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Handy Phone Recording Rig Improves Webinar Audio

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Sometimes high fidelity takes a back seat to content. In this case, a company that does a webinar series asked for my help in cleaning up and streamlining their audio. I heard their previous webcast and, wow was it bad! Way too much data compression. Intelligibility was suffering. Surely I could do better.

I was asked to record the panelists in different cities in advance of the event, sequence their audio with other voice recordings and create a master .wav file that would be the bulk of the presentation. There would be a live Q&A and a pre-recorded good bye. The webinar would run a little less than an hour. I was also hired to be on site for the show to help the producer.

There are lots of ways to get audio off a phone line. Perhaps the easiest is a phone hybrid and I just happened to have a JK AutoHybrid in my bag of tricks. The phone line plugs into the hybrid and although it has XLR in and out, it has no level adjustments.

Between the hybrid and Pro Tools, I put a Sound Devices MixPre-D so that I could adjust levels as needed. I could have used the MixPre-D's USB out to get into the computer, but chose my RME ADI-8 DS A/D converter. It uses ADAT optical to get into my Digidesign DIGI003R.

In this particular situation, my voice would not be heard so I only used the hybrid as a way to grab the phone audio and bring it into the Pro Audio world. I called each interviewee first to make sure they had a line that sounded good enough. One was on a cell phone and the audio was pretty messy, but she was able to get to a hard line.

A "hard line", BTW is, historically, a copper connection consisting of an overhead or underground pair of wires that bring the phone service into your home or office. Copper is an expensive, aging infrastructure in the USA. As the lines and insulation deteriorate, water from rain seeps in and compromises the connection. Bugs, birds and squirrels also do their unknowing best to destroy the system. The result is a noisy phone line.

Copper lines have worked well over the years, digging into a bundle of copper cables to find problem pairs and fix them is expensive and time consuming. Where ever possible, copper is being replaced by fiber optic cable. Fiber optic cable offers many advantages, but can also be physically compromised.

While the number of analog conversations coming down a copper pair is normally limited to one single conversation, fiber optic lines with digital audio can accommodate many more. That makes fiber optic delivery more economical for the phone companies once they suck up the cost of replacing the copper. Using data compression, the phone companies can fit even more conversations on the line, but as they increase the number of conversations, the fidelity of each one deteriorates. Think of what happens to CD audio when you make low bit rate mp3 files for your personal music device.

You're also aware of this phenomenon on today's cell phones; mushy, swishy artifacts that sometimes make the conversation difficult or impossible to understand. Although we forget, analog cell phones sounded a lot better. My point here is that, these days, phones that plug into a wall may not be hard lines. In some cases, a good cell phone on a good day may sound better than a bad home or office phone on a bad day. If the problem is not in the telephone instruments themselves, sometimes just hanging up and recalling can result in a better connection because the call will be routed through a different path in the communications maze.

We thought to try SKYPE. I've used it in the past and have been very pleased with the quality of the audio, but unless the user has a headset that allows them to have the mic very close and a headphone to keep the computer audio from getting back into the mic, the results can be ugly. And, not all phone headsets are the same; some sound pretty good, others simply suck. The participants were not phone ranger types so I tried to make things as simple as possible.

After a quick call to each person to determine if the connection was good, I was satisfied that each person's phone would do the job, I made appointments for a short interview; about 20 minutes each. The recordings went off without surprise, but I did have to remember to mute my own phone from the phone handset when I recorded them to keep my voice and any noise out of the track.

After editing, I applied parametric EQ and iZotope RX3 Advanced Noise Reduction and Dereverb. I used Pro Tools' stock 7 band parametric EQ to scrape off any low or high frequency junk and to enhance the voice.

Seven bands provides a lot of sound shaping ability. The interface is quick and intuitive. Sometimes the voice would sound a bit thin. I used the EQ to add some upper bass or low midrange to warm the voice up. If the voice is a bit dull, adding energy around 3 kHz to 6 kHz will bring it back to life and dragging the points around in real time while hearing the changes works very well.

The powerful noise reduction in RX3 pulled some of the various low level noise out of the recording and it also
got rid of a lot of the ringing artifacts you hear on phone audio. Dereverb reduced the echo that frequently accompanies phone audio these days. That echo is pretty tight and at first I didn't really think much about there being an echo. After I inserted the plugin, I was surprised by how much it was taking out, and without further damaging the audio. No, the result was not studio quality, but it was obviously better than what I started with.


I put a compressor followed by a limiter in the Pro Tools two-mix to pack the audio; just enough to make it thick. At that point, I attached my Sound Devices 664 mixer/recorder to my Digidesign DIGI 003R and recorded my mix onto the 664. An odd choice perhaps, but the MacBook and webinar software I was using required external audio coming into a MacBook. The 664 makes a very nice playback device. A bit overkill, perhaps, but very solid. The client was very happy.  

Technique, Inc. © Copyright 2014 All Rights Reserved

Contact Ty Ford at www.tyford.com





ZOOM H5 - Somewhere Between H4 and H6

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This may be the longest review I’ve written in some time. No magazine I‘ve ever written for would allow me to put this much information in a review. There's just not enough space available. I felt obliged to cover the many features offered by the Zoom H5. If you begin to lose traction, swim to the edge of the pool and hold on until you get your breath. Then dive back in.

Don’t need six tracks that the Zoom H6 offers? (I know, who doesn’t need more tracks?) Like the idea of real level knobs instead of menu buttons? 

EXH-6 XLR/TRS Input Module
For $269, the Zoom H5 sits between the $229 H4n and $399 H6 and strikes me as a nifty little hand held 4-track recorder that’s capable of doing a number of odd jobs. Let’s see what it can do, and what it can’t. First off, no surprise - no SMPTE.

I was also sent the EXH-6 Dual XLR/TRS input module and SGH-6 Shotgun with furry, but not the MSH-6 Mid/Side mic or any other accessories. The H5 also takes the H6 modules. 

The H5 and all of the modules have a very solid feel; like professional tools. Metal bars wrap around the input gain controls to prevent them from being tweeked and to prevent them from being damaged in a fall.
SGH-6 Shotgun Module
MHS-6 Mid/Side Module

BEFORE YOU RECORD
Unlike many manuals, the 115 page H5 manual is actually well written and has a good balance of words and diagrams. You may think you know how things work, but you may find new powers and helpful capabilities that you weren’t aware of unless you read the manual. 

The audio interface input settings alone require some pondering. Lo-cut, comp/limiter, 12, 24 or 48V DC Phantom Power, pad, Mid/Side matrixing, monitor mixing, loop back, internal mono mixing and line output level make it obvious that the H5 is a feature laden box. If you get lost and things aren’t working, you can restore the default settings. 

Did I mention you should read the manual?

COMPRESSOR/LIMITER
Here’s a prime example of why you need to read the manual. There are three compressor and three limiter settings. The only adjustment you have is the input level. Used without some thought, your tracks will suck mightily. The specs in the manual are great for old schoolers who know what attack, release, ratio and threshold settings do. If you don't then my advice is simple; don't use the compressor/limiter or don't push the audio very far past the threshold until you know what that's going to sound like.

The “Standard Compressor” threshold is at -48.7 dB. It has a 9:1 ratio, a 7.2 mSec attack time and a 968 mSec release time. The “Compressor for Vocals” has a -8.4 dB threshold, a 16:1 ratio, a 1.8 mSec attack time and a 8.7 mSec release time. The “Compressor for Drums” has a -48.2 dB threshold, a 7:1 ratio, a 12.3 mSec attack time and a 947 mSec release.

The “Standard Limiter” has a -14.4 dB threshold, a 60:1 ratio, a 6.4 mSec attack time and a 528 mSec release. The “Concert Limiter” has a -13.8 dB threshold, a 32:1 ratio, a 1.9 mSec attack and an 8.5 mSec release. Finally, the “Studio Limiter” has a -12.0 dB threshold, an 8:1 ratio, a 6.5 mSec attack and a 423 mSec release. Much more friendly, but still watch the levels so you don't munch the audio. Many people like to leave dynamic range control for later. 

In my first attempt with the “Studio limiter”, I pushed the input levels a bit too far and heard the limiter action while tracking an acoustic guitar track. Next time I let the studio limiter just catch the peaks. I saw peak readings around -10 dB, maybe a little higher. The compressor or limiter you choose can be applied to either the X/Y or 1&2 inputs, or both. Respect the Comp/Limiter and it will respect your recordings! 

H5 Left Side with HS1 Camera Mount
ON THE SURFACE
The outside of the H5 is simple, but no less powerful than the inside. On the left side is an unbalanced 1/8” TRS Line Level out. The output is adjustable from -30dB (mic) to 0dB (line) in 5 dB increments. The 1/8” TRS stereo headphone jack is next and it’s loud enough to power my Sony MDR7506 headphones. 

A headphone volume rocker switch is next followed by a 5V DC USB jack and the On/Off/Hold switch. It allows the H5 to be turned on and off and it also has a HOLD function that locks out all button pushes but the ones on the RCH-5 hardwired remote control.
H5 Right Side

On the right side are the jack for the hardwired remote, an SD card slot, the menu button and menu navigation control. Obviously designed for right-handed (or right thumbed) people, the menu navigation controls quickly become natural to use. You push in and navigate with the toggle button and back out with the menu button. Simple and nicely done. 
H5 Back Side

On the back is a small speaker, the battery compartment for two AA batteries and a threaded nut for mounting the H5 on a camera stand or camera adapter like the HS1 or MA2. The HS1 allows the H5 to be mounted on a camera shoe. The MA2 acts as a handle which greatly reduces handling noise when you’re using the X/Y mics and is long enough to slip into most mic clips for mounting on a mic stand. These accessories are extra.

H5 Bottom


On the bottom of the H5 are two XLR/TRS jacks for inputs 1 and 2 for line, mic or instrument level.


FACE TIME
On the face of the H5 is a smallish LCD display that remains backlit for a while. The LCD display is packed with information and status icons. Pages 10-11 in the manual do a great job of sorting the bits out, but I’m guessing it’ll take a while before you know what each one means. 

A little homework here will make your experience with the H5 more valuable and productive. Translation, “Read the manual.” There are also record level controls for channels 1 and 2, record enable buttons for all channels, Stop, Play/Pause, Rew, FF and Record. These inputs have minor zipper noise as you change gain from 0 to 7 and back, but none between 7 and 10. 

MEMORY
The manual says the H5 supports up to 32GB SD cards. I liked the 2GB micro and SD adapter card that came with the H5. They were tucked away in the same cubby hole with the foam windscreen; so unobtrusive that I didn’t even notice it and reached out to ask where they were. The H5 wouldn’t format one of my 64GB SD cards, but did format a 16GB (30MB/sec Class 10) SanDisk Ultra.

There’s also an SD Card Performance Test, but the manual says that even an SD card that passes the test does not guarantee error free operation. If you have older Zoom recorders, or SD cards that were used in older H4 and lower (3, 2, 1) Zoom recorders, the H5 will probably read them. It will not read H6 files.

The manual warns that you should power down before inserting or removing cards to prevent data loss. I read that after a week of blissful ignorance without data loss and continued to do so, but don’t say I didn’t warn you. In MULTIFILE mode there is a rebuild feature that may help with corrupted projects or files. Underline MAY

Within the REC menu your choices are STEREOFILE and MULTIFILE. In STEREOFILE mode you get 16 and 24 bit 44.1, 48 and 96 kHz, plus MP3s at 48, 56, 64, 80, 96, 112, 128, 160, 192, 224, 256 and 320 kbps. I’m not sure you need all of those mp3 rates, but you got ‘em. In MULTIFILE mode, you’re limited to 16 and 24 bit 44.1 and 48 kHz. 

REC and REC MODE are two separate menu entries and it took a few minutes of fumbling around to notice. I think I would have put the STEREOFILE and MULTIFILE choices within the REC MODE menu, but once you know where they are and how they differ, you’ll be OK. 

There’s a directory structure for both MULTIFILE and STEREOFILE. Each of these directories has ten folders and audio can be recorded to whatever folder you choose. While recording, use the menu toggle control or PAUSE/PLAY button to drop marks which can then be fast forwarded or rewinded to during playback. 

H5 GINSU
But wait! There’s More! You can rename, normalize, trim and divide files. You can record automatically and adjust the start/stop threshold levels. There’s a two second pre-record buffer. In MULTIFILE, you can overdub and even if the overdubbed track already had a recorded file, the original will not be overwritten. Instead, a new file will be created. I don’t know that I’d want to work this way, but if you’re on a desert island with a good supply of AA batteries and a lot of time on your hands, you'd probably due quite well.

MULTIFILE also allows you to record 128 kbps MP3 voice memos attached to a file using the XY mics. You only get one memo per file. Each time you press the Record button, the previous memo is overwritten.

