Review
Monday, February 28, 2011
Acid Test: A Look At iZotope RX 2
A welcome upgrade from its predecessor and what should be a required tool in every engineer's bag of tricks.
Imagine for a moment one of the “must have” tools in your arsenal. I’d bet a large number of you out there jumped to ProTools (perfectly valid), or a favorite piece of outboard gear (I’ve been there).
However, one of my “can’t live without it” tools is noise removal and spectral repair, which is why I was so thrilled to have the opportunity to check out iZotope RX 2 when it came out this past fall.
However, the reason for my excitement was far different that you may imagine. After all, yes, everyone likes to be the first to check out new gear. Rather, I was intrigued because I’d already been an avid RX Advanced user for a number of years, and my mind reeled at what they could have done to possibly improve upon the software.
Boy was I surprised!
Two new features I’ve really enjoyed in the new version are improved batch processing and the upgraded selection methods. The new method of batch processing has made an impact on my workflow because you can chain multiple modules together as well as process multiple files while using multiple CPUs which can really save some time.
Also, the new selection methods which include a lasso, brush and magic wand tools have made it far easier to edit quickly and accurately. My favorite is the magic wand which can automatically select a sound and its harmonics.
As always, there are several features in RX Advanced which really make the jump to Advanced worthwhile. The new adaptive denoiser mode, which is a part of their all new denoising algorithms, is a whiz at dealing with dynamic background noise.
Additionally, azimuth alignment, which is a component of the new channel and phase operations feature, does a great job of correcting misaligned tape heads if you find yourself regularly restoring tape.
There are so many great additions to this update that I was left wondering how I’d ever worked without them, like session support, third party plug-in support, and if you’re using RX 2 Advanced for forensic work then the ability to export your work history as an XML file will come as a welcome addition.
My first true test of iZotope RX 2 Advanced came while hurriedly editing a project for my wife. A middle school teacher, she’d recorded her students reading aloud for a podcast. Knowing it was just a school project we opted to go the quick and dirty route for recording, an iPhone.
While the source was captured just fine, I’d underestimated the level of background noise in the 1920’s building. Against a rather tight deadline, I was able to run several hour long files through RX 2 Advanced, using the new adaptive denoiser mode, some moderate EQ, and the occasional spectral repair, which all came out sounding far better than I had expected given their original state.
In speaking with a colleague of mine after cleaning up these recording I said “I’ve got to write about this and tell everyone about this great software!”, to which he replied, “No, I don’t want anyone to know. Its my secret!”
That seems to be a common sentiment among iZotope RX users, as those who use it regularly are constantly pleased and amazed by its power, but always hopeful they can keep the app as their own secret silver bullet of post production.
Give it a shot; I’m certain you’ll see what I mean.
iZotope RX 2 / RX 2 Advanced Specifications
Windows (XP, x64, Vista, 7)
Mac OS X 10.5 or later (Universal Binary)
Standalone application
Plug-in formats: Pro Tools 7+ (RTAS/ AudioSuite), VST, MAS, Audio Unit, DirectX
iZotope Website
The Technologist, a.k.a. Kyle P. Snyder, is an audio engineer with innumerable credits in the public and private sector, writing about audio engineering, recording technology and a multitude of other topics as Associate Editor of ProSoundWeb. Find out more about Kyle at his website and blog http://kpsnyder.com.
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Friday, November 20, 2009
Barry’s Toolkit: First Look At The New Lexicon Native PCM Reverb Plug-In
Barry provides detailed look and hands-on review of Lexicon's Native PCM Reverb plug-in bundle, the company's long-awaited entry into a software-based reverb
The Lexicon Native PCM Reverb plug-in offers great, usable factory preset reverb sounds and, at the same time, “deep as you want” intense programmability for modifying or designing your own unique reverberant treatments.
Programming is now more precise due to the vast collection of algorithmic specific parameters and the three-dimensional RTA (real-time analyzer) color display that allows the visualization of the reverb’s shape, character and evolution over time.
Ported in VST, AU and RTAS Native formats for both MAC and PC, this bundle consists of seven plug-ins each using its own specialized reverb algorithm.
The seven reverbs are categorized as: Chamber, Concert Hall, Hall, Plate, Random Hall, Room, and Vintage Plate. Because each reverb plug-in has a dedicated “engine”, each comes with its own set of relevant parameters that are fully adjustable within the intuitive GUI.
Yet the interface’s look, feel and operation remain consistent across all seven plug-ins. For any one who has ever used the LARC to control either a Lexicon 480L and 960L reverb,
Lexicon Native PCM is an immediate “install and use” reverb - for the most part—no manual reading required. For those new to reverb tweaking will find the most important, salient parameters immediately available and easily adjustable.
The manual gives excellent descriptions of all the parameters and how they change the sound of the reverb.
It’s All In The GUI
Once you’ve decided upon and instantiated one of the seven reverbs best suited for your application, you must define or specify its exact nature using the Category pull-down at the top of the GUI.
The Chamber plug-in has subcategories called small medium, and large chambers; Concert Hall reverb is divided into Rooms, Small Halls, Medium Halls, Large Halls and effects; Hall has Small Spaces, Small Halls, Medium Halls, Large Halls, and Huge Halls.

Sample graphic: Lexicon PCM Plug-in Concert Hall Multiband Bandpass. (click to enlarge)
The Plate plug-in has Small, Medium, and Large Plate Reverbs, and Random Hall offers Small Spaces and Small, Medium, and Large Random Halls.
Lexicon’s Room reverb provides Small, Medium, Large Rooms, Drum Room, Small and Medium Halls, Large Halls, Exterior Places, and Effects.
Finally, the Vintage Plate reverb contains plates designed for Instruments, Vocals, Live Sound, Drums and Percussion uses.
Each of these subcategories of reverbs comes with 50 or more evocatively named factory presets sonically tailored in very specific ways using all the different combinations of that reverb’s individual set of parameters.
All parameters are available for editing and all are automatable within your DAW’s automation system.

Sample: PCM Plug-in Concert Hall Multiband Notch. (click to enlarge)
Below the Category and Preset pull-down menus are stereo I/O level meters, two equalizers available for both the Early Reflection signal and the Tail portion of the reverb, and a real-time audio analyzer display.
The I/O meters are standard LED trees with a handy signal present indicator. The EQ section has choices of one-pole (6 dB/octave) or two-pole (12 dB/octave) filters in Lo Pass, High Pass, Bandpass or Notch topologies.
There is a graphical display showing the filter curve imposed on the early reflection sound, colored in blue, and the reverb tail EQ curve shown in red.
Real-Time
The real-time analyzer in the center of the upper half of the GUI is a spectacular, three-dimensional multi-band tool for actually “seeing” the reverb sound build up, play out and evolve over time. The time line runs from right to left with reverb level represented vertically.
Once a reverb signal is present, different color-coded audio waveforms parade across the screen. Each of the moving waves represents a frequency band from 50 Hz to 12.5 kHz with the lower frequency bands in the back and higher frequencies in the front.
I found the RTA edifying—perfect for adjusting the Early Reflection and Tail EQs, predelay, Bass RT, Reverb Out level and Size parameters.
Being a summed mono display, for stereo parameter settings rely on your ears, as you should for all adjustments. The RTA confirms what you are hearing and is mesmerizing too.
There are also two other ways to “read” the reverb sound: a more conventional and colorful 2-D spectrum analyzer bar graph and a linear moving amplitude waveform display.

