Article

Wednesday, February 08, 2012

How To Archive Multitrack DAW Recordings

The archived recordings must be prepared to weather obsolescence

 
Multitrack DAW recordings are dependent on a complex system of primary and secondary technologies.

As discussed in An Introduction to Archiving Music Recordings, each of these technologies represents an obstacle to the long-term viability of a multitrack archive.

Simply put, if the various software and hardware products you’re using today aren’t going to be around in their current versions for the useful life of the sound recordings you’re creating (i.e. the copyright term), the archived recording must be prepared to weather that obsolescence.

The goal of preparing multitrack DAW data for archive is to minimize the layers of technology necessary to completely reconstruct the master recording in the future.

This article will introduce some basic techniques for creating both Consolidated and Flat Multitracks for archival purposes.

What Is A Consolidated Multitrack?

A Consolidated Multitrack is a digital audio fileset that completely expresses the EDL (Edit Decision List) information from a multitrack master recording. Specifically:

—Each DAW track is expressed as a single, continuous Broadcast Wave file (BWF);
—All of the consolidated audio files share the same start times and durations;
—All of the consolidated audio files share the same digital audio precisions, i.e. sample rate and bit depth;
—All of the consolidated audio files share the same descriptive naming convention, e.g. trackname_songtitle_artistname.wav.

If all of the above specifications are met, a folder containing the consolidated audio files could be used to perfectly reconstruct the multitrack recording as far into the future as the Broadcast Wave file format remains viable.

Since the Broadcast Wave file is a widely accepted standard file format for media producers, its long-term viability (and eventual uniform migration) is virtually guaranteed.

Creating a Consolidated Multitrack:

1. From your last active session/project file, ‘Save As’ to create a discrete file from which you will create a Consolidated Multitrack.
2. Hide or delete any auxiliary signal path to simplify the working environment.
3. If additional Takes or Playlists are to be included in the Consolidated Multitrack, create new tracks to allow all of the source audio to be simultaneously visible/accessible.
4. Using session boundaries, location markers, or some other timeline tool, establish a repeatable global timeline selection that includes all audio from the earliest drop-in to beyond the longest running audio file.
5. Once your global selection is made, use the Consolidate or Merge functions to create a single continuous audio file that expresses the EDL information for each track.
6. Carefully, consistently label all of the newly consolidated audio files to reflect enough information that they could completely identify themselves by name, e.g. bassamp_take2_ohbabybaby_jimmysingsalot.wav

Once the above steps have been followed, a choice has to be made about how to present these consolidated audio files as a discrete multitrack recording for archive.

Minimally, a folder that follows the same naming convention as the consolidated audio files should be created to contain all of the associated audio files and metadata (like screen shots, rtf files containing session notes, credits, etc.). This method works fine, but will always require the multitrack to be reconstructed in a DAW for playback.

Alternately, a facility like Pro Tools’ ‘Save Session Copy’ could be used to create a new, independent playback session for only the archival material.

Using this method one would need to be careful to remove any non-archival audio and metadata from the source session before saving the copy.

This approach would facilitate more convenient short-term use of the archive, but doesn’t actually provide any additional content.

What Is A Flat Multitrack?

A Flat Multitrack is a digital audio fileset that completely expresses the EDL information from a multitrack master recording, but also expresses some subset of DAW metadata. What metadata is ‘flattened’ into the archive is up to you, your client, or contractual obligations, but it could include:

—Plug-in processing like amp simulation, ‘printed’ effects from auxiliary channels, or automated processing;
—Automation data, like the fader rides on a lead vocal track;
—Bounced submixes that would otherwise require reconstructing both complex routing and plugin processing.

It is critically important to note that a Flat Multitrack should never be archived instead of a Consolidated Multitrack, but only in addition. The Consolidated Multitrack is the master recording; the Flat Multitrack (when applicable) is an extension of that master.

Once a Consolidated Multitrack has been created, a Flat Multitrack can be created by repeating the process with a few additional steps:

1. From your last active session/project file, ‘Save As’ to create a discrete file from which you will create a Flat Multitrack.
2. Hide or delete all auxiliary signal path and metadata that is not going to be flattened.
3. If additional Takes or Playlists are to be included in the Flat Multitrack, create new tracks to allow all of the source audio to be simultaneously visible/accessible.
4. To flatten real-time processes like automation, time-based effects, or submixing, bounce/re-record the appropriate track outputs to new tracks, and remove the source tracks from the session. Note what metadata has been flattened.
5. Flatten additional metadata by processing audio files with offline versions of real-time plug-ins. Note what metadata has been flattened.
6. Make a global timeline selection, and use the Consolidate or Merge functions to create a single continuous audio file that expresses the EDL information for each track (including whatever metadata has been flattened into them).
7. Carefully, consistently label all of the newly consolidated audio files to reflect enough information that they could completely identify themselves by name, e.g. bassamp_take2_flatcompression_ohbabybaby_jimmysingsalot.wav

Since it would be unlikely that every track within a DAW project would have metadata worth flattening, there will likely be some tracks that remain in their consolidated form. I would caution that it would be both redundant and confusing to include these audio files in a Flat Multitrack archive.

Preferably, an additional folder of flattened audio files can be clearly labeled, and organized with the Consolidated Multitrack data. Future users can then reconstruct the Consolidated archive, and opt-in to any of the available, clearly labeled, flat content.

Contents Versus Carrier

It should be noted that this tutorial only addresses the form of the contents of a multitrack archive. The question of how to effectively store this information is an entirely additional- though related- matter.

Anybody who is serious about the subject should examine the Producer and Engineers Wings’ “Recommendation for Delivery of Recorded Music Projects” (pdf). It contains an example of a widely-adopted approach to redundant archival storage.

Rob Schlette is chief mastering engineer and owner of Anthem Mastering (anthemmastering.com) in St. Louis, MO, which provides trusted specialized mastering services to music clients across North America.

Be sure to visit the Pro Audio Files for more great recording content.