The H5 also supports conversion from .WAV to a variety of MP3 rates from 48 to 320 kbps. Handy if you wanted to record full .WAV files, but needed to export an .MP3 for transcription. The H5 doesn’t convert as fast as iTunes, but it’s not bad. When you back out to the directory with the Menu button, you see your new file named the same as the .WAV file, but with an .MP3 extension.


H5 Mid/Side Decoding Software
The Main H5 web page is here. There are ASIO Drivers for the PC Platform and Mid/Side Decoder plugins for Mac and PC on a special page on Zoom's site. They also have manual downloads and if there's a firmware upgrade, you'll also find that on the Zoom site. 

2 + 2 = 6
But are there really just four tracks? No, there are actually six! You can enable a “Backup-Record” pair of stereo tracks that are set to 12 dB below the X/Y mics in case something gets UNEXPECTEDLY LOUD. You don’t get backup tracks for Inputs 1 and 2, just the top L/R pair.

I found the X/Y mics a little brighter than I like, but actually pretty nice. If you like the Neumann KM184’s brightness, you’ll probably like the X/Y mics. If you prefer the KM84, then maybe not.

If you don’t like the small diameter stereo X/Y mics on the top of  the XYH-5 module, there’s a 1/8” TRS unbalanced stereo jack on the side of the XYH-5 module that takes mic to line level signals. 

I tried this with the 1/8” unbalanced stereo line output of my Sound Devices 442 mixer and had to back off the input gain control of the XYH-5 to 1 (out of 10) to keep from overdriving the H5. This made me notice a nice feature of the H5; the red record meters blink when the inputs are overdriven, bringing your attention to the fact that you need to do something. Thank you Zoom.

I tried plugging in an Audio-Technica AT8022 stereo mic to the 1/8” TRS jack. It’s an electret condenser that can be powered by external Phantom Power or an internal AA cell. I had to boost the XYH-5 gain dial to at least 7 to get a good level and that’s where the hiss began to creep in. If the ambient noise had been as high as average street level, it would have masked the hiss. A mic with more sensitivity (or a louder source) would require less heavy lifting from the XYH-5 module and result in lower noise. There’s no zipper noise when raising or lowering the XYH-5 mic gain control.

EXH-6 Module
EXH-6 MODULE
I swapped out the XYH-5 for the EXH-6 input module. It has two XLR/TRS connectors and a switchable 20 dB pad. Each input has a separate gain control. Again, using the Audio-Technica AT8022 mic in battery powered mode, this time with XLR connectors, I was able to get a very satisfactory recording with average noise and the EXH-6 gain controls set to 7-8 (out of 10). The EXH-6 sensitivity knobs don’t have the protective roll bars and although it does provide the 2.5V plug-in power some mics need, it doesn’t provide Phantom Power. Plug-in power is switchable in the Input/Output menu.

Next I plugged my semi-acoustic Telecaster into input 1 on the bottom  and an Audio-Technica AT897 battery-powered electret condenser shotgun into input 2 of the EXH-6, adjusted levels and had a blast recording a couple of tunes at 24-bit, 44.1 kHz, using the headphone feed from the H5 to power a set of Sony MDR7506 headphones. I pulled out the SD card and pulled the files into Pro Tools 10 using a card reader. I stuck the SansAmp PSA-1 amp simulator in the guitar track to make it a little nasty and did a quick mix. Good, solid sound.

The SGH-6 Shotgun Module is a mono mic and it records to one track. It’s OK, but would require the MA2 handle or something else to prevent handling noise. I guess you could mount it on a mic stand and set it up like a boom mic, but I’d want to be able to hear the output and that means a long headphone cord.

FEATURE SALAD
I probably wouldn't use all of the features on the H5, but you might like them. Take the tuner; no big deal, right? But what if its chromatic, guitar, bass, open A, open D, open E, open G and DADGAD? If you play in open tunings, that’s a plus. The tuner is not as fast as my Snark. It takes more time to re-register that I have moved on to the next string, but seems about as accurate. The tuner requires that you plug into inputs 1 or 2. 

PLAYBACK SPEED is an interesting feature. Its range is 50% to 150% with no pitch change. Glitching occurs pretty quickly as you slow down, as expected. Speeding up works better. In MULTIFILE mode the change applies only to each project. This is a per clip adjustment, not global. You select one clip at a time and change its playback speed with the thumbwheel. In STEREOFILE mode, the change is for all stereo files. 

KEY CONTROL changes the pitch but not the speed from six step below to six above normal pitch. In MULTIFILE mode, pitch change will affect the entire project. 

USB COMPUTER INTERFACE
The H5’s USB port allows you to access files from any SD cards inserted. Just remember to dismount the H5 from your computer properly or you’ll get a warning that you didn’t do that correctly. 

The H5 may also be used as a computer audio interface. I had to restart the H5 for it to be recognized and then I was led through a series of screens after selecting AUDIO INTERFACE. First, STEREO or MULTITRACK, then BUS POWERED or BATTERY POWERED. The H5 front panel informed me that it was going to record at 48kHz until I committed to a new Adobe Audition session at 44.1 kHz. Then the H5 switched to 44.1 kHz.

Using Adobe Audition 6, I chose bus powered and multitrack and plugged in my D28S Martin. The manual says that input 1&2 will take an instrument like a keyboard, but not a passive guitar pickup. I have a passive Pure Western Mini from K&K, comprised of three piezos. Zoom is correct! Plugging directly into the H5 resulted in serious clacking and quacking and finger squeeks were huge. I tried both the XLR/TRS inputs on the H5 and the EXH-6 module and got the same results. I suspected an impedance mismatch because the Pure Western Minis like seeing a very high input impedance. I inserted my Red-Eye Twin DI between the guitar and the H5 and tamed the squeaking and clacking.

Then I ran into some sync and sound issues. While recording through the H5 to Audition, the guitar sounded chorused, just going in. There were also some ticks that I usually associate with clock errors. The H5 was chosen as input and output device from the Mac Audio Control panel. On the Audition preferences page, I couldn’t choose anything else. Master Clock was H5, Source was Internal. I thought that might be the problem but was unable to change it. Interestingly, the sound was chorused while recording, but not while playing back. I did hear ticks during playback though. The same was true for both the H5 XLR/TRS inputs and the EXH-6 inputs and bus powered or battery powered made no difference.

I finally got rid of the ripples and ticks by starting a new Audition session and choosing Pro Tools Aggregate I/O for the output. I was hearing the record through the H5 headphones, but playback from Audition came from the Mac Tower. Choosing the Digidesign 003R as playback didn’t work either. It locked up the Audition transport. Pro Tools can still be picky and sticky even though it's supposed to play nice from V. 9 on. Choosing H5 allowed the transport to run, but the chorusing was back in playback. I decided to write off the idea of using the H5 with Audition or Pro Tools and moved on.

CAFE TROIA
I had friends, Kirby Storms and Don Armstrong, as Jazzmatazz playing at Cafe Troia, a local, upscale Italian restaurant. Flute and electric or acoustic guitar with some well recorded loops through a better than average PA.; Perfect! They were performing outside on a walled in patio. We got there just in time for the set and go a table about ten feet from the stage, with no one in between us and the action. There was the typical amount of table chatter, but it was mostly behind us. I didn’t bring headphones, so I wondered as I tasted Cafe Troia's amazing dark chocolate gelato, how much talking we’d get. The next day, during playback in my studio, I could definitely tell we were in a place with people talking in the background, but the H5 did a very nice job of capturing the PA sound. If you have a good PA mix and want a good simple recording of someone performing live through a PA, the H5 will give you that. 

OH POWER WHERE ART THOU?
The H5 runs on two AA batteries or external 5V USB power via a USB mini jack. You need to access the battery page and tell the H5 whether you’re using alkaline or rechargeable NiMH batteries. I put in two fresh Alkaline batteries. I put the machine into 6 track record with 48V DC phantom ON, using the two pairs of main tracks with the XYH-5 on top and a pair of Schoeps CMC641 on inputs 1 and 2 and the pair of backup tracks at 24-bit, 44.1 kHz and let it rip. 

An hour and 15 minutes later I came back to check and to plug in headphones and set them at an average listening level. Only one of the three battery power bars showed. I stopped recording briefly watched the timer take about 10 seconds to close the file. I then hit RECORD again because I wanted to completely dump the battery and find out what happened when the unit lost power while still in record. When I came back the batteries were dead. I transferred the files to my Mac. I got a total of ninety-three minutes and the recorder shut down properly, saving the last file I was recording at the time.

That’s somewhat shy of the 15 hours I had seen in other published reviews. I decided to waste another pair of alkaline batteries on a less strenuous test. Inputs 1 and 2 only, with the Schoeps and Phantom Power on, 24-bit 44.1 KHz. Headphones plugged in and raised to an average level. This time I got two hours and sixteen minutes before the batteries failed. Stunned by the unexpected under performance, I tossed a third pair of AA alkalines in, left the Phantom Power on but unplugged the headphones. Two hours and thirty-five minutes later the recorder stopped. The file was saved but still nowhere near 15 hours of recording. Could Phantom Power suck that much life out of a set of batteries? Maybe, but operating with Phantom Powered mics is not unusual.

Eight Hundred and Forty-Seven Minutes, Forty-Five and 282/1000 Seconds
Finally, in an effort to get to the bottom of the battery life issue, I put in another fresh pair of alkalines and recorded using only the XYH-5 mics that mount on top of the H5. Eight hundred and forty-seven minutes, forty-five point two eight seconds later, the batteries died. That’s a little over 14.1 hours of continuous recording at 24-bit, 44.1 kHz. Not quite 15 hours, but impressive, nonetheless. 

There were seven files, most of them 2GB in size and when butted together on the timeline, the playback was seamless. I’m going to presume that using the unbalanced 1/8” TRS mic/line input on the XYH-5 instead of its XY mics would yield the same record time.

IN CONCLUSION
It’s really difficult to believe how much Zoom packed into the H5 for how little it costs. I have no problems with the quality of sound, especially at its price point. I won’t be swapping out my Sound Devices 664 and buying an H5, but it is a very potent tool. Oh, right, and read the manual!

Technique, Inc. © 2014 All Rights Reserved
Contact Ty Ford at www.tyford.com

The Audio-Technica High Sensitivity AT 4080 Ribbon Mic

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Two Audio-Technica AT4080 in Blumlein
Ribbon microphones have been in service since the 1930s. RCA mics like the 44B and 77DX are now considered vintage. You can spend $1200 to $1500 or more for one. The trick is finding one in good shape because the original ribbons are relatively delicate. Not because of age. That’s just the way they were originally designed. Or, you can try a new Audio-Technica AT 4080 bi-directional (figure of eight), dual ribbon mic that streets for about $999. Seven years in the making, it boasts a 150dB SPL level and the sensitivity of a studio condenser.

Why bother with a ribbon? Well, for one thing, it sounds fundamentally different than dynamic or condenser mics. That’s why recording engineers have continued to use ribbon mics all along. Ribbon mics use a different principle of physics to capture sound than dynamic and condenser mics. Ribbons are rectangular strips of metal, usually aluminum, in a magnetic field. 


Signal flow is a direct result of the induction that occurs as the ribbon moves in the magnetic field. In dynamic and condenser mics, a round diaphragm is used. In dynamic mics, the diaphragm is attached to a coil suspended in a magnetic field. In a condenser mic, the diaphragm is one half of a capacitor. Instead of a magnetic field, the signal is generated by the varying voltage caused the diaphragm moving closer to and further from the other half of the capacitor.

The ribbon element in a ribbon mic has to have unique properties to sound good and be durable. Audio-Technica’s ribbon material is, “pure aluminum from a famous plating company.” That’s all AT would say. The other “magic” in a ribbon is the way in which it is crimped; a process by which the ribbon is shaped and imprinted with a textured pattern, so it holds its shape over time. Audio-Technica has a patent-pending MicroLinear ribbon imprint, which they believe results in a more durable ribbon with better resistance to lateral flexing. The AT 4080 comes with a 5-year warranty on both the mic and the ribbon. Not something you’d want to offer with a vintage mic. Plosive protection, always an issue with ribbons, is achieved with an ultrafine metallic mesh placed over the ribbon so as to be as acoustically invisible as possible.

Early ribbon mics, even the higher sensitivity 77DX, were still relatively low in sensitivity, requiring a lot of preamp gain and accompanying hiss. Audio-Technica solved that problem by using stronger magnets, longer, dual ribbons and a small, low noise amplifier on board the mic. What this works out to, for practical purposes is that the AT4080 is about 20 dB more sensitive than an ElectroVoice RE27 N/D. In fact it’s about 2 dB more sensitive than an AKG C414 B-ULS condenser mic in figure of eight pattern. So forget anything you thought about ribbon mics not having enough sensitivity. The onboard amp uses Phantom Power, so, if you’re old school enough to have learned not to run Phantom Power on ribbon mics, you’ll have to change your ways. 