Sample: PCM Plug-in Hall Frequency LoPass. (click to enlarge)
The manual points out that these dazzling displays require additional host DSP so after impressing that client watching over your shoulder, save some DSP and turn them off before closing the plug-in’s window.
Reverb Controls
The remaining lower 2/3 of the GUI is devoted to the control faders. I found these faders to “mouse” very responsively without delay or glitching like other plug-ins.
Depending on the reverb plug-in you’ve called up, this section can be populated with up to nine parameter faders.
I liked that the most salient and useful faders are shown and for the most part, similar parameter faders from one reverb plug-in to another reside in the same slot—such as one of the most changed parameters—RT (reverb time).

Sample: PCM Plug-in Hall Impulse LoPass. (click to enlarge)
For the most part, there is little need to drill deeper than this page.
There are so many presets in each Category, I found toggling through them quickly gets you close to what you want and then a quick fader move or two gets you all the rest of the way there.
Each parameter fader position is named and has a value box for manual entry. Clicking on Edit allows any of the parameter faders to be re-tasked or changed to control any of the other available parameters by using the pull-down menu under the Modifier button.
So if you’d like the parameter Reverb Time to always be the first fader or any have other parameter(s) that does not normally show up here to be visible, you can make it so.
While in Edit mode, for deeper programming, the Soft Row becomes visible for selecting another row of parameters for access to more of the plug-in’s under-the-hood divinity.
Depending on the reverb plug-in instantiated - I’ll use the Chamber reverb as an example—the Soft Row will have the button names: Input & Mix, Reflections, Reverb and Echoes. When any of them are clicked, sub-parameters faders appear.
For example, when Input & Mix is clicked, Mix (wet/dry), Diffusion, Shape, Spread and Predelay parameters become available.
If applicable, additional buttons above these sub parameter buttons will show up.
In the above example, Predelay (as do all delay parameters in all plugs) has a toggle switch for either Absolute (the fader positional value) or Tempo-based from a 32nd note to a half-note value predicated upon the session’s tempo.

Sample PCM Plug-in Vintage Plate Display Off LoPass. (click to enlarge)
It sounds more complicated here than it really is.
Know that there is the good ol’ Compare button to show the preset default parameter settings if you get lost in a wilderness of bewilderment.
Lastly, the Lexicon Native PCM bundle has its own comprehensive interface for managing, naming, and saving modified user presets.
As user presets are accumulated, they are listed in the aforementioned Category pull-down menu.
User presets are stored within the plug-in itself, instead of in the DAW’s plug-in folder, as usually the case.

Sample: PCM Plug-in Vintage Plate Multiband Notch. (click to enlarge)
This means that they, along with all the Native PCM reverbs, are available for other programs in your computer. If you sequence in Logic and mix in Pro Tools, you can keep the reverb sounds consistent across both platforms.
You could also share them (as XML text files) with other systems or between Mac and PCs. (Now that is cool!)
Let’s Use This Thing!
I installed Lexicon Native PCM Reverb into a MAC PPC Quadcore running OS 10.4.11 and open it as an RTAS plug-in in Pro Tools HD session.
While the software runs fine in this old OS, it runs better in 10.5 or above. It authorizes via iLok and there are mono, stereo and mono (input)-to-stereo (output) versions available.
Great room sounds might be the ultimate quest - the ‘holy grail’ for high-end reverberators with percussive sounds the most challenging sound sources.
How does the reverb algorithm handle percussive attacks and simulate the thousands if not millions of reflections that happen in a real room? Does it ‘boing’ when a sharp-sounding snare drum is put to it?
In short, how realistic does a synthesized room sound?
I have to say I was impressed from the moment I first heard this plug-in at the recent New York AES convention - the true, great sound of a Lexicon room in a plug-in! I am so ready for this!
In Session
Here at my Tones 4 $ Studios, I was in the middle of a hard rock project at the last tweak stages and I wanted to replace other reverbs with Lexicon Native PCM to see if I could get something better going on.

Sample: PCM Plug-in Vintage Plate Frequency Notch. (click to enlarge)
The current pop/rock music aesthetic does not call for much long reverb treatment but the producer and artist wanted the drums to sound like they were recorded in a big room - which of course, they were not.
The first song moved at a fast clip - 153 BPM so I couldn’t add a giant room with a lot of lengthy aftermath. It had to build quickly and decay fairly quickly.
I chose the Medium Vocal Concert 1 in the Medium Halls category of the Concert Hall plug-in. I set Predelay to a 32nd note, Reverb Time to 2.3388 seconds, Reverb Out Frequency to 7125Hz, and Diffusion to 70 percent.
Tail Width sets the stereophonic width of the reverb’s tail and I narrowed it down to 38 degrees because the default wider stereo setting tended to wash the stereo drum stage out.
In the Reverb sub parameters, I set Bass RT to 2.75X because I put some of this reverb even on the kick drum. Size was set to 39 meters. The GUI looked like the Barry’s EXAMPLE 1 graphic here.

Barry’s EXAMPLE 1 graphic. (click to enlarge)
By adding a small amount of this reverb, I got drums to sound like they were playing in small hall or large reverberant club show room.
The drum sound was present, realistic and powerful sounding. While a drummer playing at this tempo cannot do many fills or anything else, whenever there were breaks in the music, the reverb tail sounded like room decay.
The second song was at a more stately tempo of 66 BPM—an old-school MOR ballad the producer wanted “big boy” reverb cake on the drums and most everything else.
I started with the Large Hall preset in the Large Halls Category in the Hall Reverb plug-in. I set Predelay to 16th-note and Reverb Time to 1.9801 seconds. Tail Width was at 99 degrees but check this in mono—you may not like what you hear and return it to normal stereo.
Reverb Out Frequency was 6,500 Hz but RT Hi Cut was at 6,250 Hz (default) and I also kept the default size to 32 meters, as shown in the Barry’s EXAMPLE 2 graphic below.
This sound, including the 16th note predelay, was perfect for this song. This Hall preset sounded huge with super clean tails that faded into the mix’s noise floor (what little there was).
I subsequently added another stereo reverb to fill in the 16th note predelay gap and the phasey tail out of the first reverb. I wish reverbs had a blend control where you could “leak” around the predelay section for filling in that space.
In both of these two brief examples, I got the reverb sounds very quickly - but, in fact, there are so many excellent preset choices with so many ways to “dial them in,” making decisions takes more time than actually getting a sound.
It’s easy to go very surreal with a lush, wonderful and huge reverb as in the case of the ballad, or more realistic with a tougher and harder sound I developed for the faster rock song.