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Tuesday, February 07, 2012

In The Studio: Three Mid-Side Processing Tricks (Includes Audio Samples)

A form of processing on stereo sources for practical or creative effects

 
In this article I’ll explain how I use mid-side (MS) processing on stereo sources for practical or creative effects.

Mid-Side?
Two channels of audio can be combined in a way that gives us control over what is the same in each signal, the middle, and what is different, the sides.

The middle is where the kick drum, snare, bass, vocals and a lot of other instruments are, the sides have any hard-panned instruments and spatial effects like reverb.

It can be pretty interesting to listen to music like this, there can be a lot hidden in the side channel.

MS is also a stereo microphone technique using a cardioid microphone facing the source and a bidirectional mic turned 90 degrees away just picking up ambience.

In this situation the two signals would need to be decoded into stereo. The side mic signal is duplicated, polarity inverted and the two side signals are then panned hard left and right. This is not a true stereo mic technique but can sound very nice. The balance of mid and side signals can be adjusted as needed by changing the level of the three tracks.

You can manually encode and decode stereo files to MS and use mono plugins to process mid or side individually. A lot more plugins have an MS mode now. Many of the modules in the T-Racks suite allow mid side processing, as does Ozone, a few compressors and equalizers and a distortion also come to mind.

You can do this for subtle or crazy effects, its a fun way to experiment with plugins and get some unique sounds.

Loud & Wide
For a recent mastering job I used a Fairchild compressor plugin in MS mode (Lat/Vert) to compress the middle and increase the level of the sides. I did this in parallel so I could blend the effect in easily. I was also using this to get a lot of extra loudness. You can call this parallel MS compression.

Compare -

The master without parallel MS compression: listen

With parallel MS compression: listen

With parallel compression soloed: listen

No More Messy Verb
Someone asked ma about clearing up the middle of a mix when using a lot of reverb. Using MS compression on the reverb return can work well. Compress the middle more than the sides and increase the side volume if you want more width.

Here is an example of that on some drums - Steven Slate playing in KONTAKT. The whole kit is sent into Valhalla Room. With the Fairchild after the reverb I’m lowering the middle by 2 dB and raising the sides by 2 dB.

Here is this effect with lots of reverb on the drums: listen

And now with MS compression on just the reverb bus: listen

There is NO compression on the drums themselves, I’m only compressing the reverb return and widening it.

Wacky Effects
Here is an example of what you can do with a stereo loop and any plug-in. This is a little more complicated, and only works if there are hard panned sounds.

The loop I started with had a hi-hat that wasn’t panned very hard -  I copied it to a new track, filtered out all the lows, boosted some highs and then panned it hard left. Then I recorded the combined original and panned track to a new file.

Here is what I’m starting with: listen

Now that I had something on the sides I could mess around with MS processing.

The first thing you have to do is convert left-right to mid and side. I use the free +matrix MS decoder from SoundHack.com. After that I used a delay plugin to add some filtered echoes just to the middle by disabling the right side input.

In the next insert I used a distortion on just the right side. This brought out a lot more of the reverb than was heard in the original loop. Lastly, second MS decoder was used to bring it back to stereo.

Here is how the loop sounds now with delay in the middle and distortion on the sides: listen

Pretty cool right!? I hope you have found these tricks useful.

Jon Tidey is a Producer/Engineer who runs his own studio, EPIC Sounds, and enjoys writing about audio on his blog AudioGeekZine.com. To comment or ask questions about this article go here.

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Mojave Audio Debuts MA-101SP Matched Pair Cardioid Condenser Microphones

Mojave Audio has introduced the new MA-101SP, a matched pair of MA-101fet cardioid condenser microphones for use in a range of stereo recording and live sound reinforcement tasks, with instruments such as drums and guitar amplifiers, as well as capturing room ambience and general stereo recording.

Each MA-101fet in the matched pair provides warm, full-bodied reproductions of instruments without the shrillness and high frequency artifacts so often encountered with modern condenser microphones.

The microphone’s warm FET circuitry and externally polarized capacitor mic element combine to deliver low noise and high quality performance.

The MA-101fet features both omni and cardioid polar patterns by way of interchangeable capsules and is outfitted with a 3-micron thick, .8-inch diameter gold sputtered diaphragm.

As one would expect from a David Royer designed microphone, each MA-101fet in the MA-101SP matched pair offer performance specifications that are impressive. Frequency response is 20 Hz - 20 kHz (+/- 3 dB), sensitivity rating is -40 dB (1 volt per pascal), and the distortion rating is less than 1 percent @ 120 dB SPL (-15 db pad off) and less than 1 percent @ 135 dB SPL (-15 dB pad on). The microphones operate on standard 48-volt Phantom power.

Mojave Audio president Dusty Wakeman states, “The new MA-101SP matched pair of microphones is the result of countless requests from the audio community. Drawing upon the strengths of the MA-101fet, these mics are a terrific choice for a wide range of stereo recording tasks where imaging is critical. 

“Engaging the 15 dB pad allows one to take advantage of the fast transient response on instruments such as snares, toms and loud guitar amps. The MA-101SP is a remarkably versatile general purpose recording and sound reinforcement tool that, I’m confident, will find a home in a wide variety of environments.”

The new Mojave Audio MA-101SP ships in a single carrying case that includes a stereo bar. MSRP is $1,195, and availability is Q1, 2012.

Mojave Audio

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Monday, February 06, 2012

Radial Introduces Shuttle Multi-Function Effects Insert Module For 500 Series

Radial Engineering has introduced the Shuttle, a new multi-function effects insert module for the 500 Series frame format and the Radial Workhorse.

The Shuttle offers three insert loops:

—Loop-1 is a front-panel insert that employs 1/4-inch TRS connectors for fully balanced connectivity
—Insert-2 is an unbalanced insert that is also front panel mounted that easily interfaces to standard effects devices
—Insert-3 is available on the Workhorse using the Omniport, which is wired following convention with tip-send, ring-return, making it ideal to interface with a remote patchbay.

All three loops are equipped with an insert switch that lets the user compare the wet and dry signal paths.

The insert points may also be used as inputs to feed a signal into the Workhorse mix bus. This opens the door to using the Workhorse with source devices such as CD players and iPods or with multi-channel fader packs and so on.