The Audio Technica AT 4080 has a published frequency response of 20 Hz -18 kHz. Like most ribbon mics, it doesn’t have a presence peak, so it doesn’t sound as bright as a condenser mic. With few exceptions, e.g. the beyerdynamic M500, ribbon mics don’t have a lot going on above 10 kHz. The AT 4080 is down about -3 dB between 5 kHz and 10 kHz, but pops back up to 0 dB by 13 kHz before slowly fading away. The capsule, frequently called “the motor” in a ribbon mic, is well isolated from the body.

IN USE
After recording a voice track, I used a high pass filter at 30 Hz with an 18 dB/oct slope and a parametric EQ set fairly wide at 125 Hz and pulled down 6 dB. With that adjustment, the AT 4080 flattened out nicely. With further EQ adjustment, I was able to make the top end shine some; at least as much as a good dynamic mic. I think that in addition to working well on my baritone narrator voice, the AT 4080 would also smooth out edgy, unpleasant voices. The 150 SPL rating also makes it a good choice for the loudest screaming voices I can imagine.

Do you want one in your TV studio? That depends on what you want to record or broadcast and the acoustics of your space. Like most ribbon mics, the AT4080 is a bi-directional figure of eight. It picks up sound from the front and back (0 and 180 degrees), but very little at 90 degrees off axis. If you have two people in the same room and need to mic them both, theoretically, you could put them on the front and back sides of this mic and record both voices with one mic. They would have to have equally loud voices and be the same distance from the mic and close enough so as not to hear much of the room itself. If the two people had unequally loud voices, you could position the quieter voice closer, but now were entering that area of recording where things get iffy. If one voice required EQ and the other one didn’t, you’re stuck.

If you have a small booth with a large plate glass window behind the mic, although the nulls of the sides of the figure of eight are quite deep and effective, the backside of the figure of eight will pickup the bounce from the window, or any hard, reflective surface. If you turn the mic 90 degrees so that one of the nulls is aimed at the reflective surface, you’ll reduce the “roominess”, assuming that the back is not aimed at another hard surface.

If your facility uses a lot of compression and limiting to increase relative loudness, you may be aware that dynamics processors generate gritty distortion artifacts, especially when using relatively brighter condenser microphones with more aggressive presence peaks. The harder you hit the processing, the nastier it sounds. The Audio-Technica AT4080 should keep the sound from getting edgy as quickly.

If you record or broadcast music, a ribbon mic comes in very handy. It’s directionality is a very useful tool on a music sound stage; almost like a free isolation booth. With a little practice, you can capture one instrument nicely and position the mic to put another instrument in one of the mic’s two nulls to avoid picking up that instrument. Ribbon mics are frequent choices for horns, strings, banjos, accordions and edgy electric guitar amps because their softer high frequency response takes the edge off of those instruments. They can also do well as drum overheads if the space has been designed properly.

With two AT4080 ribbon mics, you can experiment with Blumlein Recording. Noted electronics engineer Alan Blumlein applied for a UK patent in 1931, describing the use of two figure of eight mics positioned coincidentally on the vertical axis (one directly over the other) and with each mic’s lobes positioned in the nulls of the other mic. So, as viewed from above, instead of a mono figure of eight, you get a stereo clover. Sound sources are usually placed at or in between the front lobes. Experimentation with four or more sources in a circle that surround the Blumlein Pair, in a proper acoustical environment, have yielded some exceptional recordings, some that yield a sort of  surround sound.

JOSHING WITH JOSH

Shown here are Josh Polak on mandola, Rabbi Shuviel Ma'aravi on guitar, Esther Polak on woodwind and Mike Abramov on hand drums. Shuviel's guitar and Mike's drums were holding down the center of the stereo spectrum with Josh's mandola and Esther's woodwind to the left and right, respectively. I started by positioning Shuviel and Mike at 0 and 180 and got the volume balance by having them move toward or away from the mics. I had to move Josh and Esther around a bit to  balance the loudness of their instruments and get them to the right places on the stereo spectrum, but playing in a circle allows for optimum musician eye contact which normally translates into better performances. To hear a short section of what we recorded, follow this link to my SoundCloud account.

MIKE, DAVE AND VAN
Next in were Mike White, Dave Mattheiss and Van Ertel with a Mike White original. To get the mix right using only the two AT4080 mics, I had to position everyone exactly where I wanted them for level and pan. I'm a one room operation, so I need accurate headphones. My choice are the Audio Technica ATH-M50.  

I started by putting Mike, who was singing and playing, between the front lobes of the two mics, at Zero Degrees. Van's pedal steel amp speaker is on the Black & Decker WorkMate centered between the back lobes at 180 degrees. Dave Mattheiss, with the white hat, we moved around until we got his guitar in the right spot. Because Mike's guitar body is slightly right of center, I put Dave to the left so his guitar would be on the other side of the stereo spectrum. We "mixed" levels by moving toward or away from the mics. We did a Mike White original tune called "Living Will." Here's a link to it. To get a better look, click on the pictures. 

BANG A GONG
Then, Yoga Master Joe Roberson came in with his amazing gongs. Gongs can be very challenging to record without distortion. The sound field generated by a gong is very complex and can easily modulate the diaphragm or ribbon element of a mic, causing unpleasant distortion, even though the record level is well below zero. 

At a foot away, even at low playing volumes, I was getting breakup. I reset and backed off to about 18 inches to 2 feet. Because these AT4080 ribbons are as sensitive as a studio condenser mic, I only needed to use 35 dB of gain from my GML preamps to get a good level into Pro Tools.

Here's a track we recorded. As you listen to the piece Joe performed for me, note the dynamic range; how quiet the space is and how little circuit noise and hiss you hear from the mics at low level and how that sound builds as Joe wakes up one of his gongs. That's a measure of the low self noise of the AT4080. As I mentioned earlier, Audio Technica's effort to increase ribbon length, increase magnet flux and add a small Phantom Powered amp chip to each mic really make a difference. Audio Technica also figured out how to capture more high frequencies with their ribbon.

JAN SEIDEN - FLUTE
It occurred to me that if the gong can be fragile and overpowering, the flute, with its seemingly simple sound is actually rather complex. Internationally known flautist Jan Seiden lives nearby (Baltimore is a very cool town). I asked what she could bring to the AT4080 party and she arrives will all flags flying.

After listening to three of her dozen flutes, I chose a double chamber, double octave, E minor, native american style flute made by Dana Ross at Falcon Flutes. Jan played "Highland Heather" one of her original pieces which you may have heard on her Memory Of Time CD.

As you can see from the picture, this is no ordinary flute. As you will hear by clicking on this link on my SoundCloud site, it can produce multiple tones simultaneously.

Jan stood at the zero degree point, between the two front lobes of the AT4080 mics. After setting a level based on the volume of her playing I hit RECORD and we were done in one take. I did use a bit of EQ and several stereo reverbs for this 24-bit, 44.1 kHz recording, but no limiting or compression.

MADRIGALS
I've worked with madrigal singers Larksong for over 15 years. We have recorded at least three CDs and usually go to St. John's Episcopal Church on the west side of Baltimore where one of the member attends church. Getting six, and now seven singers around these mics while they weave their lines for this sort of recording created some challenges. We had to play around a lot with placement, who stood where, to get them all on the proper place on the stereo spectrum because with Blumlein, there is no postproduction/mixing panning of individual sounds. Attention to the relative volume of each performer is also extremely necessary. If you're too loud, you have to remember to sing softer or step back some.

After a few numbers, it became clear to me that I needed to put the baritone and bass voices at zero and 180 degrees. I also put the only alto there. Then I flanked them with the tenors and finally put the sopranos on the outside edges.

This picture shows the six person arrangement with alto, Martha, to the right at zero degrees and Dave, bass baritone to the left at 180 degrees. Sopranos Andrea and Janet flank Martha and are facing the front lobes of each microphone. Tenors Drew and Jim are facing the rear lobes of each microphone. When I added Kevin, I had him behind Martha at zero degrees.

On this piece, "April Is In My Mistress' Face", you can hear the blending and weaving as each vocal part comes and goes. I've added a little reverb and some minor gain reduction.

Come back again as I add recordings of other instruments to this blog post. Contact me about them at tyford@tyford.com.

SUMMARY

When a company like Audio-Technica spends seven years developing a mic to get it right, you can expect good performance for your investment. While you can use the AT 4080 for VO work after some EQ and in a properly designed booth, it can also be used for a variety of other applications as described above. It’s a worthy addition to your mic locker because it’s sturdier, more sensitive and has more high frequency response than vintage ribbon mics. 

Technique, Inc. © Copyright 2010, 2013 All Rights Reserved

Fast Facts

Application: Studio recording
Key Features: Sturdy, high sensitivity figure of eight ribbon mic
Price: $1245 with suspension mount.

Contact:
Audio-Technica U.S., Inc. 
1221 Commerce Drive 
Stow, Ohio 44224 
Tel: 330-686-2600 

Neumann TLM 107 - Medium Diaphragm, 5 Patterns, Quiet

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Neumann TLM 107
With the U 87 ai, U 89 i and TLM 170 cresting past $3,000 USD, what’s an average guy or gal to do to get a multi-pattern Neumann in the studio? My ears told me that the TLM 107 has more in common with the U 89i than the U 87ai. I've always enjoyed the more natural response of the U 89 i, TLM 170 and TLM 67 mics in Neumann line. Here's my TLM 67 review. 

The TLM 107 is a pin two high, transformerless, five-patterned, medium diameter, studio mic with, high-pass and pad features. The three main patterns are augmented by a wide cardioid and a hypercardioid.

Christopher Currier
According to Sennheiser's Product Specialist, Christopher Currier, "The U-89i and the D-01 are the mics we compare the TLM 107 to in terms of frequency response. It’s not as bright as the U 87 ai or the TLM 103."  

"It’s designed to be a little more linear. The navigation switch is a cool feature. The new grille has been designed to reduce sibilance and popping. The headgrille mesh is new and it's breaking up the high frequencies more than with the TLM 103." 


Jürgen Breitlow
According to Neumann’s Jürgen Breitlow, Director Research and Development, "We have a new generation of engineers in Berlin and with the TLM 107 they are showing how to make a modern microphone which is still very well recognizable as a Neumann in design as well as in sound. My part in this game was the overall concept, I was leading the project and had still a lot of influence on the capsule design.

While many manufacturers have gone Asian, all the parts are from Germany. “The goal in development was an affordable universal and modern microphone.”, says Breitlow, “It should be transparent without being boring.” 

"The diameter of the capsule is equivalent to the K 89 and K 102 used in the U 89 i and TLM 102. Unlike the U 87 ai in which the diaphragms are charged with a centerpoint termination, the TLM 107 back plate is charged and its 6 Micron Mylar diaphragms are conductive from a point on their edge, like the U 89 i."

Design Details
On Neumann’s SD capsules and the LD M7, the diaphragms are glued (not screwed) into place. According to Breitlow, “Pairing front to back is easy because front and back electrode can be measured and selected for distance and tension. Matched pairs still have tighter tolerances than "normal" production. I am not sure if we ever will be able to be so precise that matching is no longer required. That would require tolerances much tighter compared to the diaphragm thickness of 6 microns, which is really tough.” Also, just a small note, the standard mic clip is made out of metal, not plastic. I've run into a number of these plastic clips supplied by other makers and you have to very very careful not to strip them or break them.


TLM 107 Rear Switches
The high-pass filter corner frequencies are 40 Hz and 100 Hz. The pads are -6 dB and -12 dB. There’s a new fangled control button for changing patterns, pads and high-pass filter settings. It works well, but the etched numbers for the pad and filter may be smallish and difficult to read for old eyes. Not that you really have a lot to read. Once you learn the lights' positions, you really don't need to read the numbers. The LEDs go out after 15 seconds or so and if you want to know what the settings are you need to poke the button to turn the lights back on.

TLM 103 (L) - TLM 107 (R)
Sensitivity on the TLM 107 (11 mV/Pa) is noticeably lower than that of the TLM 103 (21 mV/Pa), and the TLM 103 also beats the TLM 107 by having a lower self noise; 10 dB-A for the TLM 107 versus 7 dB-A for the TLM 103. Having said that, the balance between self noise and sensitivity of the TLM 107 is better than my U 89 i, and I’ve been quite happy with it for almost 20 years. When I have needed to record very quiet sources, a double harp comes to mind, the two TLM 103 I have worked just fine.