Barry’s EXAMPLE 2 graphic (click to enlarge)
I found it super important to learn the exact nature of the seven reverb plug-ins - the Categories.
For me it was like learning the categories in the Lexicon 480L or 960L reverbs: once I developed a sonic familiarity of each, I usually “nailed” my initial choices first time rarely starting over by changing categories.
I have no other reverb, plug-in or hardware, with this much versatility - but then I don’t own any Lexicon reverbs except this one! If I were allowed only one reverb bundle, this would be the one for me!
I am anxious to get into my next mix project where I can start fresh and run several Lexicon Native PCM reverb plug-ins working together.
Lexicon Native PCM sells for $1,899.95 MSRP. Lexicon will start shipments of the plug-in to authorized dealers the first week in December. Purchase and download of the plug-in directly from http://www.lexiconpro.com will occur in January 2010.
For more information, go here, and also be sure to check out videos here.
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Barry Rudolph is a veteran L.A.-based recording engineer as well as a noted writer on recording topics. Visit his website at www.barryrudolph.com
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More Reviews & Articles By Barry Rudolph On PSW:
A Wide Variety Of Microphone Techniques For Recording Drums
The Tale Of A Project-Saving Monitoring Technique
Test Driving The Focusrite Saffire PRO 40 Firewire Audio/MIDI Interface
Rhythm Section Tracking In The Studio
Does The WAVES Hybrid Line Of Plug-Ins Enhance The Creative Process?
Creative Uses For Loudspeakers To Enhance Your Recordings
The Shure 55 Microphone Has Deep Roots, But How Does It Hold Up Today?
Thumbs Up Or Down For The Marshall MXL V89 Studio Condenser Microphone?
Inside The Peluso P12 Tube Condenser Microphone
Barry’s DAW Toolkit: Review Of The Novation Nocturn With Automap 3 Pro
Barry’s Recording Tips: Figure Of Eight Royer For Electric Guitars
Review Of The X-Tempo Pok DAW Wireless Footswitch Controller
Barry’s Toolkit Of Handy DAW Products
Recording Gear Hits At The 2009 Winter NAMM Show
Working At Recording Success: Taking Elemental Steps Can Make All The Difference
Recording Tip: Successfully Dealing With A Dead Room
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Monday, November 09, 2009
Road Test Underway: Switchcraft SC800 Instrument Direct (DI) Box
Veteran audio professional Mark Frink provides perspective and an overview of a recent entry in the passive direct box field, the SC800 (Part 1 of 3)
Ten years ago, in the process of putting seven active direct boxes (DIs) - including the original Radial JDV - to a critical measurement and listening test, I discovered what has gone on to become an industry standard, Radial’s JDI. passive DI.
You know, the green one with a genuine Jensen transformer inside.
Now there’s another classic passive direct box, the Switchcraft SC800.
Over the years, Jensen transformers have established themselves as a reference benchmark.
Other manufactured Jensen DIs include the Whirlwind Direct-JT and D.W. Fearn PDB passive direct box.
Both employ the Jensen JT-DB-EPC, which is a PC-mount version of their classic JT-TB-E.
Many an old-timer earned his DI badge building a JT-DB-E into a bud box to make a homemade direct box: the way they were all made before the 1980s.
Not surprisingly, all these excellent passive DIs exhibit similar specs and sound, leaving most comparisons to design, features and packaging.

Jensen JT-DB-EPC transformer.
The SC800 passive direct box is built from a black anodized, laser-etched rectangular aluminum extrusion, with countersunk #2 Phillips screws fastening the recessed end panels.
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At 2 inches high by 3.5 inches wide. it’s similar in shape to many familiar DIs.
Its 4-inch length is shorter than most others by about an inch, and there are two strips of rubber on the bottom to act as isolation feet. I like that there’s enough room to tape a business card on one side with clear packing tape to ID it when lost or misplaced. The SC800 looks sharp and professional.
A passive direct box simply converts the outputs of high-impedance unbalanced line level audio devices or musical instrument pick-ups by means of a transformer to match balanced XLR low-impedance professional mic pre-amps and mixers.

Front panel view of the SC800. (Click to enlarge)
By eliminating the need to use a microphone, they also remove troubles of microphone choice and placement, as well as leakage into the mic from nearby sound sources.
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A passive DI doesn’t require phantom power, further simplifying their operation.
Active DIs can be a good choice for matching the lower signals provided by passive instrument pickups.
However, many acoustic guitars and electric basses have on-board active electronics (“battery inside?”), making a passive DI a good match for their higher output levels.
Other applications include electronic keyboards, and various “Prosumer” line level playback devices with unbalanced outputs.
The connectors and switches on each end – which of course are all Switchcraft - are on anodized cerulean blue panels, recessed to protect their “short handle” pad and ground-lift switches.
One end has the usual pair of quarter-inch TS jacks for signal input and through-patching, with a 20 dB pad switch between. The other end has a male XLR and a pin 1 lift switch, plus a second quarter-inch through jack. The SC800’s rugged industrial design is an instant classic.
The specifications match the Jensen transformer inside, like the common mode rejection listed as greater than 100 dB at 60. Input impedance is greater than 150,000 Ohms at 1,000 Hz and output impedance is less than 170 Ohm.

Back panel view of the SC800. (Click to enlarge)
Stated phase response is less than 3 degrees at 20 Hz, less than 1 degree at 1,000 Hz and less than 16 degrees at 20,000 Hz. With a +4 dBu input (1.23 volts RMS), Total Harmonic Distortion (THD) is less than 0.05 percent at 20 Hz and less than 0.006 percent at 1,000 Hz.
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It has a maximum input level of +21 dBu, or 8.7 volts, which is where THD at 20 Hz rises to 1 percent. With the 20 dB pad engaged it will take a +41 dBu signal, enough for most small guitar amps if there’s also a speaker or resistive load attached.
Stated frequency response is 10 Hz to 40 kHz, plus or minus 0.3 dB, and stated voltage gain is minus 23 dB, or minus 43 dB with the 20 dB pad engaged.
Finally, the SC800 is hand-built in Chicago, Illinois.
Next I’ll take it to a few shows and try it on stage with a variety of audio inputs…
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Mark Frink is a long-time audio professional and is also associate editor of Live Sound International magazine.
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Friday, August 07, 2009
Recent Wireless Technology Series: Shure UHF-R Series
This is the latest installment in a series detailing the latest technology in wireless microphone systems. Click to read the other installments:
AKG DMS 700;
MIPRO ACT Digital;
Lectrosonics D4; and
Sennheiser 2000 Series.
The Shure UHF-R currently reigns as the most widely rented and specified wireless microphone system.
It’s been on the market for four years and has garnered legions of fans.
Let’s look at some of the features and technical specifications that make the UHF-R system tick.
Components
Receiver options include the single-channel UR4S and the dual-channel UR4D. Most often seen, of course, is the dual-channel variant.
Three transmitters are available, including the UR1 standard bodypack, UR1M micro bodypack and UR2 handheld.
The bodypack units are connected to lavaliere and headworn microphones via a TA4, and in the case of the smaller UR1M, a 3-pin Lemo can be ordered as an option.
The handheld transmitter can be outfitted with a wide variety of Shure mic capsules, ranging from the SM58 to the SM86 to the high-end KSM9.
This is indeed arguably one of the stronger aspects of this series – the direct compatibility with Shure’s industry-standard range of capsules.
The handheld transmitter is available in standard black finish as well as the more “sexy” satin nickel. In the past few years, the old rule that “microphones must be black” (otherwise they might reflect light into TV cameras and/or be noticed by the audience) has abated, so this additional finish option fits right in with the current preference of options.
Technical
The UHF-R system employs Shure’s proprietary and patented Audio Reference Compander system.