The Shuttle also enables those who own a Workhorse to easily integrate older 500 Series modules into the Workhorse mix buss. One mounts the non-Radial module next to the Shuttle, engages the feed function, and the signal will automatically be routed.

“As soon as our engineers started to integrate the Workhorse within the digital studio environment, they immediately noticed a need to simplify the process of patching effects in and out following what studios would normally do using a patch bay,” says Radial sales manager Steve McKay. “And as we delved further down the rabbit hole, we realized that the 500 Series was limited with respect to performing functions such as overdubbing. The Shuttle addresses these limitations while opening the door to creative new patching options.”

Radial Engineering

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Lexicon Offering Individual Plug-Ins From PCM Native Effects & PCM Native Reverb Bundles

Lexicon has announced the availability of the individual plug-ins from its PCM Native Effects and PCM Native Reverb Bundles, with a total of 14 plug-ins available, including Pitch Shift, MultiVoice Pitch, Chorus, Resonant Chords, Random Delay, Dual Delay, Stringbox, Vintage Plate, Plate, Hall, Room, Random Hall, Concert Hall and Chamber.

“Offering the individual plug-ins from our PCM Native Effects and PCM Native Reverb Bundles represents our commitment to provide Lexicon users with greater flexibility and ease to obtain exactly the sound quality they are looking for from the specific plug-in(s) they need for any project,” says Rob Urry, vice president Harman Professional Division & GM of Signal Processing and Amplifier Business Units.

The PC- and Macintosh-compatible plug-ins are designed to work with popular DAWs like Pro Tools, Logic and Nuendo, as well as with any other VST, Audio Unit or RTAS-compatible host.

Each plug-in can be run in mono, stereo or mono in/stereo out, and on-screen input and output meters are provided for precise level setting.

All Lexicon plug-ins are Native only, and require iLok2 authorization. The individual plug-ins will be available in February 2012.

Lexicon
Harman Pro

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In The Studio: Elliot Scheiner Interview Excerpt

His approach to mixing as well as some insight on some of his projects

 
Here’s an excerpt of an interview with Elliot Scheiner from The Mixing Engineer’s Handbook.

Ells has long been recognized as one of the finest engineers working today and has a shelf full of industry awards (five Grammys, four Surround Music Awards, Surround Pioneer Award, Tech Awards Hall Of Fame and too many total award nominations to count) from his work with The Eagles, Beck, Steely Dan, Fleetwood Mac, Sting, John Fogerty, Van Morrison, Toto, Queen, Faith Hill, Lenny Kravitz, Natalie Cole, Doobie Brothers, Aerosmith, Phil Collins, Aretha Franklin, Barbra Streisand and many, many others to prove it. He’s also one of the nicest guys in the business.

In this interview, Elliot talks not only about his approach to mixing but about some of his projects as well.
——————————————————————————————————-

Do you have a philosophy about mixing?

Elliot Scheiner: I’ve always believed that if someone has recorded all this information, then they want it to be heard, so my philosophy is to be able to hear everything that was recorded. 

It’s not about burying everything in there and getting a wall of sound.  I’ve never been into that whole concept. It was more about whatever part was played, if it was the subtleties of a drummer playing off beats on the snare drum next to the backbeat, obviously he wants that heard. So I always want to make sure that everything that’s in that record gets heard.

If you were able to accomplish hearing every single instrument in the mix, that was a huge achievement. Granted, maybe there wasn’t as much information when I started as there is now. I myself have come across files that have been a hundred and some odd tracks, so it’s not as easy to do that today. 

I have to admit that the way some people record things today is a bit peculiar. All of a sudden you’ll be dealing with 7 or 8 different mics on the same instrument. Like, for example, an acoustic guitar will all of a sudden have 7 different viewpoints of where this guitar’s being recorded. 

It’s mind boggling that you have to go and make a determination and listen to every single channel to decide which one you want to use. And if you pick the wrong ones they come back at you and say, “Oh, we had a different combination” or “It doesn’t sound quite right to us”, but they don’t tell you what they did!  So granted, it is a little more difficult to deal with those issues today, but I still take the same approach with every mix.

If you have a hundred tracks, will you try to have them all heard? Or do you go in and do some subtractive mixing?

Elliot Scheiner: Well, it depends if that’s necessary. I don’t usually get those kind of calls where they say “Here’s a hundred tracks. Delete what you want.” It’s usually not about that. And I have to say that I’ll usually get between 24 and 48 tracks in most cases and hardly ever am I given the liberty to take some of them out. 

I mean if something is glaringly bad I’ll do that, but to make a judgment call as to whether background vocals should be in here or there, I generally don’t do that.  I just assume that whatever an artist and producer sends me is kind of written in stone. They’ve recorded it, and unless they tell me otherwise, I usually don’t do subtractive mixing.

How long does it take you to do a mix on average?

Elliot Scheiner: Depending on how complicated it is, it usually takes anywhere from 3 hours to a day.

3 hours is really fast!

Elliot Scheiner: Yeah, well a lot of time you just get a vibe and a feel for something and it just comes together. Then you look at it and say “How much am I actually going to improve this mix.”  I mean if it feels great and sounds great I’m a little reluctant to beat it into the ground.

For me it’s still about a vibe and if I can get things to sound good and have a vibe, that’s all I really care about. I still put Al Schmitt on a pedestal. Look at how quickly he gets things done. He can do three songs in a day and they’ll be perfect and amazing sounding and have the right vibe. So it’s not like it can’t be done. Some people say that you can’t get a mix in a short time and that’s just not true and Al’s my proof.

Where do you usually start your mix from?

Elliot Scheiner: Out of force of habit, if there’s a rhythm section I’ll usually start with the drums and then move to the bass and just work it up. Once the rhythm section is set I’ll move on to everything else and end with vocals.

How much EQ do you use?

Elliot Scheiner: I can’t say that there are any rules for that. I can’t say that I’ve ever mixed anything that Al has recorded, but if I did I probably wouldn’t have any on it. With some of the stuff done by some of the younger kids, I get it and go, “What were they listening to when they recorded this.” 