According to Breitlow, "The noise figures are very often compared to the TLM 103. This is technically not "fair". A dual diaphragm capsule has to use both sides of the capsule. The backside is coupled to the signal as a load, reducing the the sensitivity by 6dB. With this reduced sensitivity the noise goes up by 3dB because the equivalent selfnoise takes the ratio of electrical noise to sensitivity. For this reason, dual diaphragm microphones have very often higher selfnoise figures compared to single diaphragm. The good part in the calculation is that with 6dB lower output you gain 6dB higher max SPL."

"In the U87 ai, the backside of the capsule is taken off electrically for the cardioid pattern to get a better noise figure. But this results in changing sensitivity between the different polar patterns. I preferred to have consistent sensitivity - so that gain can be kept when switching polar pattern."

GETTING TO KNOW YOU
For me, nothing takes the place of sitting in the quiet of my studio with a mic on a stand or in my hands; listening to what it sounds like in different patterns, with different filters and at different angles. After a 17-year career as a radio announcer followed by ongoing freelance voiceover work, I’ve become very intimate with how my voice sounds through a pair of headphones. I’m my own personal “test tone.” The TLM 107 gives my voice a nicely crisp sound with a palpable chest tone. The 10 dB-A selfnoise is more than quiet enough, but may allow you to hear background noises that other mics masked with their higher selfnoise. Ready for some listening?

Here's a link to my first voice test. 
Here's a link to my exploration of the TLM 107 pattern with voice. 

TLM 107 3-Layer Headgrille
The TLM 107 has a three-layer headgrille that Breitlow designed to not only reduce popping but also to control high frequency response. According to Breitlow, “The result is a very good impulse response which shows up in transparency. We got a very nice consistency between on axis to 90 degrees and between the different polar patterns.”

The TLM 107 worked well on my D28S Martin acoustic guitar and J-28LSE baritone Martin. The D28S has a very balanced tone across its six strings. The J-28LSE baritone occupies a space between a six string guitar and a bass. Its strings are thicker and it’s tuned four semitones (four frets) below that of a guitar. The low end can overwhelm mics and require them to be pulled back a bit to keep from being muddy. I spent some time doing a pattern and EQ response study to find which settings worked best for the J-28LSE. Here's a link to that file. It's really interesting to hear the differences each time I make a change. The sound got better and better.

Here's a main link to all of the audio I recorded with the TLM 107. 

BEAT IT
Recording Engineer, Drummer John Wilhelm
With a 141 dB SPL rating, the TLM 107 may also be used on drums. I headed out to Total Recording Studio, in Timonium, MD for an evening of trying the TLM 107 on John Wilhelm's drum kit. We used it as an overhead and also tried it on snare and kick. John's main monitors are four Mackie HR-824 (front and rear), a Genelec 1038a for center and a Genelec 7070A sub-woofer. I suspect as a drummer, John especially likes hearing the kick that way.

John’s drum room is small with a low but treated ceiling. John normally uses a pair of Audio Technica AT4033 for overheads, Sennheiser 421 on his snare, (a Maryland Drum 7" x 15" with Evans G1 coated head) and an EV RE20 on kick, (a 22” Birch Recording Tama, with beater head only.)

TLM 107 (L) and SKRM-100 (R)
He also uses a signature model Yamaha SKRM-100 Subkick Low Frequency Capture Device, which is basically a speaker converted into a mic to capture the very low frequencies from the kick. We were both struck (pun intended) by how much of the whole kit we were hearing with only one TLM 107 in overhead. Everything, even the kick was very clear.
Reach John Wilhelm at totalrecording@verizon.net

At one point I asked John to make sure he didn't have another mic open somewhere around the kit. Nope, no other mics. John was surprised by the TLM 107 and based on his short experience with it, likes it for a mono (or stereo) jazz drum overheads and would like a stereo stereo pair in the room for a live to 2-track recording. They'd also be great for live audience.

TLM 107 on snare

As we moved to the snare and kick, trying cardioid and hypercardioid, we were still hearing a lot of the rest of the kit due to the size of the room. The sound simply had no place to go. Even though John is a forceful drummer, we never even had to engage the pads on the TLM 107. As I left him that night, he said hearing the TLM 107 was a mixed blessing. He enjoyed our tests, but had to put the TLM 107 on his "wish list." Ready for some more listening?


TLM 107 in wide cardioid overhead
TLM 107 cardioid snare
TLM 107 hypercardioid snare
TLM107 cardioid kick

Again, here's a link to the entire playlist of files I recorded with the TLM 107.

Don Armstrong (L) & Kirby Storms (R)
Jazzmatazz
Don Armstrong and Kirby Storms are Jazzmatazz; a jazzy-rock flute and guitar duo who play restaurants, clubs, wineries and corporate events. They had been asking me for suggestions to replace the flute mic which Don also uses for vocals. 

They came into my “green room” and we tried the TLM 107 in a quasi live environment. Both the TLM 107 and my Martin D28s’ K&K Pure Western Mini pickup system were fed to my Fishman 220 SoloAmp which was positioned about five feet away and angled directly at them. A pretty tough test for the directionality of the TLM 107. I sent a mono mixed feed with effects to a Sound Devices 664 to record the track and also fed the 664 output to a Zoom Q4 camcorder set to record at 24-bits, 48 kHz. You can see and hear the results below.

Here are Don and Kirby live from my green room. 


One of Don’s problems had been where to place a mic to balance out the mechanical key sounds, breath sounds and feedback. I found positioning the mic evenly between the keys and breath hole solved those problems. Then it was a case of choosing the right pattern. I started with the wide cardioid to capture more of the key and mouth sounds, but ended up with the hypercardioid because even the cardioid put us right on the edge of feedback given our proximity and angle on the SoloAmp. In a proper venue, I think the cardioid would be usable unless you had the monitors cranked.

The only problem, then, was an inconsistency with Don’s performance across the three registers of the C Flute. His top register notes were 10 dB to 13 dB louder than those in the two lower registers. (He says now that the other problems are dealt with, some practicing may solve that.) Both Don and Kirby liked the smoothness and openness of the sound and the way the two instruments combined.

60's Fender SuperReverb & 1971 Telecaster
Vintage Fender Guitar & Amp
At the very end, just before returning the TLM 107, I had time to pull out my mid-1960s Fender SuperReverb and 1970's modified Fender semi-acoustic (Thinline) Telecaster. I had the original pickups replaced with humbuckers years ago and had a precision bridge installed at the same time. The SuperReverb has four 10" speakers. I had the TLM 107 placed about a foot away from the upper right speaker. 

This time I needed the pad. The SuperReverb only has a gain control for each of its two inputs. There's no way to get crunch without volume. I only had the gain up to about 4, which was plenty loud and found I needed the 12 dB pad to prevent clipping the input of my GML mic preamp. 

I recorded some rhythm chops with the bridge pickup and with both pickups for a meatier sound and then backed off with some finger picking. No sweat. Nothing but net.

Fender SuperReverb and Tele both pickups in Figure of Eight @ 1 foot
Fender SuperReverb and Tele bridge pickup with wide cardioid @ 1 foot

In Conclusion
There's a theory about how much cost and effort it takes to get a project from 50% to 100%. From 50% to 75% takes another 100% increase cost and resources. From 75% to 85% it takes another 100%. From 85% to 90% it takes another 100%. From 90% to 95%, another 100% You get the idea. The TLM 107 may not cost as much as the U 89 i or U 87 ai, but having spent this time with it, I think it fits well within the Neumann bloodline at a price that, while not insignificant, is a lot more manageable and that fits into the scale of cost versus benefit. Nicely done!


Technique, Inc. © 2014 All Rights Reserved
Reach out to Ty Ford at www.tyford.com

Zoom Q4: 1080p with LCD viewer and 24/96 kHz audio

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For $299, I am amazed by this little "toy." I have locked my wallet and credit card in the trunk of my car for the last year to keep the "impulse buy" demons at bay from this sort of technology. The GoPros are also very hard to resist. When I had the chance to get one here to play with, without nicking my credit card, I did. After looking at both GoPro and Zoom, I chose the Zoom for a test ride because of the audio and because of the detachable LCD screen.

This review will be an editorial departure for me in that I'm letting the video do most of the talking. This first "show and tell" video is a tour of the Zoom Q4 features. Blogger doesn't seem to allow text beside videos, so there's more white space than usual.



AUDIO CHECK
I shot the video below here in my studio. In it you can hear the difference between a Schoeps cmc641 and a Sound Devices MixPre-D fed to the Zoom Q4 and the Zoom Q4 mini stereo mics.


INSIDE-OUTSIDE
Back to the video side. Below is a short section I shot leaving the house and headed for the great outdoors to demonstrate the auto exposure features of the Zoom Q4. Notice the blow out as I exit the house and how well the auto circuitry corrects after I get outside. Also notice the critical focusing distance, about four inches, as I push in to a plant on the porch and the brick wall.


CHECK YOUR TEMPERATURE?
I noticed some white balance differences and realized that the Zoom Q4 seems to balanced only for daylight. Below is a tour of my studio with four different light sources. Notice how that compares to the daylight shot samples.


MUSIC VIDEO
Below is a simple shot of two of my good and very talented friends Don Armstrong and Kirby Storms (Jazzmatazz) playing flute and guitar in my green room. The mic Don's playing flute into is the new Neumann TLM 107. Kirby is playing my D28S Martin. The Martin has a K&K Mini passive, piezo pickup system.  Both the Neumann and the Martin were fed to my Fishman SA220 SoloAmp. I took a mono feed from the back of the SA220 with reverb and fed it to a Sound Devices 664 mixer/recorder so I could control the level going into the Zoom Q4 properly. The audio was recorded at 24-bit 48 kHz, but has been squished down a bit by YouTube. Still, it sounds very good. What you're hearing is the Zoom Q4 audio.


Notice the lighting. I have large opaque, white, accordion pull down/pull up shades in this room. They do a nice job of softening the daylight and, again, the color balance is very good.

OUTSIDE
On a warm, humid rainy Labor Day. I took the Zoom Q4 with me to document a tremendously fun backyard music festival held every year for the past 30+ years here in Baltimore by Kurt and Cathy Milam. Next year may be the last year of this Labor Of Love. Again, daylight, so no color temperature issues. The only mics I used were the stereo pair on the Zoom Q4 with the "dead squirrel" wind protector. This is Amy Hopkins and Dave Dilworth doing a Jeffrey Foucault tune. I got their permission to publish this.


IN CONCLUSION
Apart from the color temperature white balance issue, I'm still pretty amazed by what the Zoom Q4 delivers. Nice Job!

Here's a thought. Got a DSLR with bad audio capture? Bring the Zoom Q4 along and use it for stereo audio.  Oh, and yeah, maybe another angle of video.

PS: This is the most video intensive post I've made so far. I thought it was the best way to tell the story. What are your thoughts?

Technique, Inc. © Copyright 2014 All rights reserved.



Audio-Technica System 10 - Now With Battery-powered Receiver & More

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System 10 with AC-powered Receiver
I was very interested in Audio-Technica’s System 10 roll out a year ago. The first System 10 only offered an AC-powered receiver. That kept it from camera or bag use for location audio. It appeared that Audio-Technica was looking first at the Church, PA and Musician markets and coming in with some nice pricing. $300 for a handheld mic and receiver is head turning. The audio is compressed, using an APTX codec.


Now there's a new battery-powered receiver designed to fit on a camera and a stomp box for guitar players. That certainly changes the game, but let's backtrack.

System 10 is more than just a new, remarkably affordable wireless system, it's a technology shift. Not VHF. Not UHF, but up in 2.4 GHz with Wi-Fi. I need to make a point here; System 10 does not require Wi-Fi to operate. In fact, a thick Wi-Fi environment may make it operate closer to Audio-Technica published specs. In addition to passing 24-bit, 48 kHz, 3.8 mSec low latency audio, the System 10 transmitter and receiver are in constant communication with each other and they shift automatically as needed and without dropout or frequency coordination. Read on…!

Maximum Performance?
OK, the manual for this receiver makes me a little concerned. "Place the receiver at least three feet above the ground and three feet from any wall or metal surface. In bold letters; Keep System 10 receiver 30 feet from any wireless access points. In multi-channel systems, keep receivers at least three feet apart." Well that means you can't mount two of them on top of a camera or in a location audio bag. But what about the AC-powered receivers? Those you can stack eight high? That doesn't make sense.

Audio-Technica's
Gary Boss
called Gary Boss at Audio-Technica US for some background. "We've been working with digital wireless for more than a dozen years, if you count when we began research. Our current upscale product, SpectraPulse Ultra Wide Band, has been out for seven years. When we submitted it to the FCC for approval, we had to send one of our engineers because the FCC couldn't find the signal. The nature of the system has the signal buried in the noise floor."