Single-channel UR4S and the dual-channel UR4D (click to enlarge)
Audio companding is a “necessary evil” in any analog wireless system, because otherwise there’s not enough dynamic range available in the link to provide an appropriate full-range musical signal.
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The Shure approach is very well regarded by engineers and artists alike for its natural sound quality. Dynamic range is specified as >105 dB, A-weighted, which is quite good - better than CD, in fact. Most high-quality wireless systems today have a similar specification for dynamic range.
Overall frequency response of the system is listed as 40 Hz -18 kHz +1, -3 dB. That’s a good specification and compares well to other high-quality analog systems.
In contrast, digital and digital hybrid systems extend a bit lower and a bit higher, with overall flatter response.
Nonetheless, the audio range of the UHF-R is more than enough to satisfy the demands of touring and installed sound markets, as evidenced by the high level of acceptance.

The UR2 handheld transmitter can be outfitted with a variety of Shure mic capsules(click to enlarge)
Transmitters in the UHF-R line allow for two different RF power settings, which I mentioned in an earlier post is something we’re seeing more.
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For the U.S. market, the UR1 bodypack transmitter can be switched between 10 mW and 100 mW. The UR1M microbodypack and the UR2 handheld can be switched between 10 mW and 50 mW.
For Europe, the handhelds and bodypacks all offer selectivity between 10 mW and 50 mW, to satisfy different regulations. The basic idea behind switchable power is that the user can choose between “low battery consumption” and “long range”.
The different in range between 10 mW and 100 mW (a 10 dB difference in power) is about 50 to 60 percent, which can be significant depending on the application.
And, of course, with all questions of range there are a vast number of external factors involved as well. I plan to get into this topic in more detail in a separate post – stay tuned.
The frequency ranges offered in the UHF-R package cover the entire available spectrum.

The UR1M micro bodypack (click to enlarge)
The G1 range covers the very low end of the available UHF spectrum, from 470 MHz - 530 MHz, followed by the H4 range from 518 MHz - 578 MHz.
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Note that these two ranges overlap a bit - a nice feature in some cases, but users should be aware that it is possible to have a transmitter at the top end of G1 step on one at the bottom of H4.
The J5 covers 578 MHz - 638 MHz and the L3 range covers 638 MHz - 698 MHz (the top of the current legal UHF range for wireless microphones in the U.S.). The system is also available in bands for Europe, Japan and Korea, among others.
Note that each of the U.S. ranges covers a fairly wide 60 MHz and provides nearly 2,400 frequencies in 25 kHz steps. The frequency ranges for Europe tune as wide as 75 MHz.
Networking
The UHF-R system was among the first to include complete networking capability, which greatly enhances the ability to quickly coordinate all frequencies and implement them throughout the receivers and transmitters.
Other companies have followed suit: Sennheiser with the Net1 system, and more recently, the 2000 Series; AKG with the DMS700, and Audio Technica.
Most, if not all, professional wireless systems will have these features within a few more years, but Shure was very wise to take such a comprehensive approach early.
Pricing
The UR4D receiver typically sells for about $2,800, while the standard beltpack transmitters are about $800 and handheld transmitters range from about $900 to about $1,800, depending on the capsule chosen.
These prices compare directly to the Sennheiser 2000 Series (see previous post) and thus are in the “upper middle” range. The Sennheiser 3000 and 5000 Series systems are considerably more expensive, while the Lectrosonics 400 Series and AKG 700 are somewhat less expensive.
Conclusion
It is obvious that the UHF-R system from Shure is well thought-out, with an excellent set of features and with a high level of quality that satisfies professional customers in many markets.
Shure did its homework when designing this system and it has paid off, and most certainly will continue to do so.
Find out more about the Shure UHF-R Series here.
Signing off for now…
Mike Wireless
Mike Wireless is the nom de plume of a long-time RF geek devoted to better entertainment wireless system practices the world over.
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More posts by Mike Wireless:
The Myths of Wireless System Transmitter Power
Latest Wireless Series #4: Inside the Sennheiser 2000 Series
The RF Spectrum Before & After The “Big Day”
Latest Wireless Series #3: Inside The Lectrosonics D4
Latest Wireless Series #2: Inside The MIPRO ACT Digital
Latest Wireless Series #1: Inside The AKG DMS 700
Is The UHF Spectrum Going To Ease Up After June 12?
Change The Only Constant In Marketplace For Wireless System Spectrum
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Monday, July 20, 2009
Recent Wireless Technology Series: Sennheiser 2000 Series
This is the fourth in a series detailing the latest technology in wireless microphone systems. Click to read the other installments:
AKG DMS 700;
MIPRO ACT Digital; and
Lectrosonics D4.
Sennheiser has long been one of the top wireless system manufacturers, respected for solid technology, high audio quality and a wide range of products, all evidenced by a long and varied user list.
Earlier this year at the NAB (National Association of Broadcasters) show in Las Vegas, Sennheiser introduced the 2000 Series, its latest wireless system development.
Pricing of the new 2000 Series is in line with the Shure UHF-R Series, and it now represents the “middle” of the Sennheiser line, with the 5000 and 3000 Series at the top, followed by the 2000 Series, and then the evolution Series, and finally, the Freeport system that’s primarily marketed in MI stores.
Components
The 2000 Series lineup includes the following transmitters: SK 2000 belt pack, SKP 2000 plug-on, SKM 2000 handheld, SR 2000 in-ear monitoring (IEM) and SR 2050 dual IEM unit.
Receiver choices are comprised of EM 2000 single-channel rack unit, EM 2050 dual-channel rack unit, and EK 2000 camera mount unit, and the EK 2000 IEM beltpack receiver.
Additionally, each generation of evolution wireless is compatible with the 2000 Series line, as are capsules and accessories.
Technology
The EM 2050, likely to be the standard receiver in the 2000 Series, provides up to 75 MHz switching bandwidth and 3,000 frequencies to choose from, with 25 kHz steps. This is compared to 36 MHz with 1,440 frequencies for the popular evolution G2 Series and 42 MHz with 1,680 frequencies in the newer G3 Series.
Therefore, the 2000 Series offers quite a wide tuning range and plenty of frequencies to choose from, which can be a real advantage in today’s challenging and changing RF environment.

Sennheiser EM 2050 dual-channel rack unit (click to enlarge)
All 2000 Series transmitters (U.S. models only) offer switchable RF output power levels at 10, 30, 50 and 100 mW, to better meet the needs of different applications.
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This is a nice feature, and it’s beginning to become more common.
One of the main features in the 2000 Series is the Ethernet port and WSM software – both used for system management, monitoring and control.
They offer the user quick access to all system parameters and conditions, along with the ability to quickly implement an entire frequency plan.
Interestingly, even as we see more digital and “hybrid” wireless systems coming into the market, the 2000 Series utilizes Sennheiser’s HDX analog companding technology that was introduced about 10 years ago with the evolution G1 Series. Don’t get me wrong, though, because this is not a detriment.