So in some cases I use drastic amounts where I’ll be double compressing and double EQing; all kinds of stuff in order to get something to sound good. I never did that until maybe the last 5 years.  Obviously those mixes are the ones that take a day or more.

When you’re setting up a mix, do you always have a certain set of outboard gear, like a couple of reverbs and delays, ready to use or do you patch it as you go?

Elliot Scheiner: Usually I don’t start out with any reverbs. I’m not one for processing. I’d like to believe that music can survive without reverbs and without delays and without effects. Obviously when it’s called for I’ll use it, but the stuff I do is pretty dry. The 70’s were a pretty dry time and then the 80’s effects became overused. There was just tons of reverb on everything.

Most of your Steely Dan stuff is pretty dry, isn’t it?

Elliot Scheiner: It’s pretty much dry. What we used were plates usually.

Real short ones?

Elliot Scheiner: Not necessarily. In the days when I was working at A&R [studios in New York city] we had no remotes on any of our plates there. Phil [Ramone - producer and owner of A&R] wanted to make changing them difficult because he tuned them himself and he really didn’t want anybody to screw with them.

There would be at least 4 plates in every room. Some of them might be a little shorter than another but generally they were in the 2 to 2 1/2 second area. There was always an analog tape pre-delay, usually at 15 ips, going into the plates. The plates were tuned so brilliantly that it didn’t become a noticeable effect. It was just a part of the instrument or part of the music. You could actually have a fair amount on an instrument and you just wouldn’t notice it.

Is the sound of the A&R plates something that you try to get today?

Elliot Scheiner: Oh, I’m always trying to get that reverb sound If I’m using plates either at Right Track or Capital, I’ll still use an analog tape delay going into it.

For more of this interview, check out The Mixing Engineer’s Handbook

Bobby Owsinski is an author, producer, music industry veteran and technical consultant who has written numerous books covering all aspects of audio recording. For more information be sure to check out his website and blog.

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RE/P Files: A Quadraphonic Microphone Development

From the archives of the late, great Recording Engineer/Producer (RE/P) magazine, this feature provides an interesting look back at quadraphonic recording. This article dates back to September of 1970. (Volume 1, Number 3). The text is presented unaltered, along with all original graphics.

As a complete oversimplification, a microphone is an instrument which measures differences in air pressure.

It is not surprising that somebody would, in light of the interest in Quadraphonic sound, experiment and perfect an instrument which would measure and transduce the differences in air pressure around a full 360 degrees - to effectively create a quadraphonic microphone.

Such a truly Quadraphonic device, developed by engineer Carl Countryman and producer Brad Miller, is in external appearance no different than the several models of standard microphones (Figure 1).

This Quadraphonic microphone has been designed and built using the case and chassis of a Neumann SM-2, into which four independent microphone heads have been built to provide full 360-degree pick-up.

The pick-up patterns (Figure 2) are cardioid, front and back, and figure-8 at the sides.

Although the obviously complicated matrixing data are proprietary, and unavailable for publication, the discussion of pickup patterns, generally, yields an understanding of how the design provides excellent separation and naturality of sounds.

Cardioid, also sometimes called unidirectional, is a heart-shaped response. It is resultant of an omnidirectional and figure-8 pickup.

The signals are superimposed on each other; at the very rear they are anti-phase, and so cancel out.

At the front they are in phase, hence the tapering hear-shaped response toward the rear.

Figure-8, or bi-directional pickup-patterns, are the result of two directional pickup patterns, one in phase and the other anti-phase.

The output at the front and the back are equal, although opposite,.

As the input signal moves to the side, the output is gradually reduced until at 90 degrees, the two patterns have, for all intent and purpose, canceled each other out.

Figure 3 shows microphone capsules as they are arranged in the microphone head.

“Front to Back” and “Left to Right” are one above the other at 90 degrees to each other.

Three demonstrations, on very spontaneous, served to convince that development of the unit is very nearly complete.

The microphone was hung in Miller’s back yard garden, surrounded by about 200 degrees of sound source emanating from a waterfall with various small tributary streams flowing from it. It presented an excellent opportunity to “hear” the complete environment; the waterfall in stereo on the two speakers in “front,” and from behind, the beautiful ambiance of the total environment and the reflected sound.

Several minutes into the demonstration, on the Southern Pacific tracks bordering on the rear of the Miller garden, a slow-moving freight train ambled by. The completeness of the sound, the way it engulfed the listening room, is difficult to describe. It was totally complete… almost frighteningly so.

Miller completed the demonstration by playing a 4-track tape of his “Mystic Moods Orchestra” on an especially adapted Sony. The machine (Figure 4) has been adapted for 4-track, in and out, and will be able to accommodate 10-inch reels of 2-inch tape.

The machine is the forerunner of a new design from the Countryman/Miller collaboration which will weigh in the vicinity of 20 pounds.

The “Mystic Moods” piece only served to further impress that Quad or Multi is certainly on the way… with an endless spectrum of sound combinations and tonal effects.

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.

Take the PSW Photo Gallery Tour of audio equipment ads appearing in RE/P magazine, circa 1970

Our sincere thanks to Mark Gander of JBL Professional for his considerable support on this archive project.

Please send all questions and comments to ProSoundWeb Editor .(JavaScript must be enabled to view this email address).

 

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Sunday, February 05, 2012

Unit Audio Announces Affordable New Line Of Passive Summing Mixers

Unit Audio has introduced the Milli-Unit and Micro-Unit, two new 8-input by 2-output compact passive summing mixers for studio/recording applications. 

Both units are outfitted with eight balanced line-level inputs and two balanced microphone level outputs, all with Neutrik TRS connectors.

Input impedance is 20 Kohms, while output impedance is 220 ohms. Resistors are hand-selected, metered Xicon 1/4-watt, with 1 percent tolerance.

The units are hand-wired at the company’s headquarters in Nashville, TN, and are housed in rugged aluminum cases.