Boss also explained that the specs and system use limitations of System 10 are typical of Audio-Technica's conservative approach to specs. "We are super cautious, to a fault. We don't quote the best case scenario, we quote the worst. So, for example, the specs on the product sheet are for not just one unit in operation, but eight units in simultaneous operation. I've seen System 10 in operation with a stomp box at a trade show just ten feet under a remote multi-anntenna Wi-Fi array with no problem. And we have gotten more than eight System 10 units running, but under laboratory conditions." 

The range quote on the stomp box is only a 60 foot radius. (which, when you stop and think about it is pretty remarkable.) The other System 10 data sheet talks about expecting a range of 100 feet for the body mics, but I and others have already exceeded Audio-Tecnica's range estimates, getting out to 125-150 feet or more. 

And what about the 2.4 GHz operating frequency. As transmitter/receiver wavelengths get shorter, it’s presumed that reception can become more fragile due to increased multipath. I was out last week in a busy and very full parking garage. Lots of reflective metal and rebar, both moving and stationary. Metal, as you may know bounces the signal around and can play havoc on wireless reception. We got 125 feet and more with one System 10 transmitter and one of the new battery-powered camera mountable receivers and I had the receiver in a shirt pocket. Very impressive. 

But back to diversity. Historically, VHF and UHF wireless systems have done better when the receiver has complete dual diversity. That means dual antennae and dual tuners in one receiver. Below that, the next level of diversity uses one tuner and switching antennae to prevent drop out due to multi-path reception. That's only Space Diversity, because the two antennae are in different spaces.

System 10 operates with Frequency, Time and Space diversity all at the same time, using 2 Mhz of bandwidth. So the playing field is very different. With System 10, Frequency Diversity sends the signal on two dynamically allocated frequencies simultaneously. Time Diversity sends the same audio bit twice within the same frame, so there's redundancy. Finally, per the above, Space Diversity uses two antennas on each transmitter and receiver to increase signal integrity. This protocol would seem to offer a much more robust operation than pretty much anything that has come before. How come they didn't win a TEC Award for this technology? I mean REALLY. How Come!

Simple Operation
All of this Diversity is happening without human intervention. The systems comes ready to go right out of the box with no setup required. The numbers 1-8 are not dedicated channels, just system IDs for identification purposes. The reality is you will never know what frequency you're operating on. However, if you want to assign different ID numbers, turn on receiver, select unique ID, pair with corresponding transmitter, done. There is no frequency selection by the operator. The system figures it out. Turn on a receiver, let it find a channel. Power up a transmitter, hit the pair button on the receiver and wait a few seconds. Done! Up to eight wireless systems can be operated simultaneously, for now. The digital audio sounds great and I think we can say goodbye to analog companding and noise reduction for wireless. 

Steve Oakley
This does beg at least one question, forwarded to me by video professional Steve Oakley. What about high density situations like the Super Bowl, in which, historically, a person called a frequency coordinator charts everyone's wireless systems and lays out a frequency plan so that no one steps on someone else; either by direct frequency conflict or by intermodulation. And near a big Wi-Fi network with hundreds if not thousands of smart phones in a stadium that size, what happens? 

According to Steve, "I talked with the officials and they thought that using wireless in the Wi-Fi band probably wouldn’t be safe or reliable, especially if anyone was making a Wi-Fi hot spot off their phone or laptop….which while forbidden to be done by the pro’s, may be done by someone in the audience because they don’t know or don’t care, and good luck finding 1 guy in 50,000!"

With so much wireless at major events, finding good frequencies can be difficult. Steve was getting ready for work on a Packers/Detroit game and just received his allocated frequencies. He offers this "from the trenches" comment, "Believe it or not the Sennheiser G3’s I’m running have, so far, been much more reliable than digital Lectro’s. Go figure! In fact the wireless mics being used at the NFL games on the parabolic mics are also Sennheiser, but the higher up models with presets and the option for higher output RF. So this choice of gear probably makes the frequency coordinators job a little easier since we are all using the same systems. In fact you’ll find most of the other ENG crews at the games are also using Sennheisers."

It may take some time before "the establishment" accepts digital audio in Wi-Fi space, but continued testing will soon provide workable answers. How hard can you push this breakthrough concept before the envelope breaks? According to Boss, "In one school setting A-T had a total of 12 channels spread out over 12 different classrooms. (one system per room)  Although the system is only speced to operate a maximum of 8 simultaneous systems. The 2.4GHz band and multiple levels of diversity allowed us to take advantage of the natural range restricting cinder block construction to achieve 12 simultaneous systems in one building. 

System 10 With Stompbox Receiver
System 10 Models
As noted above, Audio-Technica offers System 10 with body packs, lavs and headworn mics with prices varying depending on the specific configuration, and a guitar stomp box version System 10 with the receiver in the metal-cased pedal (for $349.) The pedalboard-mountable receiver is designed to stand up to the abuse of a typical pedalboard, and has dual outputs that allow you to route your signal to multiple amps, or you can set one output to mute while routing the other unmuted signal to a tuner for quiet on-stage tuning. It gets power from a universal 9-12 V DC power supply.


System 10 Transmitter, AT-MT830c Lav 
and Camera-mounted Receiver $449
DSLR Friendly
The camera-mount version offers a battery powered transmitter including AT MT830c lav, receiver with a slip-in case that mounts the receiver on a camera shoe and is powered by an internal, rechargeable battery for $449. The AT MT830c is a very nice side-address lav that's been in Audio-Technica's line for years. It's not the smallest lav on the planet, but the larger diaphragm results in lower noise

The ATW-R1700 Digital Receiver is a tidy package, weighing about four ounces with battery. On the back, are two brass screw-on fittings onto which you screw the antennae. The manual says they are most effective when positioned at the angle shown below. In between the antannae is the power switch. 
System 10 ATW-R1700 Digital Receiver

Along one side are two switches and a 1/8" jack. These are the Dual Mono/Balanced switch, the three position 0 db, -10 dB and -20 dB switch and the 1/8" output jack from which you feed the camera. The Dual Mono/Balanced switch allows you to output balanced mono or unbalanced stereo from the 1/8" output jack. On the other side of the unit are a USB port for powering and recharging, a 1/8" TRS jack for headphones and a small headphone volume control. It's really nice to have the headphone option, because many DSLRs don't have a headphone output. On the front are a System ID display and choice button, a Pair button, Pair light an audio peak indicator. A 1/8" TRS to TRS cable is supplied to connect the receiver to the DSLR

The rechargeable battery is not removable, You charge it via a USB power supply/charger. That makes a lot of location audio people a little nervous. What if you forget to charge it the night before? You can’t just swap out the battery. The receiver battery takes about 4.5 hours to fully recharge. A new or freshly charged battery lasts for 12 hours, but what happens as the battery ages and holds less charge? Audio-Technica says the battery is factory replaceable.
                                                     
The Ginsberg Report
Fred Ginsberg of FilmTVSound, who's had good practical experience on the System 10, had a solution for the concerns about the non-removable 12-hour battery. The ATW-R1700 receiver uses a USB port to charge the battery. You can use the very same port to externally power the receiver. Fred found a "lipstick" power supply rated at about 3000mAh for $19. The mAh rating of the battery in the receiver isn't listed in the specs, but larger batteries are easily available. You can also run System 10 receivers off of your laptop USB port in an emergency.

Fred Ginsburg CAS PhD MBKS
Here is the link to the YouTube video that Fred posted with 3 of the walk tests that he did with the System 10 camera-mount wireless. According to Fred, "The unit achieves better than expected range; easily delivering the advertised 100 feet. On multiple occasions, I was getting 150 to over 200 feet."

Fred reminds us that these are low cost systems and not intended to compete against the (high end, and very expensive) wireless used on major shoots. I agree, but for an inexpensive unit with low latency, digital audio, I think they perform surprisingly well.

Fred brought the system with him to the Cinema TV Arts Dept at California State University Northridge for more tests. He describes the situation, "CSUN has a soundstage for the film program that is approx 50 feet by 75 feet. Camera and the System 10 stayed in the rear corner, while I moved around the entire space. Even while hovering near a laptop and the lighting dimmer board -- the System 10 performed flawlessly. And, yes, there is Wi-Fi in the stage. We went into the main corridor of the film/TV building (Manzanita Hall), which is dotted with electronic RFI emitters and WiFi routers (according to the department's Chief Video Engineer). The System 10 transmitted perfectly the entire length (approx 150 feet)."
"Heading outside onto campus, on multiple tests they achieved 150 to 250 feet. Although they did encounter the occasional minor dropout on one or two of our footpaths, the unit was always reliable up to 125 feet; and depending on the pathway, sometimes good past 200 feet. I included one of those walk tests, where they made it 220 feet before the signal began to cut out."

Fred said the aforementioned Video Engineer was curious about transmitting through walls, so Fred stayed inside the 'Corridor from Hell' while the Engineer went outside and roamed around the exterior of the building. "Signal held up nicely passing through just one exterior wall, but did drop out when he turned (still outside the building) 90 degrees from the corridor and thus placed multiple interior rooms in their signal path."

I'd like to see a full-on trial with, say, multiple, fully deployed eight channel systems in a dense Wi-Fi environment, to see if the System 10 could crack the Broadway Theater market. I don't expect Audio-Technica's System 10 to have an easy ride into the high-end professional marketplace because the established wireless user base is too well invested in Lectrosonic, Sony, Sennheiser, Zaxcom, Audio Ltd., Q5X and other more expensive gear. What I do see happening is their adoption by schools and other corporate entities first. I also see sound departments buying several sets to experiment with and using them as needed. 
System 10 needs to earn its stripes. Apple's FCP X caused much ranting and raving by 'the establishment' when it came out. "It's a toy! iMovie on steroids! Bah!" Apple listened and began adding features based on who screamed the loudest (and most intelligently.) FCP X has steadily increased in popularity ever since. I don't know why they didn't get a TEC Award for System 10. They certainly deserve it. 
Wish List For System 10 (or any location gear)
Smaller transmitters that are easier to hide (for everyone)
Plug-on Transmitter with Phantom Power (contributed by Jay Massengill, a producer in Burlington N. C.)
A two channel receiver with headphone jack for camera mounting (for DSLR users)
Transmitters that also record audio to a removable, internal card (for film production)

SpectraPulse Addendum
Audio-Technica's SpectraPulse Wide Band runs at 6.3 GHz. It's a 16-bit system with a 24kHz sample rate. The UltraWide Band data is transmitted in extremely short-duration pulses sent in a timed sequence over a wide frequency spectrum. To decode the pulses, the digital receiver module must know exactly when, where and how to listen. This makes SpectraPulse inherently secure. In addition, there is optional encryption that meets NIST-approved AES 128-bit standards. It runs 14 channels without the need for human frequency coordination.

Technique, Inc. © Copyright 2014 All Rights Reserved
Contact Ty Ford at www.tyford.com





RME ADI-8 DS Mk iii A/D D/A Conversion - Time marches on, and gets more accurate.

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Ty Ford
I’m writing this article in February 2015. If you’re still arguing that analog is better than digital, I can’t help you. I’m OK with the idea that drums and other sounds with high transients may be recorded on properly aligned and biased analog tape recorders and that the non-linearities of that system may result in a sound you like better. The rounding off by tape compression is an acknowledged, if not expensive, effect in music production today. I get it.

File that on the shelf next to “Ford’s Theorum” (something I’ve said for years) which states that, “Good analog is better than bad digital and good digital is better than bad analog.” Further, that, “Good solid state is better than bad tubes and good tubes are better than bad solid state.” You can argue the boundaries of those intersections, but you can’t argue that they don’t exist.

The transformer - no transformer debate is sort of a corollary of the above. Yes, iron changes the sound and in some cases can be said to make things sound more pleasing. Yes, the right iron and the right number of windings can help low sensitivity mics do better by providing some “free” gain. 

Once you get tired of arguing any of the above, there’s still A/D conversion to hammer out. Fortunately, we are years and many revisions past the early and ugly attempts at 16-bit, 44.1 kHz. Yes, there was a quantum leap when good 24-bit was achieved. Yes, we can and have taken it farther, but In a world where the largest market segment is happy listening to mp3 files on cell phone earbuds, how far do we really need to go?

Audio is seldom the prime mover in the Technology Circus. More typically, audio is the happy recipient of advances in the video, medical technology and scientific measurement industries. Below those impressive moments is...wait for it...Audio. 

George Massenburg GML 8204
In my own pursuit of “financially limited perfection”, I realized that there are others like myself who can hear things that others can’t. 

Likewise, there are things I’m relatively deaf to that others can hear. We are all in pursuit of satisfaction; our own individual happy place that makes us smile when we hear “the right stuff.” There are many paths.

I improved the audio on my first Pro Tools rig almost 15 years ago by carefully selecting some good preamps - GML 8204 and Millennia Media STT-1. After settling in with them, I took aim at the A/D conversion. The hardware I was using had an ADAT optical port, and because of my budget, I tried the RME ADI-8 DS shortly after it came out in 2000. This was back before AES standards had evolved and Tascam’s TASCAM Digital Interface (TDIF) and Alesis’ ADAT ruled the swamp.