Sennheiser SKM 2000 handheld transmitter (click to enlarge)
The primary advantage of analog companding is that it avoids any latency – all digital or hybrid systems incur some signal delay, due mostly to the A/D conversion process.
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A disadvantage is some signal degradation, particularly with certain types of source material - high-frequency transients are particularly challenging for companders.
Nevertheless, HDX technology has pleased many users and no doubt will continue to do so.

Sennheiser SK 2000 body pack transmitter (click to enlarge)
Features
Like many other comprehensive wireless microphone systems, the 2000 Series allows for synchronization of the transmitters via an IR port on the receivers.
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This certainly speeds setup when compared to manually transferring frequencies.
The 2000 Series rack-mount receivers can also be daisy chained up to 16 channels without the need for external antenna distribution.
One additional nice feature on the EK 2000 portable receiver for video users is the addition of diversity (a change from the evolution G2 EK 100, 300 and 500). This should provide greater range and resistance to dropouts.
Also, the EK 2000 portable receiver offers synchronization of the transmitters via an IR port, a very convenient feature for on the go frequency coordination.
Finally, the SK 2000 body pack transmitters use the Lemo 3 connector, shared by Sennheiser’s SK 3063, SK 50, SK 5012, etc.
Thus, if you have lav mics already wired for these upper-tier transmitters, you can use them directly with the new 2000 Series bodypack.
A unique feature with the SK 2000 for guitarists is a built in guitar tuner and a virtual cable emulation that gives the user the opportunity to select a preferable impedance.