The Micro-Unit is also outfitted with two pan switches that allow for placing channels 1 and 2 in monaural (center), or hard left (channel 1), or hard right (channel 2).

“Is analog summing going to make your recordings sound like a Nashville studio with a billion dollars worth of equipment? Probably not, but you will notice a difference in your mixes using a Unit Audio summing mixer,” states Terry Auger, Unit Audio design engineer.

“Loosely quoting Shakespeare, one might say ‘To analog sum or not to analog sum?’” Auger continues. “This has been a point of controversy with digital recording for quite some time. With modern DAW software, mixing within the computer has resulted in some great sounding recordings, but I have long been intrigued by the concept of analog summing. I was not prepared to pay $800 or more to test that theory, so I engineered and built my own.

“Then to test the theory, I set out to see if there was any difference in the mixed sound. Much to my amazement and pleasure, I did notice a subtle but very pleasing difference in the stereo separation and placement of the instruments compared to my ‘in the box’ mixes.”

The Milli-Unit is priced at $149, while the Micro-Unit carries a price of $189. Both units can be ordered directly from the company website.

Unit Audio

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Friday, February 03, 2012

Blue Announces Mikey Digital Microphone For iPod touch, iPhone, iPad

Blue Microphones announces Mikey Digital, a plug-and-play external microphone for recording stereo audio on the latest iPod touch, iPhone and iPad models using iOS audio apps or the video function. 

It offers automatic and manual gain control and a multi-source auxiliary input for direct recording of guitar and other sound sources.

“Mikey Digital is the highest quality and most versatile solution for capturing professional recordings on your iOS device with most audio apps or the built-in video camera,” says John Maier, CEO of Blue Microphones. “Mikey Digital brings more than professional-quality recording to these mobile platforms, it also turns your iOS device into a studio interface for recording guitar, connecting a lavaliere mic or capturing line level audio straight to your iPod touch, iPhone or iPad.”

It includes two custom-tuned condenser capsules for capturing studio-quality audio—the same capsules used in Blue’s Snowball and Yeti USB microphones.

Mikey Digital users can switch between Automatic and Manual gain settings, allowing the automatic gain control to intelligently adjust to fluctuating volume levels or manually locking in a high or low sensitivity level. 

Mikey Digital is also equipped with an LED clipping indicator that signals whenever volume levels are producing distortion, providing immediate feedback to adjust for best recording results.

Furthering its capabilities, Mikey Digital also features a multi-source 3.5mm auxiliary input jack for stereo line-in, instrument-in and mic-in. This versatile input turns it into a multifaceted professional recording interface for direct recording of other sources including guitar, handheld mics, mixers and more.

Using a built-in, high-quality mic preamp and a CD-quality A/D converter, Mikey Digital records the highest fidelity possible on the iOS platform and can handle the high SPL of loud environments.  It also features a USB pass-thru connection for charging your device while in-use for continuous recording without draining battery life.

Outfitted with a 230-degree rotating head for optimal positioning, it can also capture enhanced audio while recording video by rotating the mic to align with the front- or rear- facing camera. \]

Further, it incorporates an advanced microprocessor allowing for upgradeability and future control through iOS applications. Plug-and-play, Mikey Digital directly connects to latest iPod touch, iPhone 4/S, iPad 2 and iPad and is instantly recognized for use with most audio and video applications.

Blue Microphones

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Thursday, February 02, 2012

MXL Introduces V67 Recording Microphone Kit

MXL has introduced the V67 Recording Kit, which includes the V67GS microphone as well as two V67N microphones, as well as mounts, adapters and a case.

The V67GS is a large-diaphragm cardioid condenser mic that combines Class-A FET circuitry and a transformer /coupled output for an open sound. The gold-sputtered diaphragm is 6 microns thick, and the capsule measures 1.26 inches. Frequency response is 30 Hz to 20 kHz.

The V67GS offers a smooth rising presence peak which is preferred for vocals when there is a heavy low end in the mix, and it includes a high-pass roll-off switch and additional -6 dB pad.

The V67N is a pressure-gradient condenser mic with cardioid/omni pattern, with a signature marked by the right blend of power, presence and clarity.

Hand-selected components and transformer-balanced output help deliver a solid bottom end and exposed top, making it good choice for drums, piano, guitar, choirs, and other applications.

The V67N’s 6-micron-thick diaphragm is also gold-sputtered, with the capsule measuring .87 inch. The mic is internally wired with Mogami cable for added sonic integrity. Frequency response is 20 Hz to 20 kHz.

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In The Studio: The Attack Of The Lopsided Stereo Monster

Nowadays I’ll either go with a single mic or use two mics in an XY configuration. Why?

 
A reader is puzzled by stereo (2-mic) acoustic guitar recording:

I recently got into mixing acoustic guitar with 2 mics. The problem is that I do not know how to create as much ‘space’ as some tracks I know of. I’ve tried XY, ORTF, and spaced pair.

XY and ORTF are too narrow. Spaced pair seems reasonable (following the 3:1 Rule), but the mic pointed closest to the body becomes overly ‘bassy.’

How can I balance the stereo image? EQ can control the problem but not by much. How would you go about on fixing this problem?

I know mic position has to do with it but I don’t know where to start. Just wondering if you had to overcome this type of problem before.

As much acoustic guitar recording and mixing as I do, I’ve dealt with problems like this a LOT.

(And this applies to ANY instrument, not just acoustic guitar.)

First things first…

Mic Placement Is Everything

I’ve played the “Hey, I’m Just Going to Throw a Couple of Mics in Front of the Guitar and Hope it Works” game.

It’s not a very fun game, trust me. You always end up losing.

Whenever you’re recording an acoustic instrument, always plan to give yourself at least a few minutes to try a few different mic techniques. I

s one mic (mono) appropriate? Does it need the wider sound of a stereo (2-mic) technique? If so, which technique is best?

There are a lot of options, and it would behoove you to try at least a couple of them before committing the recording to tape. (Tape…who says tape anymore?)

I feel like I’ve come full circle when it comes to stereo recording. I used to love a nice, wide acoustic guitar sound. But the last year or so I’ve simplified a lot.