Millennia Media STT-1
Almost 15 years later, my ADI-8 DS is still in the rack and apart from requiring a recent power supply replacement, has been powered up and quietly doing its job all along. I did the power supply swap out with a little help from RME US rep Jeff Petersen at the Synthax in Ft. Lauderdale office, distributors for RME/Alva Cableware and Ferrofish in the USA.

Synthax's Jeff Petersen
The thing is, A/D converters often don’t get the respect they deserve. Petersen puts it nicely, “Converters aren’t sexy, they’re just clean and effective.” He’s right. 

The RME ADI-8 DS Mk iii that's the object of this review, and an upgrade from the piece I've ben using is just a one rack-space box with a few lights and switches on the front and holes in the back. 

It’s supposed to be inaudible, invisible. If you’re an electric guitar player with a pedal board, you want each pedal to change the sound. Fine, but the job here is to allow as much original detail though as possible.

Back then, combined with the better preamps, the first RME ADI-8 DS made a noticeable improvement in my audio at 44.1kHz and 48 kHz. The Lavry and DB Technologies I tried had more detail, but were too expensive and I was satisfied by the improvement brought by the ADI-8 DS. I could go ADAT light pipe right in to the Digidesign hardware. That was a big plus.

By replacing Digi preamps and A/D conversion, I had bypassed the entire front end of the Digidesign hardware. I don’t use the D/A conversion, just the A/D. No disrespect to Digidesign. They had a piece at a price. I just built out from that with gear that I could hear made a difference and that (Including five preamps and eight channels of A/D conversion) cost nine times my original investment in Digidesign hardware and software. I expected that would pay off for my host-based system as computers got faster and storage got cheaper. It did.

I think I started with Pro Tools V 5 and am now at V 10. I use it every day and make a living with it. Pro Tools 11 is current and, for the first time, offers faster than real time Bounce To Disk, but I’ll need to upgrade my entire Mac platform and all the good and bad can be expected to go along with that. I’m just not ready to do that at the moment.

RME ADI-8 DS Mk iii
RME ADI-8 DS Mk iii
The most notable operational difference between my ADI-8 DS and the newer ADI-8 DS Mk iii is that the D-subs connections now go to AES/EBU I/Os instead of TDIF. As the AES/EBU spec has evolved, more AES enabled gear has been created; including this upgrade for the ADI-8 DS Mk iii. About how the line of RME converters plays out, Petersen says, “The ADI-8 Pro (48k max) was the original, the ADI-8 DS (96k max) would be the Mk ii essentially, and the ADI-8 DS Mk iii (192k max) is the 3rd generation of this venerable 8 channel converter. The QS is a different animal.” I took the opportunity to give it a test run during the time my “original recipe” unit was out of the rack for its power supply exchange.

The Mk iii benefits from the 14 years of advancing technology since the first ADI-8 DS was released. The most noticeable spec difference is the lower latency of the Mk iii. It’s at 12 samples, the old one was at 40 samples. I am not bothered by the old unit’s 40 samples, but if that was a deal killer for you, see how you like 12. 

Like the earlier model, The ADI-8 DS Mk iii is an 8-channel AD and DA converter in a standard 19", 1U box. This includes ICC Intelligent Clock Control (ICC), SyncCheck™, four hardware reference levels up to +24 dBu (instead of just three), AES/EBU and ADAT I/O, with up to a 192 kHz sample rate. 

According to RME, “Our unique SyncCheck and AutoSync technology has evolved into the new Intelligent Clock Control of the Hammerfall DSP system. HDSP is the only digital I/O-system worldwide capable of measuring and displaying the frequency of all clock sources. Even word clock! Based on validity and current sample rate the system then decides which clock source should be used - fully automated and performed in hardware. With this the HDSP system offers the easiest handling of the present clocks.”

Word Clock Implementation
The signal present at the Mk iii BNC Word Clock input can be single, double or quad speed and when lock is achieved the WCK LED stops blinking and glows steadily. RME uses what they call a “Signal Adaptation Circuit” and “Automatic Signal Centering” to reshape the Word Clock signal. This reshaping addresses several possible fault problems, including poorly shaped, low or high signal at the input and sends the new signal to the Mk iii WCK BNC output. The WCK BNC output also has built-in 75 Ohm termination, should you need to slave the Mk iii at the end of a chain of digital devices.  

SteadyClock is RME’s answer to the problematic Superclock and has been added since the earlier ADI-8 DS. It uses a 22 MHz clock signal that RME believe better answers the stability problems raised by Superclock. The Dither feature of the earlier model is also absent from the Mk iii.

Analog-Land
Analog inputs are via eight 1/4” TRS jacks or one 24-pin D-sub connector. The TRS input jacks are servo-balanced to accept either balanced TRS audio or TS unbalanced audio. You may see the term “double balanced” in the marketing information. I had no idea what that term meant, so I inquired. Turns out it means that both the TRS and D-Sub connections are each balanced, hence “double balanced.” I would hope so, but you never know these days. Another term, Full Symmetrical, caught my eye and I was told that meant that all stages are fully balanced instead of single ended. 

While you may use only one of the inputs, 1/4” or D-sub, the Mk iii outputs have separate drivers enabling both 1/4” (TS or TRS) and D-Sub outputs to be used simultaneously. The TRS/TS output is servo-balanced, but the D-sub is not. When connecting with unbalanced audio gear, pin 3 must be lifted.

As with the analog input, the analog output also has four selections for level; +24 dBu, +19 dBu, +13 dBu and +4 dBu (-10 dBv).

A/D Input
If you have analog gear with different output levels, the DS Mk iii is even more obliging than the original DS. There’s a fourth analog sensitivity setting from which to choose. So now you have +4.2 dB (-10dBv, +13 dBu, +19 dBu and +24 dBu. These changes are all made before the A/D converter. 

The analog LED input level lights on the original DS flashed at 2 dB below FS, which I initially found disturbing, but learned to ignore. The Mk iii partially lights at -2dB FS and goes to full brightness at 0 dBFS. In the heat of the moment I don’t know if I’ll notice the difference. I’m still in ignore/denial mode. I’m not always looking at the rack, so LED hold lights may be helpful if you’re the type that likes pushing input levels as high as you can get them.

Digital Patch Bay
After A/D conversion, the audio is available at both AES and ADAT outputs simultaneously, except when in Patch Mode. The Patch Mode lights indicate the choice you’ve made and which output is active. 

Maybe the key differentiation feature for is the Digital Patch Bay. If you have a variety of digital audio boxes in your studio and they need to talk to each other, you may have had to stop, get behind your racks and swap cables and/or connections to do certain things from time to time. 

The RME web site explains this simple, push-button feature quite well, “A digital patchbay with free choice of source and destination setup can be used to convert ADAT to AES, AES to ADAT, cross-convert them at the same time, pass ADAT on to ADAT while monitored analog, and many more. The ADAT outputs also feature copy mode for connection of two different ADAT devices. These powerful and easy to use modes add significant value to the already outstanding conversion quality.

During single speed A/D conversion (up to 48 kHz), both ADAT and ADAT AUX outputs carry the same audio. A/D conversion is not available during AES input to ADAT and AES output
ADAT input to ADAT and AES output
AES input to ADAT output and ADAT input to AES output
AES input to AES output and AES input to AES output. 

In normal studio operations, the DS Mk iii will be the master and any digital devices connected to it will be slaves. That’s how it is with my setup. The Digidesign 003R is the slave and that is a choice in the Pro Tools software. Within Pro Tools, the ADAT protocol and optical source (because ADAT is optical) are the way to go. the When devices are locked correctly, the Mk ii Sync Light is on and steady. A blinking light means there’s a problem.

In Conclusion
As mentioned before, decidedly unsexy, but very important to the integrity of the signal chain. I recorded voice, acoustic guitar and drums separately on both my original ADI-8 DS and the Mk iii at 24-bit, 44.12 kHz. Try as I could, I could not detect any real difference in the sound. 

So it’s primarily about AES/EBU implementation, the lower latency, the added flexibility of the patch bay, SteadyCLock and an added input sensitivity and output level increment. My ADI-8 DS has been in the rack and powered up 24/7 for about 15 years before the power supply checked out. That’s got to count for something, too.

Technique Inc. © 2015 All Rights Reserved 
www.tyford.com

Here’s the bullet list from the RME web site.

8-channel AD converter, full symmetrical design, 119 dBA
8-channel DA converter, double balanced output, 120 dBA
Low latency conversion: less than 12 samples delay
4 x AES/EBU I/O per D-sub, 8 channels @ 192 kHz
2 x ADAT I/O, 8 channels @ 96 kHz, 4 channels at 192 kHz
Word clock input and output
2 x 8-channel level metering
Comprehensive digital patch mode for full interconnectivity


RME ADI-8DS SPECS

Input AD: 1/4” TRS jack and 25 pin D-sub, servo balanced, all symmetrical audio path
Output AD: 4 x AES/EBU, 2 x ADAT optical
Input DA: 4 x AES/EBU, 2 x ADAT optical
Output DA: 1/4” TRS jack servo balanced, to +21 dBu. 25 pin D-sub, balanced, to +24 dBu.
Dynamic Range AD: 119 dBA
THD AD:< -110 dB (< 0.00032 %)
THD+N AD:< -104 dB (< 0.00063 %)
Crosstalk AD:> 110 dB
Dynamic Range DA: 120 dBA unmuted
THD DA:< -104 dB (< 0.00063 %)
THD+N DA: < -102 dB (< 0.0008 %)
Crosstalk DA:> 110 dB
Input level for 0 dBFS: +24 dBu, +19 dBu, +13 dBu, +4.2 dBu
Output level for 0 dBFS: +24 dBu, +19 dBu, +13 dBu, +4.2 dBu
Input/Output level at 0 dBFS @ -10 dBV: +2 dBV
Sample rates: 44.1, 48, 88.2 , 96, 176.4, 192 kHz, variable (sync/word)
Frequency response AD/DA -0.1 dB: 10 Hz - 23.2 kHz (sf 48 kHz)
Frequency response AD/DA -0.5 dB:< 5 Hz - 43.0 kHz (sf 96 kHz)
Frequency response AD/DA -1 dB:< 5 Hz - 85.0 kHz (sf 192 kHz)
Power supply: Internal switching mode ps, 100 V-240 V AC

Rode NTG4 and NTG4+ Shotgun Microphones - New Tools

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Rode NTG4 & NTG4+
Rode continues it’s pursuit of the audio for video market with a new pair of shotgun mics; the NTG4 ($369) and NTG4+ ($399). They are basically identical except that the NTG4+ has a USB rechargeable Lithium Ion battery. 

The NTG4 runs off of 12 V, 24 V or 48 V Phantom Power, but the NTG4+ only runs off 48 V Phantom Power or its battery. That battery, when new, takes two hours to charge and runs for about 150 hours. You can charge it with any USB charger or a USB port on a computer.

This means you can use the NTG4+ with standard professional audio gear that supplies Phantom Power, or with cameras and devices that may not have Phantom Power. With 150 hours run time, you could shoot 24/7 for six days without recharging.

Note that not all Phantom Power supplies are the same. If your Phantom Power supply is only 12 V or 24 V and you try to use a mic that needs 48 V, it may work, but there will be some degree of distortion because you’re starving it. 

Rode tweeked the capsule or capsule response in these two new mics, rolling off a bit of the low end that their NTG1 and NTG2 capture. I don’t find it to be a problem. There are shotgun mics like the Shure VP 89 that are obviously designed for run and gun ENG work in very noisy environments. On that series, the low end is markedly rolled off. Not so much on the NTG4 and NTG4+

If you can't read the specs to the right, here's a link to the data sheet. 


Triple Treat

There are three buttons on the NTG4 and NTG4+; High Frequency Boost, Low Frequency Cut and -10dB Pad. The HF Boost feature is usually used when the mic is mounted in a blimp/zeppelin wind stopper to offset the high frequency loss caused by the fur and material. You could also operate with it on if you just liked the sound of the peaky response or if the person you’re micing has one of those really soft, dark voices that needs a little help.




And here's my hands-on demo so you can see and hear what the NTG4+ sounds like. Yes they have revoiced the capsule, relative to the NTG1 and NTG2. I like the change; a little tighter low end.





Both the NTG4 and NTG4+ come with the RM5 non-suspension mic clip, ZP2 padded storage pouch and foam windshield. The NTG4+ also includes the micro USB charging cable. 

Rode is big on support. Just register the NTG4 or NTG4+ at their web site and your new mic will be under warranty for ten years. 