Sennheiser SKP 2000 plug-on microphone module (click to enlarge)
Pricing
As noted earlier, the 2000 Series is priced similarly to the Shure UHF-R Series – this probably isn’t an accident. The EM 2050 receiver can be found for a street price of $2,798, while the bodypack and handheld transmitters run at just under $800 each.
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This equates to a per-channel price (if counted in multiples of two) of about $2,200.
Conclusions
While certainly not the most expensive system out there, the 2000 Series is a serious investment when compared to the evolution Series as well as models from Audio-Technica, Lectrosonics and the aforementioned Shure.
Yet the fact remains that the 2000 Series offers a number of very nice features – particularly the stuff that makes the user’s job of setup and monitoring easy and quick. Sennheiser retains its place among the leaders of wireless technology, and the 2000 Series should make a lot of pro users (with pro budgets) happy.
Find out more about the Sennheiser 2000 Series here.
Signing off for now…
Mike Wireless
Mike Wireless is the nom de plume of a long-time RF geek devoted to better entertainment wireless system practices the world over.
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More posts by Mike Wireless:
The RF Spectrum Before & After The “Big Day”
Latest Wireless Series #3: Inside The Lectrosonics D4
Latest Wireless Series #2: Inside The MIPRO ACT Digital
Latest Wireless Series #1: Inside The AKG DMS 700
Is The UHF Spectrum Going To Ease Up After June 12?
Change The Only Constant In Marketplace For Wireless System Spectrum
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Thursday, July 02, 2009
Product Review: DiGiCo SD8 Digital Mixing Console
What can this new digital console offer your church sound system?
DiGiCo has come out with a steady stream of digital consoles utilized in a wide range of sound reinforcement applications, including larger church sound reinforcement systems.
The company’s most recent development, the SD8 digital console, has 60 stereo or mono channels available (the equivalent of 120 channels of DSP)—plenty for most churches—and certain key functions are available on all channels all the time, in contrast with some digital consoles that don’t offer consistent processing on all channels.
How does it work, and how well does it work? Is it worth a look?
Click here to check out John F. McJunkin’s in-depth audio product review of the DiGiCo SD8 digital console featured in the June 2009 issue of Church Production.
Other recent Church Production reviews:
Peavey Sanctuary S-32 Console
DiGiCo SD8 Digital Console
Sony UWP Wireless System
JBL EON 515 Loudspeaker
Audio-Technica AT2020USB Microphone
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Monday, April 20, 2009
Barry’s Toolkit: Inside The Peluso P12 Tube Condenser Microphone
An audition on a recent studio project by Pat Benatar for this microphone that's inspired by and styled after AKG's legendary C12
The “love” is obvious in every microphone John Peluso turns out by hand at his Floyd County, Virginia lab. Peluso’s 29 years of experience repairing and restoring all brands of vintage and modern microphones, his extensive stock of vintage parts, and diaphragm repair lab make him precisely qualified to produce his modern classics.
The P12 is inspired by and styled after AKG’s legendary C12 and is indicative of his whole line of microphones with its custom design and handcrafted quality. He uses only the finest components and builds the mics using the Austrian manufacturer’s original published specifications and frequency response graphs as design goals. AKG’s original testing standards are employed for quality control.
Peluso obtains all mic parts, raw materials and machine-shop work from around the world: the silver-foil output capacitors, resistors and capsule Mylar come from Germany; outer cases and capsule back plates are machined in China; and the output transformer is a custom remake of the T-14. The power supply and transformer are made overseas.
The P12 has nine polar patterns switchable at the power supply. Peluso controls cost with a standardized power supply design that will also run some of the other mics he makes—such as another favorite of mine, the 2247 SE large diaphragm model.
The P12 comes as a complete kit packed in an oversized flight case with a velvet-lined, wooden mic storage box, power supply, seven-conductor cable, and shock mount. Like all Peluso mics, its’ covered by a three-year limited warranty
In the Studio
My latest studio project is a new album by Pat Benatar and it afforded me the luxury of auditioning several different vocal mics they’ll use at their own studio. She was as curious about the differences as her husband and producer Neil Giraldo, and she “nailed” each mic’s general character, the differences they each made to her sound, and the way they influenced the emotion of her vocal performances.
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With this much riding on the sound and feeling of the microphone, Neil and Patty wanted to match the mic’s sound exactly to each song. They’ll probably use at least four different mics during the recording of the lead vocals for this record.
The P12 faired well under this kind of scrutiny—as expected, it reproduced airy high frequency sonic qualities along with a large low frequency stage like the original AKG C12 quite well.
Compared to some of the other large diaphragm tube condensers in their collection, the P12 had a taller sound—more top without shrillness and deeper bass without boom.
Neil Giraldo explained his preference for the P12 this way: “Vocals recorded on the P12 have a very smooth top end that sits nicely within the track—especially when singing at softer levels and in songs the require an intimate setting.”
As “mastering insurance” on one song, I will de-ess a couple of strong “Ss” but they were not overly sibilant.
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I do this so that the mastering engineer may brighten the mixes as he sees fit without resorting to de-essing the entire stereo mix. (I hate that).
In addition, the extended low frequency of the P12 necessitates a pop screen and my Pete’s Place Blast Pad Filter worked great. The included shock mount isolated the mic’s body well against subsonic bumps and foot tapping.
Across town at David Gamson’s studio (David, in a past life, was half of the pop duo Scritti Politti), the P12 sounded great on an acoustic guitar.
David reports: “It had a nice glassy top end without sounding brittle and, because of the slight mid-range scoop in the frequency response of the microphone, very little carving out of the midrange was necessary to get the acoustic guitar to sit nicely in the mix.”
He continues with: “we tried the P12 on a male voice-over session and found that it has a fantastic presence without sounding thin or harsh—very little post processing was necessary.”
David’s assessment lines up with what I’ve found that, just like the C12, the P12 tends toward a hyped top end but retains warmth.
He goes on with: “the vocal had an ‘in your face’ quality coming straight off the mic making it a good call for recording a female pop singer. “
I used the P12 in the same ways I would use a good sounding AKG C12, and found it more reliable and consistent than most vintage mics.
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The P12 excels in vocal recording due to it’s open and smooth sound. It will not flatten out (compress) under loud moments.
In a mix with heavily processed guitars, synths, and drum machines, vocals recorded with the P12 sound present and natural without having to resort to generous, additional equalization—usually the case with recordings made with some other tube microphones.
The P12 performs brilliantly through a wide dynamic range making it the first choice especially for singers looking for bright sound without harshness.
It sells for $1,499.99 MSRP directly from www.pelusomicrophonelab.com
Peluso P12 Vacuum Tube Microphone Specs
Type: Condenser Pressure Gradient w/ 34mm edge-terminated capsule
Frequency Range: 20 Hz/24 kHz
Polar Pattern: 9 - Switchable from omni- to bi-directional
Sensitivity: 11mv/pa
Impedance: 200 ohms
SPL: 136 dB
Equivalent Noise: 15 dB (A-weighted)
Tube Type: 6072A-M
Power Requirements: Dedicated power supply
Size: 45mm x 240 mm
Weight: 750 g
Price: $1499.75
Barry Rudolph is a veteran L.A.-based recording engineer as well as a noted writer on recording topics. Visit his website at www.barryrudolph.com
More Reviews & Articles By Barry Rudolph
Barry’s DAW Toolkit: Review Of The Novation Nocturn With Automap 3 Pro
Barry’s Recording Tips: Figure Of Eight Royer For Electric Guitars
Review Of The X-Tempo Pok DAW Wireless Footswitch Controller
Barry’s Toolkit Of Handy DAW Products
Recording Gear Hits At The 2009 Winter NAMM Show
Working At Recording Success: Taking Elemental Steps Can Make All The Difference
Recording Tip: Successfully Dealing With A Dead Room
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Tuesday, April 14, 2009
Me & The Audio-Technica 4000 Series (Or How A Single Wireless System Can Make An Impact)
Often, the best place to start in terms of improvement is at the source; in other words, the microphones capturing the performance must be of quality, or the rest, literally, is just noise
Times are tight. The last thing most of us want to do right now is spend scarce church funds on new audio equipment.
The solution, however, lies not in completely shutting the door, but in setting clear priorities, choosing carefully, and investing wisely. In doing a complete inventory of what we have that works well, works OK, and works horribly to the point of seriously compromising our primary goal of the Word being heard.
Because right now, as we grapple with life’s rougher seas, hearing the Word is paramount.
With that in mind, I have a recommendation for any of you who might be in serious need of a new wireless microphone system, or are looking for a worthwhile upgrade.
A couple of months ago, Audio-Technica sent me a 4000 Series Artist Elite wireless microphone system, both for evaluation in terms of my “day job” as a systems integrator and also to take for a test-drive at my church, where I serve on the technical team in support of both contemporary and traditional worship services.
While the 4000 Series was introduced over five years ago, it’s still formidable in every aspect, including my two primary factors: sound quality and rock-solid RF performance. If it doesn’t sound good, forget it. If it’s plagued by drop-outs and other anomalies, double forget it.
The 4000 Series also has a deep set of really smart features, and with an MSRP (retail/list price) that starts at $1,059 and ranges to $1,939 (and estimated “street prices” - what you’ll likely end up really paying - ranging from $769 to $1,399), it better be good. No, it better be great.
In fact, it better make a difference.
What I found in my experience with the 4000 Series is that it does indeed make a difference, both in terms of vocal sound quality as well as making life easier for system operators. Here’s why.
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Overview
With 200 channels per band and 2 bands available, 541.500 - 566.375 MHz (TV Channels 25 - 30) and 655.500 - 680.375 MHz (TV Channels 44 – 49) this offering from Audio-Technica will fit your every wireless need, from a single mic for a talking head to a complete array of wireless for a full blown production.