Nowadays I’ll either go with a single mic or use two mics in an XY configuration. Why? Because having a really “wide” recording isn’t all it’s cracked up to be.

Even though XY doesn’t give the widest stereo image, it doesn’t lend itself to phase issues and lopsided recordings.

Speaking of lopsided, let’s talk about that stereo recording that has too much bass in one mic versus the other.

Even if you do your job on the front end with mic placement, sometimes one mic (the one pointed at the sound hole) picks up more bass than the other.

Here’s how I deal with it:

—Place an EQ plug-in on the bass-heavy track ONLY.
—Use the EQ to remove some of the excess low end, until the two tracks are more balanced.
—Bus the two tracks to a stereo aux track.
—Put any additional EQ and compression plugins on the stereo aux track.

The first EQ is simply corrective. It lets you balance out the sound. (No more lopsided guitar.)

The second EQ (on the stereo aux) is what you’ll use to carve the entire sound of the entire stereo recording to fit it in the mix.

As you may have guessed, mic placement and technique play a HUGE roll in how awesome your acoustic guitar recordings (and mixes) are going to sound.

Make that a priority on your next session.

If you’re interested in diving in deeper, I created a 4-week class on getting consistently awesome acoustic guitar recordings. You can join any time here.

Joe Gilder is a Nashville-based engineer, musician, and producer who also provides training and advice at the Home Studio Corner.Note that Joe also offers highly effective training courses, including Understanding Compression and Understanding EQ.

 

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Tuesday, January 31, 2012

The Differences Between Vintage And Current Instruments

 
During a long discussion about vintage instruments in the studio this week, it prompted me to think about this excerpt from The Ultimate Guitar Tone Handbook (written with writer, composer and good buddy Rich Tozzoli) that describes some of the intangible factors that went into manufacturing Gibson humbucking pickups in the 50’s and 60’s.

As you’ll see, there are a lot of external factors that went into making a pickup back then, and those factors can pretty much be applied to all instruments in one way or another.

————————————-

“As if the known factors in building a pickup weren’t enough, consider the many intangible factors as well. For instance, most pickups loose their magnetic strength over time because of environment and electrical interference. Pickups can become weakened or demagnetized completely by leaning your guitar against an amplifier with large transformers, or even from taking your guitar too close to the train motor of a subway (as happened with Andy Summers of The Police).

Another intangible is the fact that tolerances of just about every component were much looser until the 90’s. While the difference was indeed subtle, add enough components at the edge of their tolerances together and you suddenly get a pickup that sounds different even though it’s made the same.

Manufacturing intangibles are a whole other story and for that we’re going to go a bit into the history of the Gibson humbucker.

The Changes In The Humbucker
The first humbucking pickups on the 1957 models of Gibson guitars had a sticker on them saying “Patent Applied For” as the design was in the review cue before being granted a patent (see Figure 3.27). These became known as PAF pickups (“Patent Applied For”) and have become highly sought after today for their great sound.

The problem is that most PAFs sound different from one another due to manufacturing process of the time.

Until 1961 when Gibson standardized the selection process, they randomly used different strength magnets (grade 2 through 5) in their pickups, which accounts for some of the reasons for the different sounds. To make matters worse, sometimes a shorter magnet was selected (mostly seen in gold-plated guitars for some reason), which decreased the power of the magnet as well.

In July of 1961, Gibson consistently began to use all short Alnico 5 magnets, although occasionally a few Alnico 2’s showed up. In 1965, Alnico 5’s became standard in all pickups, which finally brought about a bit of consistency to the process and the sound.

If that weren’t enough, the number of windings on the pickup varied enormously as well, especially in PAFs. The early coil winding machines didn’t have an auto shut-off so the workers would shut off the machine when the bobbin looked full, which was at about 5000 turns. As a result, no two pickups were ever the same.

Even when Gibson bought a winder with an auto-stop, there continued to be problems even though the pickups became more consistent. The stop mechanism was controlled by a fiber wheel which would wear out and break, at which point the workers would approximate the number of winds by timing the wind, which resulted in more inaccuracies.

Since the humbucker is made up of two coils, sometimes the windings of each coil were different even though the total number of turns were correct. This would cause certain mid-range frequencies to stand out and give it more bite.

By mid-1962, the patent for the humbucker was granted and Gibson changed the sticker to read “PATENT NO 2,737,842” which was actually the patent number for Les Paul’s trapeze tailpiece. No one knows for sure if printing the wrong number was merely a mistake or a way to throw off the competition.

From 1963 to 1975, these “Patent number” pickups are very consistent, as are the ones thereafter when new, more precise winding machines were used (see Figure 3.28).

In the 1990’s, Gibson further refined their manufacturing and began to manufacture pickups based on the original PAF design. Thanks to precision modern manufacturing techniques, these pickups are remarkably consistent, which also means that a “magic” pickup made as a result of loose tolerances is no longer possible to get.

That being said, most experts agree that you can now get 90 percent of the way there sound-wise for 10 percent of the cost of a vintage PAF.”

Bobby Owsinski is an author, producer, music industry veteran and technical consultant who has written numerous books covering all aspects of audio recording. For more information be sure to check out his website and blog.

 

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Monday, January 30, 2012

New Wave Entertainment Utilizes Fairlight EVO and Xynergi Consoles

Supporting New Wave Entertainment in its content production for creative marketing campaigns and original TV programming is Fairlight’s EVO and Xynergi consoles.

New Wave Entertainment, located in Burbank, California, specializes in the creation of content, which spans the realm of creative marketing for TV networks and major motion picture studios to original TV programming. Their work has supported campaigns for blockbuster movies such as Avatar, Harry Potter, The Hangover Part 2, Happy Feet Two, and Alvin and the Chipmunks. Some of their recent TV promo work includes a spot for “The Ellen Degeneres Show” which was awarded a Creative Emmy, and a TV launch campaign for “Extra.”

