t sound right, so you try to turn every other track down by the same amount. Still doesn’t sound right.
You go back to work, re-balancing everything. Before you know it, your tracks are clipping again.
You think to yourself, “Did I really turn these up that much again?” You slam your fists into your desk…or kick the dog…or yell at the cat…or maybe you do all of these at the same time.
Welcome to the world of mixing.
You are not alone. This was my experience, and I bet (another nickel) that if you asked any experienced engineer, he’d share a similar story of his own frustrated journey.
My Advice to You
This is the part where I could go into a list of techniques for keeping your track levels down to prevent clipping, but you know what? I’m not gonna do that.
Why? Because I think there is one single reason why people have such trouble with clipping during mixing. I’ll get to that in a second.
The first thing you can do to make your life easier as a mix engineer is to make sure you don’t record everything at a super-hot level. I talk about this in Setting Levels for Recording.
You don’t have to peg the meters to get a great-sounding recording. If you just get a decent level, you’ll be much better off when it comes time to mix.
Sometimes you don’t have control over the levels. Perhaps you’re only mixing the song, not recording it. If so, then you’re at the mercy of the recording engineer who recorded the tracks.
Okay, back to getting rid of all that nasty clipping on your tracks. My suggestion to you?
Turn up your monitors/headphones!
Seriously, this is the biggest reason I get clipping on my mixes, I keep the volume knob too low on my monitors and headphones.
Rather than turn up the monitor volume, I push up the track levels in my mix. That’s a recipe for failure. (It’s also a recipe that will produce more dog-kicking outbursts if you don’t fix it.)
Let’s say you’re mixing a rock tune, and you’re listening to just the drums. Before pushing the kick drum up to zero or (even worse) above zero, reach for the volume knob on your speakers or headphones instead.
You’ll be able to hear everything better, and your mixing levels will be well below clipping. Remember this next time you’re mixing. I bet (yet another nickel) it will help.
Joe Gilder is a Nashville based engineer, musician, and producer who also provides training and advice at the Home Studio Corner.