Technique, Inc. © 2105 All Rights Reserved

Reach Ty Ford at www.tyford.com



Zoom Q8 - Like A Q4, But More

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 On the face of it, if you thought the Q4 was a good deal at $299, but lacked some features, maybe the Zoom Q8, at $399 is the answer. I like the Q4 for the price and found it easy to use. The picture was good for this kind of small camera. If you missed my review of the Zoom Q4, you can find it here. 

The Q8 came with foam pop filter for the X/Y mics. It's OK for indoors, but you really need a dead squirrel for outside work. The kit also includes a USB charging and file transfer cable, lens hood and lens cap.

The Q4 and Q8 are color balanced only for daylight. The lack of adjustable white balance for both cameras means, if you shoot under incandescent lighting, everything looks too red. If you shoot under fluorescent things look green and yellow and you can not correct that in the camera. You can, of course, in post, if you know how.

The Q8 doesn’t seem to make automatic exposure adjustments as well as the Q4, but perhaps thats because the LCD viewer panel makes most shots look blown out even when they aren’t. 

I don’t recall seeing the 24 Mbps 2304 x 1296, 30 fps video resolution on the 
Q4, but the Q8 offers it. Quicktime will play it, but when I imported some of these files into FCP X, I got this text box: The Video Properties of this clip are not recognized. Click OK to use the settings below or click Cancel to choose another clip to automatically set your properties from. Format 2K, 2048x1152 29.97p. Final Cut X had to re-render them before they played on the timeline.

In addition, while the clips played back in the camera, they dropped frames when played back from a USB card reader and also when played back from my computer's internal HD using Quicktime 10.1 player. Curiously, Quicktime 7 was able to play without dropped frames. I was using a SanDisk Extreme 80 MB/s 10 SD HC I SD card. 

The smaller, 24 MBPS 1920 x 1080 30P files opened right up with no problem. While most of the Q8 video settings are 30 fps, it does have 1280 x 720 and 800 x 480 modes at 60fps for shooting higher action subjects. 




SD CARDS
It should come as no surprise to anyone who’s dealt with SD cards that each device has its favorites. Blindly showing just any SD card into the slot is asking for pain and disappointment. The Q8 records directly to SDHC, and SDXC cards, up to 128GB. There’s a document on the web site that shows which cards to get. It didn’t show any 128 GB cards but that chart was put up in December 2014. Here’s the link.  

http://zoom.co.jp/download/Q8_recommended_cards_E.pdf

The Q4 only has two settings for video, standard and wide angle. The wide angle, of course had fish-eye curvature. The Q8 has a five step digital zoom, accessed by touching the LCD touch screen and then the + or - markers on the left side of the screen. If you don’t touch them fast enough, you have to touch once to activate the switch and then again to zoom in or out. The double touch is a little annoying. Five steps is probably more than you'll use, but they are nice to have for critical framing.



VIDEO
The Q8 uses a 1/3”, 3 megapixel CMOS sensor and records to MPEG-4 AVC/H.264 MOV. It uses the Ambarella AVC Encoder. Like the Q4, the lens is fixed-focus (36cm to infinity) F 2.0 with a 16.6 mm focal distance. There are HD settings, or you can shoot in standard definition (SVGA) when space or streaming bandwidth is at a premium.
The flip-out, full color touch screen LCD display is not removable as it was in the Q4. It does rotate, allowing selfies, but is not as true as the Q4 LCD. Images appear over lit or blown out on the LCD panel that aren’t when viewed on a computer monitor. 

If you can twist to exactly the right angle from the monitor, you may be rewarded by a more accurate view, but maintaining that angle is very difficult. Although you can playback clips in the camera, the LCD problems make that a questionable practice because you can’t really be sure of what you shot unless you use another monitor. I suspect that over time you just get used to it. 

Also, the surface of the LCD screen is very reflective and that makes shooting outside in daylight a worrisome process. You just can’t be sure what your getting. The micro-HDMI Type D port comes to the rescue if you have another monitor that takes HDMI.

Connecting the micro HDMI output from the Q8 to the HDMI input on my flat panel TV resulted in a much better picture. I also tried going from Q8 to a Black Magic UltraStudio Mini Recorder box, and into my MacBook Pro via Thunderbolt running Black Magic Media Express software. That was a disappointment. 

AUTO, CONCERT, NIGHT
There are three exposure settings, concert, night time and auto. While there was some difference among them, I was surprised at how similar the exposure was during playback on the SD card on my computer. I think the auto exposure circuitry works very hard to keep the picture as well exposed as it can regardless of which exposure setting you’re using. Below is some test footage. I took the Q8 outside and shot with the three exposure levels; Auto, Concert Lighting and Night. 


 Auto


 Concert


 Night Vision


Then, in an effort to get a handle on the low light capabilities. I did something similar down in the studio, in the near dark. Again, here's Auto, Concert and Night.


 Auto


 Concert

 Night Vision



FOUR RECORDING MODES
There are four different Recording Modes; MOV, MOV + WAV, STEREO AUDIO and MULTI Audio. Each one handles audio differently. Get lost here and you’re, well, Doomed! (My way of saying, “Heads Up!”)

MOV Mode
In this mode audio and video are recorded and you can use all four audio inputs. The audio is routed through the mixer and is saved as two track mono or simple stereo and is attached to the video file. In this mode you can choose to record audio as high as 24-bit, 96 kHz. 

MOV + WAV Mode
In this mode audio and video are recorded. The audio tracks are double routed. That is, one route brings all four inputs through the mixer to a two channel mix attached to the video file. In addition each track is saved as an individual file before it goes through the mixer. The on-board LR, XY stereo mics will be saved as a single stereo LR WAV file. Inputs 1 & 2 can be linked to create one simple stereo file or unlinked to create two separate mono files. In this mode, the highest audio rate is 24-bit, 48 kHz. 

STEREO AUDIO Mode
In this mode, no video is recorded. All four inputs are recorded through the mixer as one mixed stereo file.

MULTI AUDIO Mode
In this mode, no video is recorded. The audio does not pass through the mixer. You get one stereo file from the on-board LR mics. In addition, you get either one linked stereo file from inputs 1 &2, or unlinked, two separate mono files. Again, in this mode, audio may be saved only as 44.1 kHz or 48 kHz and 16- or 24-bit.

To help you from making mistakes, when you’re in a recording mode that does not record video, the LCD panel shows a graphic of a microphone instead of showing you what the camera is seeing. Once you decide which mode you want, you simply push the LR and/or 1, 2 buttons under the flip out LCD viewer, on the side of the camera. A red LED next to each of the four inputs light to let you know it’s chosen. 

The red LED clip lights flash to let you know the level is too high, as does the Red/Green LED mounted on the edge of the camera lens. This is a very useful feature because the headphone level is not easily adjustable. You have to muck about with the LCD to get to the adjustment and if you’re using the camera-mounted mics, they will pickup your manipulations. In a loud sound field, maybe a live band performance, you may not be able to rely on it to hear what you’re recording properly.

AUDIO
The Q4 and Q8 audio specs are pretty much the same; as high as 24-bit, 96 kHz and down to 64 kbps AAC compressed. You can only record uncompressed wav files with video. You can also shoot video with no audio simply by not turning the mics on.

Although it seems to escape some folks, the Q4 and Q8 can be used just to record audio; all the way up to 24-bit 96 kHz or at the lower AAC compressed rated to sacrifice quality for space. 

Zoom Q8 with Optional Mic Modules
The sound quality was very good. I was a little concerned about the Q4’s lack of balanced inputs for professional work. The Q8 addresses that with a pair of combo 1/4” TRS and XLR inputs in addition to the camera-mounted mini X/Y mics. That gives you four channels of input. 

The LR X/Y mics that come with the Q8 use the same modular connection that the Zoom H5 and H6 use. That means you can use the other Zoom mic modules; the XYH-5 X/Y mic capsule, XYH-6 X/Y mic capsule, MSH-6 Mid-Side mic capsule, SGH-6 Shotgun mic capsule and SSH-6 Stereo Shotgun mic capsule. That opens up the possibilities a lot, but those mics are still mounted on the camera and they will pick up hand held noises. 

Regardless of which modules you use on the LR mic, the result will always be a stereo track or dual mono track if the mic is a mono mic. Most DAW software will let you easily split a stereo pair. I was told by "Andy" in the NY office that the mono SGH-6 shotgun capsule feeds both LR channels, but I didn’t have one here to try. 


MIXER SCREEN
The Q8 Mixer screen, activated by pressing the fader icon on the lower left of the screen, allows quite a bit of audio routing and processing. Pans are selectable for both the LR X/Y mics and the two XLR inputs. Low cut at 80 Hz, 120 Hz and 160 Hz. There is also a Mid/Side feature in the mixer that decodes Mid/Side into LR. 

LIMITER-COMPRESSOR-LEVELER
The three inputs (LR and 1 & 2) each have simple limiters, compressors and levelers that are selectable from the mixer screen. There are no controls, they are either in or out. You can choose only one for each input. 

I didn’t care for the compressor because moments of low level sound made it dive into the noise floor and bring up all sorts of ugly things. Probably in a more consistent sound field, it would be more helpful. The limiter works well and the leveler works like a slow compressor, only bringing up the noise floor after longer periods of silence. I could whisper into my hand-held Audio Technica AE5400 condenser mic and then speak at normal volume and it tracked me, not seamlessly, but surprisingly well. 

It was only after going through this round-robin of audio inputs and operations that I found that the pot on the LR mic was defective. It dropped the left channel at the bottom and top of the adjustments and was scratchy and went dead on one side several places in the middle. 

There's also a free downloadable software called HandyShare for Mac or PC that was released in 2010. Updated to version 5.0.0.6 in April 2015, it supports audio and video file trimming and sound effects for the line of Zoom products, including Q2HD / Q3 / Q3HD / Q4 / Q8.

Technique, Inc. © Copyright 2015 All Rights Reserved
Contact Ty Ford at www.tyford.com

Q8 Features At A Glance

• Records directly to SD, SDHC, and SDXC cards, up to 128GB
• High-quality 160° wide angle lens with selectable viewing angles (F 2.0/16.6mm) 
• Uses a system of interchangeable audio input capsules that can be swapped out as easily as the lens of a camera
• Supplied detachable stereo X/Y microphone (XYQ-8) 
• Compatible with all Zoom microphone capsules (sold separately) 
• Full-color rotating LCD touchscreen
• Support for five HD video modes, up to 2304 x 1296 pixels (3M HD), as well as two WVGA modes
• Frame rates of up to 60 fps for capturing fast action video and slow motion playback
• Three imaging sense options (AUTO/CONCERT/NIGHT) for use in all lighting environments
• Self timer
• Digital zoom
• Records video in MOV format (with or without audio) 
• Records audio in AAC and WAV formats, up to 24-bit/96k
• Up to four tracks of simultaneous audio recording
• Two mic/line inputs with XLR/TRS combo connectors, each with selectable phantom power and -20dB pad
• Analog-style gain controls for each input
• Built-in audio mixer
• Stereo link function
• Built-in compressor, limiter, and leveler
• MS Matrix function converts Mid-Side signal from external mics to standard stereo
• Supplied foam windscreen and low-cut filter for the elimination of low-frequency noise and rumble
• Dedicated Headphone output / Line out 
• Built-in speaker for fast monitoring
• HDMI video output for connection to HDTVs, selectable between NTSC and PAL
• USB interface for live streaming and data transfer to and from computer editing software
• Webcam and USB mic function for use with external devices such as computers and iPads (Apple Camera Connection Kit required) 
• Direct monitoring for zero latency during use as a USB mic
• Compatible with USTREAM Producer, Flash live Media Encorder, Skype, and other popular streaming applications
• SD card reader function
• Rechargeable Li-Ion battery (chargeable via USB or optional AC adapter) 
• Battery life of more than 2 hours 
• Standard tripod mount
• Includes tripod-to-three-prong converter for compatibility with all action camera mounts

Rode NTR - A New Bi-directional Active Ribbon Studio Mic

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The Rode NTR ($799) is an active, bi-directional ribbon microphone that surpasses vintage designs primarily because of it's sensitivity (-30.5 dB re 1V/Pa (30mV @ 94dB SPL) ± 2dB @ 1kHz) and tone. In side by side comparisons with Rode's NT1 studio condenser mic, the NTR was 3 dB more sensitive. That surprised me!

Your father's or grandfather's ribbon microphone required so much mic preamp gain that increased noise floor was inevitable. How did they manage back then? Really good mic preamps, usually with a special transformer on the input to step up the voltage, and placement of the mic itself. Plus, early "hi-fi" systems didn't have anywhere near as much bandwidth as a 24-bit recording (or even an mp3). 