New Wave Entertainment has five production suites, each with Fairlight’s EVO console. One room hosts the Xynergi console with a sidecar, adding tactile mixing controls. That elevates the traditional console-like hardware interface on the Xynergi platform to a full range of audio production capabilities from recording, editing and mixing. With limited space, the, the compact digital consoles of Fairlight provide the luxury of additional rack space.

A majority of the work New Wave tackles is on-air marketing, including television spots and theatrical trailers. They also do a large amount of Blu-ray Features & Menu Design, Broadcast Design, Mobile Content and 3D Modeling and Animation, which has been enabled by the flexibility of Fairlight consoles.

According to Mark Rodrigues, Chief Engineer and Senior Mixer for New Wave Entertainment, “The biggest selling point for us is the new dual video track, allowing us to work with 3D video. As 3D advances, the Fairlight systems give us the chance to evolve as well. The dual track helps us recognize the changes being made as many eyes are constantly viewing the content and adding input.”

“The presentation of the video track makes it extremely easy to follow. The ability to take video sections from one spot and string together a bunch of segments with the added option to edit with sound included is fantastic,” Rodrigues added.

With five suites each working on the same project, New Wave can do 20 spots and convert them to 10 different versions for different broadcasts quickly. Everyone can mix the same way with default settings for standards and compression, and simply add in their changes, which helps Rodrigues to take on the substantial number of projects that they do. “A project can be posted on a FTP site and we can be ready to mix and edit it within seconds.”

For Rodrigues, the drag and drop conversion process helps to save time and energy and allows workflow to stay constant. He notes, “Quicktime support is very valuable to us since most of our approval isn’t done in-house. It’s great to send short clips and have our clients understand what we are working on and they can give their input as well. We can change the frame rate and merge different files together without a worry.”

Rodrigues says one feature he enjoys is the ability of Fairlight to have full support of native plug-ins wherever you need it. “We use it on a daily basis for surround mixing and dialogue cleaning, and it has enhanced the process of editing.”

“I’ve always loved the sound with Fairlight consoles,” adds Rodrigues. “They have a warmth that other systems lack. Everything has worked out extremely well and we are happy with everything that Fairlight offers from the speed of workflow to the support that they have given us to help us meet the needs of our clients.”

Fairlight

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Friday, January 27, 2012

Lauten Audio Unveils New Atlantis Large-Diaphragm Condenser Microphone

At the NAMM 2012 Show, Lauten Audio and its North America distributor, Audio Plus Services , introduced the FC-387 Atlantis microphone, a solid-state multi-functional, large-diaphragm condenser outfitted with multiple switches for three different polar patterns, gain, and unique timbre settings.

The FC-387 is for recordists seeking a diverse and useful modern FET studio microphone, offering a blends of full and rich low and mid-range, as well as smooth and unique high-mid and high-frequencies.

The multiple switches located on each side and back of the FC-387 Atlantis microphone allow the engineer to uniquely configure it.

First is a polar pattern switch giving recordists the option of choosing between cardioid, omnidirectional or figure-8 polar patterns.

Borrowing the popular gain options from its sibling the Lauten Audio ‘Clarion’ FC-357 microphone, the Atlantis features a -10 dB and +10 dB switch.

The +10 dB gain switch increases versatility by allowing a choice of whether to have more character from the preamplifier gain or directly from the microphone. While the -10 dB switch reduces its output and increases the maximum SPL level allowing it to record very loud sources.

In addition, a voicing switch gives recordists three very different timbres to choose from that satisfy an extremely wide range of recorded sources. The options are: gentle, neutral and forward.

The ‘gentle’ position provides maximum control of bright or peaky sources like S’s in vocal recordings. The ‘neutral’ position offers a nice even response with good control over vocal S’s and other audio peaks, while the ‘forward’ position can help bring life to dull sources without having to use EQ, but while still maintaining control over any peaks.

Brian Loudenslager, founder of Lauten Audio, explains, “The idea of the FC-387 Atlantis microphone was conceived over several years during conversations with engineer Fabrice DuPont of Flux Studios and Puremix.net. Fab had acquired several Lauten microphones over the years and I would pick his brain at industry tradeshows about the mics. I always try and speak to owners of our products to learn where they’re using a mic most, where they feel it’s shining and where it’s not working for them.”

“We’re pleased to be at NAMM with Lauten Audio to introduce the Atlantis FC-387 microphone,” states Simon Côté, manager of professional products at Audio Plus Services. “We believe in Lauten Audio’s product direction and especially their dedication to serving respected recording engineers with special designs. The ‘Atlantis’ mic will no doubt be very popular.”

Loudenslager adds, “What I am most impressed about Fab is that he is always brutally honest and never sugar-coats his opinions. It was during conversations with him that I made a challenge to myself; I wanted to design a microphone that Fab would love, one that would be part of his first choices at the beginning of a session! Over the past year I’ve had the opportunity to work with Fab and create that mic. It’s been challenging with a very rewarding end result.”

Technical Specifications for Atlantis FC-387
Type: 31.25mm dual large diaphragm pressure gradient transducer microphone
Polar Patterns: Omnidirectional, Cardioid and Figure-8 selectable
Circuit: Low-noise solid-state FET
Frequency Range: 20 Hz - 20 kHz
Dynamic Range: 120 dB minimum
Impedance: < 200 ohms
Max. SPL: 0.5% THD@1000 Hz: 130 dB
Self-noise Level: < 12 dB(A)
Sensitivity: 16 mV/Pa OR -36±2 dB 0 dB=1V/Pa 1000 Hz
Special Features:
3-way -10 dB attenuation, 0dB and +10 dB gain switch
3-way "gentle, neutral, forward" voicing switch
3-way polar pattern switch
Connector: 3-Pin standard XLR
Power Requirement: +48-volt phantom power

The Atlantis FC-387 microphone retails for $1599 (U.S.) and will be available in Q2 of 2012.

Lauten Audio
Audio Plus Services

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Thursday, January 26, 2012

NAMM 2012 Show Central: Daily Ongoing Live Coverage

All of the latest from Anaheim...

Welcome to ProSoundWeb’s ongoing coverage for the NAMM 2012 Show.