Note that the operating frequency range completely avoids the troublesome 700 MHz band that’s the subject of so much uncertainty now, and likely, in the future. (You can read A-T’s latest statement on the 700 MHz issue here.)
The specific package I evaluated is the A-T AEW-4313, consisting of an AEW-R4100 half-space rack-mount receiver, AEW-T1000 UniPak belt pack transmitter, and AEW-T3300 cardioid condenser handheld microphone/transmitter.
Additionally I was provided with an AT892 MicroSet omnidirectional condenser headworn microphone. The company offers nine different transmitter options for the R4100 receiver, accounting for the price range quoted above.
My specific package has a list price of $1,739 ($1,259 estimated street price), with the additional AT892 mic starting at $439 list ($299 estimated street price).
Right out of the box, I liked what I saw. First, and albeit it’s a small thing, the receiver unit comes with a standard “computer style” power cord - NO wall wart or bulky inline converter in the cord.
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Too many times I’ve had to search for a power cable for a receiver that of course has a voltage and current requirement that is not common, in addition to having to figure out if the connector on the power supply is tip “+” or “-”.
Beyond the power cord, the initial impression was great - everything is included to rack mount (dual or single) or set on a counter. Also in the package is a link cable to connect multiple units together to use the IntelliScan feature that automatically determines and sets the best available frequencies on all linked receivers.
In our shop, I plugged in the receiver’s power cable and fed it to an input on a small portable system, put the batteries in the transmitters, turned it all on, and everything worked flawlessly right out of the box.
An early observation is the feel of both the handheld and belt pack transmitters. The handheld “metal case” is weighty - the good kind of weighty - not heavy as in “I don’t want to hold this for two hours” and not light as in “this feels like a flimsy piece of junk.”
The belt pack was the same, and in terms of physical size, it’s big enough that you feel something secured to your belt but not so big that you feel like you’ve grown a new appendage.
Dual Service Use
To get a real feel of how this system actually works and sounds, I next deployed it on my Sunday morning gig, starting by using the AEW-T3300 handheld transmitter with the lead vocalist for our contemporary worship services.
The transmitter features the same cardioid condenser element used in the very well-received AT4033 microphone, and as I listened to the vocalist, the overall sonic signature was pristine.
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This can be credited in part to the mic element, but I also believe the dual compander circuitry of the 4000 Series wireless, which processes high and low frequencies separately, enhances the exceptional clarity I heard in the vocal.
Later, for traditional services, I asked the pastor to try out the UniPak belt pack transmitter and MicroSet condenser headworn microphone. He was hesitant, having previously switched to a headset mic that took him months to get adjusted “just right”. As a result, he didn’t want to change to a different headset - very understandable.
But I persisted and he eventually consented, and pretty quickly, he’d gotten the A-T headset at a comfortable position, to the point where he used it throughout the entire service.
As is normal with headset mics, I still needed to do a final adjustment on the mic placement - he’d inadvertently positioned it so close to his face that the mic element tapped the corner of his mouth when he talked more demonstrably.
Further, the pastor really liked the belt pack transmitter, both the way it fit snuggly and didn’t slide around, as well as the sliding door that protects the unit’s control buttons. A few years ago, he had the misfortune of accidentally leaving his transmitter on while visiting the restroom, so he’s exceptionally dogged about being sure it’s turned off before leaving the platform.
Related, of course, he’s also notorious for then forgetting to turn the transmitter back on when he returns, frantically scrambling for the button while on stage. The dual action presented by the sliding door on the A-T belt pack transmitter makes the on/off process more concrete, which he said helped him better keep in mind whether the unit was on or off.
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Further, he no longer worried about the belt pack accidentally being turned off when he moved around on the platform.
Later I realized that the lock out feature on the transmitter would be another benefit in this regard, allowing the system operator to lock the power on while allowing muting capability. This allows the user to “go silent” for private moments without having to power down, and more painfully, wait for the transmitter to power back up.
Feature Rich
Following services, I sat down with the product manual and went through the feature set, reading and then pushing buttons on the system, quickly getting comfortable and familiar with all operating parameters. It wasn’t hard, and the manual is concise and well written. You can read about all of the features and specifications here, but I’d like to highlight a couple of them.
IntelliScan - In basic terms, IntelliScan allows the easy linking of numerous 4000 (or 5000) Series systems together. One system is set as the “master” while the others in the loop are incorporated in the network as well.
Push a button on the master receiver, and IntelliScan calculates the best frequencies for all receivers in the loop by using a built-in frequency plan that is mapped against the local interference sources it detects.
Presets & Control - How many times have you seen the tacky colored tape on wireless mics in order to keep them organized? (And yes, I can say it’s tacky because I’ve used this method.)
The 4000 Series addresses this with an easy-to-use preset function. A menu permits the storing of up to five different user-definable configurations, as well as customized names, can be created and stored for presets 1–5. All of this information can be clearly seen on the transmitter’s LCD display.
Once a preset is established, a digitally encoded tone communicates the transmitter data for receiver display. Both receiver and transmitter also display the transmitter’s battery, mute, locked status.
This is really handy in applications like mine, where we have two separate and often very different services using the same wireless resources.
For example, a preset for “Mary,” our operatic soprano who sings in the traditional service, needs a -6 dB cut to avoid distortion that rivals a guitar pedal. She also can’t be given any control over power or mute, because invariably, she sets it wrong.
Meanwhile, a preset for “Beth,” a vocalist in our contemporary services with a softer voice, is set to provide +6 dB in level. She’s much more reliable on the controls, so her preset allows her to mute when she wants. (But not power off.)
As a result, two diverse participants can share the same wireless system, and other than loading the preset prior to each one’s service, I’m off the hook.
It’s so great that it counters the nostalgia I’m invariably going to feel when thinking about that yellow, blue, green, pink and red tape wrapped around our mics in the “good old days.”
Speaking of “wrapped,” it’s time for me to wrap up my remarks.
As I noted at the outset, few of us are comfortable thinking about investing big money at this point in time. And if we’re honest, it’s these restrictions that make us work harder and think smarter.
When we do this, we realize that sometimes the biggest improvements in a church sound system can come from smaller packages. And often, the best place to start in terms of improvement is at the source; in other words, the microphones capturing the performance must be of quality, or the rest, literally, is just noise.
If you’re thinking along these lines, consider upgrading your wireless. As I can testify with the Audio-Technica 4000 Series system, it’s a solid place to start in terms of making a big difference.
Gary Zandstra is a professional AV systems integrator and has been involved with sound at his church for more than 25 years.
More Church Sound articles by Gary Zandstra:
Testing Cables Is Essential To Solid Church Sound System Performance
Two Simple Yet Vital Tools Of The Trade For Church Sound Operators
Also check out:
Basic & Essential: What You Need To Know About Wireless Systems
And:
Be sure to enter the PSW Sweepstakes (click here) for the chance to win a premium Audio-Technica BP4025 ($749 MSRP) or an Audio-Technica AT8022 stereo microphone ($499 MSRP)
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Wednesday, March 18, 2009
Barry’s DAW Toolkit: Review Of The Novation Nocturn With Automap 3 Pro
If you've spent any time at all using plug-ins to coax a great-sounding mix from "inside the box," Nocturn is like manna from DAW hardware heaven!
My DAW desktop surface is more crowded than ever—-and I’m loving it! I just added the new Novation Nocturn Plug-in Controller to my small army of DAW accessories.
As DAW plug-ins become more complex and intense, their GUIs (graphical user interface) have become more jam packed with more parameter controls than ever! And in order to keep their GUI’s screen size to a minimum, the controls and buttons keep getting smaller and smaller.
Even though I get new glasses often, the controls are not getting any easier to find, touch and adjust with a mouse. Plus during the course of working many hours adjusting plug-in after plug-in, my right hand and arm get very sore.
If you’ve spent any time at all using plug-ins to coax a great-sounding mix from “inside the box,” Nocturn is like manna from DAW hardware heaven! “At last, at last, great almighty, free at last!” my carpal tunnels rejoice!
Novation Nocturn
Nocturn is a dedicated controller surface for instant and intelligent control of any automatable plug-in within every major sequencer/DAW, including Pro Tools, on both Macs and PCs. It’s totally compatible with VST, AU, RTAS and TDM host systems/computers.
Ultra low in profile, Nocturn measures 9.41 by 5.39 inches and lies flat on any desktop—I have mine between my QWERTY keyboard and monitor.
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It connects via a side-mounted USB spigot and has eight, touch-sensitive continuous encoders each ringed with 11 red LEDs. These encoder LEDs provide visual feedback that the parameters you’re adjusting on Nocturn are changing in the plug-in. It’s cool to see them change after you’ve automated a parameter(s) too.
In addition, there are eight large backlit large push buttons, a 45mm crossfader, and a Speed Dial for instant access to any plug-in parameter—just hover over any parameter on the plug’s GUI with the mouse and adjust it with Speed Dial. There are also eight more lighted buttons specific to using Nocturn and its sidekick, the Automap 3 Pro software.
Nocturn uses Automap 3 Pro software to automatically assign plug-in parameters to each of the controller knobs and buttons.