Held January 19-22 at the Anaheim Convention Center, the show drew more than 95,000 attendees from the world of professional audio and music, coming from more than 100 countries.

The show floor hosted exhibits from 1,400-plus manufacturers, with hundreds of new products expected to make their debut. NAMM also provides an ever-growing slate of educational courses and programs.

In addition, the show offered well over 100 live performances and events both on-site and in venues proximate to the convention center.

A couple of interesting notes:
—This was the 110th NAMM Show, making it one of the longest running trade shows in the U.S.
—The show annually generates more than $70 million in revenue to the Orange County economy (Source: calculated using Trade Show Week formula of economic impact)

PSW is continuing to provide updates from the show. Be sure to check back here often.

NAMM 2012 Show News
New Board of Directors Elected During NAMM 2012 Show
110th NAMM Show Reaches New Record Number Of Registrants

New Products

Loudspeakers
Peavey PVX Active & Passive Loudspeakers
On Point Audio OPA28 NP High-Output, Dual 8-Inch Loudspeaker
JBL Professional PRX400 Series Portable PA Loudspeakers
Yamaha DXR Series Active Loudspeakers
Electro-Voice ZXA1-Subwoofer
D.A.S. Audio Action Series Of Active & Passive Loudspeakers
JBL Professional VTX Line Array Series
Spectr Audio S Series Compact Active & Passive Loudspeakers
High-Power Loudspeakers Join Eminence Professional Series
On Point Audio ACTIVE Loudspeakers With Powersoft Amplifiers
Public Beta Version Of JBL HiQnet Performance Manager Software

Consoles/Mixers
PreSonus StudioLive Mixers With Smaart System Analysis Tools
Peavey PVi 8500 & PVi 6500 Powered Mixers
Yamaha MGP12X And 16X Analog Mixers
New Op-Amp Design For Mackie Mixers
Soundcraft Si Compact V2 Software
Mackie DL1608 16-Channel Digital Mixer With iPad Control
PreSonus QMix App: Monitor Mix Control Via iPhone/iPod Touch
DiGiCo UB MADI
Behringer iPad Mixers
Allen & Heath ZED-16FX and ZED-18 Multipurpose Mixers
Roland Systems Group VR-3 A/V Mixer
iConnectMUSE Palm-Sized Digital Audio Mixer For iOS Devices
Allen & Heath GLD Live Digital Mixing System

Microphones
Lauten Audio FC-387 Atlantis Condenser Microphone
Lavaliers, Earset To For Shure Microflex Microphone Line
Headset Option For DPA D:Fine Series Microphones
Audio-Technica AT2005USB Cardioid Dynamic USB/XLR Microphone
TELEFUNKEN M81 Universal Dynamic Microphone
Audio-Technica Limited Edition ATM25 Instrument Microphone
Audix Band Packs Microphone Packages
TELEFUNKEN ELA M 260 Tube Mic Stereo Set
CAD Audio Updated E300S Condenser Microphone
Audix FP QUAD Drum Microphone Pack

Wireless Systems
AKG WMS 40 MINI 2 Dual Wireless Microphone System
Shure ULX-D Digital Wireless System
Sennheiser XS Wireless Series
AKG DMS 70 Digital Wireless Microphone System

Processors
BSS Audio Soundweb London BLU-805 And BLU-325 Processors With AVB
Aphex EX•BB 500 Series Module With Aural Exciter & Big Bottom Processors
Eventide 2016 Stereo Room And Omnipressor Plug-Ins

Amplifiers
Crown Audio I-Tech DriveCore Series Multichannel Power Amplifier
Crown Audio HiQnet Band Manager 2

Monitoring
AKG IVM4500 In-Ear Monitoring System
Pivitec e32 Personal Mixer With 32-Channel Ethernet AVB Capability
Sensaphonics Upgraded IEM line With New Cable, “Crystal” Colors
Sony MDR-7550 In-Ear Monitors
Aviom Pro16 Personal Mixing Systems
Future Sonics mg5pro Ear Monitors
POSSE Audio Personal On Stage Sound Environment System

Stage/Studio
“Dangerous Source” Portable Desktop Monitor Controller From Dangerous Music
Radial Engineering Firefly Tube Direct Box
Behringer FIREPOWER FCA610 & FCA1616 Recording Interfaces
Waves Audio NLS Non-Linear Summing Plug-In
iZotope Mastering Essentials For Acoustica Mixcraft Pro Studio 6
Universal Audio Apollo Audio Interface
Auralex SonoLite Bass Traps At 2012 NAMM Show
Lynx Studio Technology Hilo Reference AD/DA Converter System
Expanded Sony MDR-7500 Series Professional Headphone Series
CAD Audio HA4 Headphone Amplifier, MH110 Studio Headphones
Auralex Portable & Stand-Mountable ProMAX Panels
Griffin Technology StudioConnect & MIDIConnect For iOS
Three New USB MIDI Keyboard Controllers From Alesis

At The Show…
“How To Get a Job In the Industry” Forum At NAMM 2012
NAMM 2012 Interactive Show Floor
Plan The 2012 NAMM Show Using Your Smartphone
NAMM 2012: Live Music In the Lobby Schedule
Comprehensive List Of NAMM 2012 Exhibitor Appearances & Events
NAMM 2012 Concerts & Performances
H.O.T. Zone Hands On Training Sessions At NAMM 2012
NAMM Hosts Congressional Briefing On Lacey Act
NSCA Education And Outreach Sessions At The 2012 NAMM Show
Educational Schedules For NAMM University At 2012 Show
Sennheiser Sound Academy Two-Day Live Production Workshop Prior To NAMM 2012
Meet The Winners Of The Third Annual Readers Choice Best Product Awards

Special Events
Audio-Technica Marks “50 Years of Passionate Listening” (Includes Video)
John Lennon Educational Tour Bus Marking 15th Anniversary At NAMM 2012
House Research Institute Offering Free Hearing Screenings At NAMM 2012
Lectrosonics Promising “Silent Booth” For NAMM 2012 Show
H.E.A.R. & NAMM Team Up To Prove Free Ear Plugs At NAMM 2012 Show

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