Nocturn Automap 3
To set Automap 3 to work programming Nocturn before launching Pro Tools, I called up the Plug-In Manager software to scan all my plugs and map them.
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The software creates a copy of the plug-in software and adds its wrapper. This wrapper adds Nocturn’s frame around the plug’s GUI so you know what plug has been wrapped when you insert it into your session.
Now when I call up any wrapped plug-in, a transparent Control Map Window appears along side the plug’s GUI to show which Nocturn controllers control which parameters on the focused plug-in.
The Control Map appears and disappears using Nocturn’s View button.
Nocturn’s GUI has as many layers available as there are map-able parameters in a plug-in. You can also repeat parameters on each page (such as the Bypass button or parameters that you would always adjust together such as frequency and boost/cut on an EQ) just use the Add A Page function—for as many (or few) parameters as you want to control.
I found this whole scheme to work amazingly well considering all the quirky developer differences between all the plug-ins out there—I know because I have hundreds of plug-ins! Auto Map 3 Pro works very well in mapping all the parameters of plug-ins with dozens of automatable controls.
Things are not perfect however. I see that most of the time the Bypass In/out button of a plug-in is mapped to a knob. It works fine but I prefer the lower left button as my standardized location for bypass.
No problemo as Automap 3 Pro has a “learn” function where any button or control knob can be assigned to any function you want.
You can drag and drop knobs to buttons within the Control Map Window or simply turn or push the button on the plug’s GUI and its operation is instantly mirrored on Nocturn. Once you save your changes as part of the plug-in’s default map, you’re done—it can’t be any simpler than this!
Automap 3 also categorizes all your control maps and a simple browsing facility lets you review all open plug-ins to quickly switch to control any of them. Automap 3 also supports standard MIDI and HUI protocols for controlling and assigning MIDI parameters to a hardware MIDI device, any non-automatable plug-ins, or mixer control in your sequencer.
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A growing number of pre-made maps will be available to download from www.novationmusic.com
Using Nocturn
Without a console or console-like controller, I mix in the box these days using a large Pro Tools HD Accel rig. With these accessories and especially Nocturn, it is wonderful to get back some tactile sensation when mixing—adjusting volume, EQs, compressors, reverbs etc.
I often automate plug-ins for both effect treatments and fixing poor sounding recordings. Be able to twist an EQ knob or move a real fader for vocal rides are about the only things I miss from old school mixing. FaderPort gives me the fader and Nocturn everything else—I have no excuses now for better mixes!!
So, in a way, I’m reconnecting with my “roots” using modern technology and Nocturn helps to make it happen. I’ve been adding and modifying new maps everyday of the plugs I often use and Automap 3 Pro has got me covered already with reverbs and other multi-effect processor plug-ins.
So get back to mixing using real controls and keep the mousing to a minimum! Novation Nocturn sells for $249.99 (MSRP). It ships with version 2.1 software and buyers can download free version 3. Automap 3.0 PRO is a payable upgrade, at $29.99 via www.focusrite-estore.com. The differences are shown at: http://www.novationmusic.com/products/software/automap/
Check http://www.novationmusic.com for much more.
Direct link to purchase Novation Nocturn.
More Reviews & Articles By Barry Rudolph
Review Of The X-Tempo Pok DAW Wireless Footswitch Controller
Barry’s Toolkit Of Handy DAW Products
Recording Gear Hits At The 2009 Winter NAMM Show
Working At Recording Success: Taking Elemental Steps Can Make All The Difference
Recording Tip: Successfully Dealing With A Dead Room
Related PSW Recording Articles
Recording Microphone Techniques To Produce Warm, Spacious Stereo, By Bruce Bartlett
What Is Greatness? (In Recorded Music, That Is…), By Fletcher
How To Compare & Hear Differences In Microphones, By Ty Ford
Watch Out For The Big Lie: “We’ll Fix That During Mastering”, By Jackson B. Jackson
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Friday, March 06, 2009
Barry’s DAW Toolkit: Review Of The X-Tempo Pok DAW Wireless Footswitch Controller
Ever since I got one of the first prototypes about a year ago, one accessory that saves time, energy and makes work more fun is Pok.
I go for any “crutch” that gets me out of a work rut. You know - where you find yourself doing the same things over and over for each gig.
Even though I rely on certain gear and plug-ins to get “my sound,” it’s a good idea to try and take a fresh approach for each project. For me, using new gear, software or a new accessory are good ways to enliven my spirit.
And if the accessory also saves time and energy, then it becomes a “must have.”
Ever since I got one of the first prototypes about a year ago, one accessory that saves time, energy and makes work more fun is Pok.
Pok, from X-Tempo, is a wireless footswitch controller that transmits pre-programmed DAW keyboard shortcuts over a radio channel back to your computer.
Hands-free control over a DAW is an immediate winner for any musician or vocalist who wants to work alone or not rely on (or pay) someone else to operate the rig while he/she records.
Pok can also work for a DJ or live sound mixer when two hands are not enough.
When mixing in Pro Tools, my sitting posture is better because in order to have my left hand on the QWERTY keyboard, right hand on my Turbo TrackBall and my right foot in reach of Pok, I have to sit upright and straight ahead in my Miller Aeron chair.
Pok has two parts. The main footswitch box measures about twelve by seven inches and is made of tough ABS plastic with a clear anodized aluminum faceplate and steel base plate bottom cover. It’s available in graphite black or moon rock finishes.
There are eight, easy-working switches made by Alps that are rugged enough for foot taps but I wouldn’t stand on them.
Since Pok transmits only very short bursts of digital data, the unit runs for months on three AA batteries. There are also three LEDs that show: low battery, low signal strength and recent pedal activity.
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The second part of Pok is a USB receiver dongle that plugs into your computer. About the size of a memory stick, the Pok footswitch and receiver uses the USB wireless standard on 2.483Ghz.
Pok requires no driver installation and supports MAC or PCs running Digital Performer, Logic, Cubase, GarageBand and Ableton Live. Although it comes with pre-programmed profile to work with Pro Tools as default, the included Pok Editor software lets you remap the footswitches in any way you want for any computer program you want.
(Be sure to check out the PSW Photo Gallery highlighting other handy, affordable DAW products in Barry’s toolkit.)
Don’t want to learn the editor? You can download many free pre-configured profiles for many other DAW and music creation programs like Sonar, Reason, Acid Pro 6, Audition, Quicktime, and iTunes.
I use the Pok every day: for punching in, dropping markers and stopping the song when I’m not in reach of the space bar. Each footswitch can store up to three commands with switch eight acting as a shift/function key. Up to 22 commands per profile are possible.
There is no latency or uncertain operation making it perfect for operating your rig from out in the studio while overdubbing vocals or acoustic guitar. The same goes for punching in—getting into record is at the speed of light.
The manual says Pok operates within 80 feet of your DAW—in reality I was running my rig from three houses down the street.
I’m not sure of the utility of that ‘feature’ but it is good to know you have a solid radio link—walls, metallic objects and electrical wiring don’t matter. It’s more solid than my Verizon cell phone!
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During initial setup the USB receiver dongle finds and pairs up with the footswitch creating a new digital radio link. This means you can use more than one Pok in the same room—either on a second computer or on the same computer for second program.
When I’m working with a nervous artist, after I’ve gotten a record level, I leave the studio and give them the Pok footswitch. The keys are well marked using the included set of removable adhesive labels so the artist can start, stop rewind record etc. I get a break and the artist gets to go it alone. What’s not to love here?!?!
Pok sells for $599 MSRP complete with carrying case, software, two-year warranty and manual. For more information, go to: www.x-tempozone.com.
ALSO, the folks at X-Tempo are offering a discount to ProSoundWeb readers on the purchase of a Pok unit. Use Coupon Code “barry” And Buy Pok At A Special Discount Price At X-Tempo’s Web Site!
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