How much ribbon you are moving also figures into the formula. That can be achieved by the length of the ribbon or how many ribbons were used in one mic. The Beyer M160 and RCA 77 series ribbon mics grabbed some extra sensitivity by having two ribbons. Think about it. You basically have a relatively fragile strip of aluminum floating in a magnetic field. A sound happens, the ribbon moves sympathetically and that movement within the magnetic field generates a small voltage. 

A lot of "magic" was invested in the early ribbon mics. The thickness of the ribbon was and still is important. The Rode NTR ribbon has a thickness of 1.8 microns. 

Each ribbon is run through a crimping mill that imparts a pattern on the aluminum. The crimp is a "special recipe." This is done to provide the ribbon with some structure to strengthen it and allow it to be tensioned properly. 


Longer ribbons generate more voltage, but longer ribbons are more fragile. The ribbon must be stretched to achieve a linear frequency response. As any ribbon mic ages, the ribbon material will sag and change the response. According to Rode Marketing Manager Scott Emerton, "Our engineers spent a lot of time optimising the ‘crimp’ of the ribbon to minimise any slack being introduced. Also there’s a multi-stage tension and relaxing process for each ribbon as it’s placed in its motor which increases that life."

Addendum: Rode covers you with one new ribbon for free. They also warranty the NTR for a year and for ten years if you register the mic at www.rode.com/warranty.


Here's an image of the crimping and tensioning jig with a new ribbon spanning across the gap and held in place by the two brass blocks. For a motion picture of this procedure, here's a YouTube link. 


Equally special is the step up transformer that increases the sensitivity voltage. The video also shows the automated transformer winding apparatus and details. The final piece, in the case of the Rode NTR, is a low noise amp stage to provide more sensitivity. Rode sources the smaller circuit components, but Emerton says the bodies, electrics and transformers are made at Rode in Australia and the microphones are built there as well. 

Ribbons can have more mass than diaphragms on condenser microphones. That prevents them from picking up higher frequencies. Most ribbon mic have great bottoms, but there's not much going on over 10 kHz. The published response curve of the NTR shows a gentler slope than traditional ribbon mics and even shows a slight but very useful peak bump at 3 kHz and from there a slope to -10dB by 13 kHz. There is a return by 16 kHz. 


Having made that point, the NTR is not a bright sounding microphone, nor are most ribbons anyway. That means they're great for use on sound sources that have a tendency to be harsh, brittle or distorted. Flute, violin, guitar amplifiers, banjo and bag pipes come to mind. There are even some acoustic guitars that sound pretty nasty that will benefit from a ribbon mic like the NTR.

PROXIMITY EFFECT
Proximity Effect is the tendency for directional microphones of any sort to be more sensitive to lower frequencies the closer the microphone gets to them. How close you may be before being in the proximity effect field remains open to discussion. Is it a matter of inches or feet? To some degree it depends on the microphone. The ElectroVoice RE15 and RE20 dynamic microphones, for example, have reduced proximity effect because of the acoustic slit that compensates for being closer to the microphone. 

With the NTR and a rosewood body acoustic guitar, I found the proximity effect noticeable to begin to build at a distance of about two feet. You can probably get closer with a mahogany body acoustic guitar. The challenge is what to do with all of that low end. Fortunately, The NTR has enough sensitivity to allow for subtractive EQ to tailor the sound. In my studio, I had no trouble getting good levels at two feet, even while fingerpicking. If your space is noisy or has difficult acoustics, you may have some audible acoustic problems when trying to record that far away. My advice is to fix the acoustics of the space first. Then everything you do will sound better.



Here on the left is a video of a Martin Grand J-28LSE. It's a baritone guitar tuned four semitones (frets) below standard guitar tuning, so it sounds sort of HUGE. A lot of low frequency energy. If you get a mic too close to it, it won't sound very good.


As you watch and listen, you'll hear that using the neck joint position or being at least two feet back really improves the sound. Otherwise, you're in the Proximity Effect zone and the low end is overpowering.




This video on the right features the Rode NTR and a standard six-string acoustic guitar; in this case a rosewood Martin D28s with middlin' dead strings. Here I demonstrate the lobes and nulls of the Rode NTR and the horizontal and vertical mic orientation at the neck joint, sound hole and bridge.

The guitar is a bit woofy until you get at least a foot away. Of course, you can always reduce some of the low frequencies the EQ before or after you record. After shooting both of these clips, I began to really appreciate how well the NTR handles voice.


MY APOLOGIES. BLOGGER LAYOUTS AND COMPOSITION ARE VERY BUGGY. THIS BLOG IS MADE WITH BLOGGER AND I HAVE ONLY LIMITED AND ERRATIC CONTROL OVER LAYOUT. WHAT YOU SEE IS NEVER WHAT YOU GET! 

Where was I? Oh, right. The video above has three of my favorite tests for mics; Front-To-Back for directional mics, Hissing for phase anomalies and beaming and the Dreaded Key Jangle Test for seeing how well a mic handles transients. The Front-To-Back quickly tells you if your bi-directional mic sounds the same on each side. That's very important if you plan to use the mic for Mid/Side recording. Hissing is your own personal "white noise generator." If the mic has a beamy, uneven response, you can hear tonal changes as you rotate the mic. The Dreaded Key Jangle Test tells you how good the transient response of the mic is. Each in its own simple way reveals something about a microphone with absolutely no test gear. 


Last night we tried the Rode NTR out as a drum overhead three feet over the kit in a simple live to disc recording of a five piece group at a house concert. The space was a living room/dining room with a vaulted ceiling, wood floors and a stair well.

There was one small PA speaker on a stand and one floor monitor. The group consisted of electric bass, small drum kit, acoustic guitars with pickups, mandolin with pickup, acoustic piano and at least two vocals. Aiming the NTR face down from 3 feet above the drum kit meant it's nulls were left and right. The performers were positioned closely together. I was impressed by how little of everything else but the drum kit got into the Rode NTR. The side nulls were doing a great job. 

Where else do you use the NTR? I think it would do a nice job warming up a dulcimer, a small parlor acoustic guitar, Baby Taylors or even Martin Backpackers, especially when close miced. Guitar amps are also a logical choice.

I find a number of vocalists have a nasty little peak around 6kHz. If you can keep them from getting too far into the Proximity Effect woofy space, the Rode NTR can smooth them right out. I was impressed when I heard my own voice on the video playback of the clips that are part of this review. I used no EQ on them. It was the Rode NTR going right into my JVC HM650 camera. On small speakers, laptops and phones, the sound works just fine. On larger more critical monitors there's the unmistakeable low end density of a ribbon. A slight LF EQ adjustment there and the voice works very well and has a very natural and smooth quality. 

I'm a member of the local SAG-AFTRA Conservatory. The Conservatory gives workshops to local union members to help advance their craft. I recently gave a narration class and setup a Gefell M71 and the Rode NTR side by side. The NTR was several inches farther away. Here's SAG-AFTRA talent Sheila Blanc. If you like her read and would like to use her for VO work, she may be reached at: sheilablanctalent@gmail.com

Addendum: I did additive and subtractive EQ on each of her clips to taste.

IN CONCLUSION
The Rode NTR has enough sensitivity to allow for EQing, if needed. And if it weren't apparent from what I've said so far, you can minimize the amount of EQ needed by how close or how far away you position the mic from the source. That may require a "head change" if you've just been sticking a mic in front of a source, hitting the record button and trying to deal with EQing the sound during mixing. 

A lot has been written about the mic placement techniques of bi-directional ribbon mics. Once you wrap your head around the sacred geometry created by using bi-directional ribbon mics in a space and where you need to place musicians and instruments in that space, I suspect you'll be making some very good recordings with fewer microphones. 

Technique, Inc. © 2015 All Rights Reserved
Contact Ty Ford at www.tyford.com

Location Audio Rental Houses Speak Out About Gear And Changing Markets

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Glen Trew
The relative health of the film and video production industry in the USA can be measured by the profusion of location audio rental houses. Glen Trew, owner of Trew Audio, with stores in Nashville, Toronto, Vancouver, LA and newest Atlanta location is the perfect model for this growth pattern. Trew says, “LA is #1. Atlanta is now considered #2. NY is #3. There are more feature films shot in Georgia now than LA, but a lot more TV series are shot in LA. New Orleans is a very close #4.”  

Rich Topham on the teaching circuit
Richard Topham of Professional Sound Services runs shops in NYC, Ft. Lauderdale and New Orleans. Topham notes the widespread growth and the tendency for it to follow the money. “Yes, they’re building studios in Atlanta, but they’re building everywhere; The Dominican Republic, Columbia, New York, New Mexico, Michigan. Wherever the incentives are, the production goes, but it changes depending on who’s lobbying the hardest and what else is happening.” It’s very clear that, at the moment, Georgia has one of the better incentive plans. These plans are typically adjusted annually. The hot state today may not be so hot tomorrow. 

While the big hubs keep the large projects turning, everyone I spoke with agreed that there is widespread growth. According to Trew, “The affordability of the new digital video gear has had a noticeable impact on audio rentals. Film has all but fallen off the face of the earth, and for every film camera not being used there seems to be five more affordable digital cameras that don’t require film stock. Lower cost camera equipment, and the lower cost and ease of editing has resulted in a lot more content but at reduced budgets and usually dramatically reduced quality in the final product.”

Guzzi and Schneider expand into Atlanta
Jim Guzzi and Peter Schneider own Gotham Sound. They follow suit, with offices in NYC and also in Atlanta. Schneider’s take is that Gotham is actually in the glue business. “We’re gluing manufacturers together, gluing products together, gluing customers together. We compete on the basis of knowledge and support and applying the technology in creative ways for both sales and rentals.” On the high end, that direction has led Gotham outside the box to invent new systems to handle the complexities of today’s production. 

IT’S IN THE MATRIX
Rentals for reality television, for example, have grown increasingly complex. Schneider says Sound Supervisors are often requested to make a mix for each camera. “The competition shows have relentless schedules and the technology, as the manufacturers sell it, doesn’t exist to route all of that audio in real time as quickly as they want. The new Dante platform prompted us to create a custom 32” touch screen surface for the control room with a 250 crosspoint matrix, for all of the audio sources and all of the audio destinations. 

Gotham Sound Switching Matrix
You have one operator routing audio. That person makes multiple mixes from wireless lavs and booms.” Schneider says the boom ops don’t wear bags. They use Sound Devices MixPres for headphone and level control and go wireless to the matrix. Levels are also controlled by the sound supervisor for the record/overall mix chain and for the touchscreen.

Schneider wrote code around a program originally written for interactive multimedia dance performances. Guzzi says it got a lot of interest from Sound Supervisors. “Some Sound Supervisors are true visionaries. They understood it and also saw how it would help them. So, yes, a bigger upfront cost, but it would give them the control they wanted at the speed they wanted.”

For less stratospheric work, Guzzi say their rental business echoes Trew’s ‘more projects, lower budgets’ business, “Maybe it’s just a Zoom, a Rode and a G3. In the middle are sound mixers that are augmenting their kit or renting additional pieces for a particular job. Then we have broadcasters and feature film rentals on the high end. Sometimes we have to put together 30 packages on a friday afternoon, but that keeps us from being worried about being dependent on one or two large clients.”

TOOL SCHOOL
All three companies practice educational outreach. Schneider says renters can come in and sit with the techs for as long as they need to to understand. “We had one customer that does a lot of hunting and outdoors shows where the producer does the sound. He really wanted to use a Sound Devices 633, but didn’t know if the crew could make it happen with two wireless and a shotgun. They brought in all of their producers and we had a free class for them for six to eight hours and now they’re all out in the field and doing pretty well.” 

Schneider says they typically don’t charge for education, because it usually results in more business. “We held a ‘Field 101’ class and charged $10 per person, just to get a handle on how many people would show up, but we gave it back as a $10 store credit.” 

SPECTRUM SQUEEZE
Topham says, with the 700 MHz band gone and the 600 MHz band on the way out, wireless manufacturers are going to have to work hard to meet the growing demand for wireless mics. “The Federal government only cares about the money they can get from telecom companies and Google and people like that. When you dangle a $27 billion dollar carrot in front of the government, who is not getting any income from the current end users, and you look at the sales differences between wireless mics and cell phones and iPads, consumer electronics win the day simply because of the numbers. The FCC is saying they won’t sell off the 600 MHz band until 2016 or 2017. That’s not really that far away. That may drive some to rent now rather than to buy now only to find they can’t use that gear as early as next year.”

Audio-Technica System 10
Topham points to the new less expensive digital wireless systems from Audio-Technica, Sennheiser and Rode operating at 2.4 GHz. “They work up to about 150 feet. Some people are trying them when range is not an issue. “If you need more distance than that, you need to buy or rent higher.”

Topham says there’s another RF gotcha. “If a mixer who already owns block 24 gear gets on a show and production says, ‘No, we’re using block 22’, then the mixer will come to me to rent block 22 because their 24 won’t match up. Especially in reality shows, because the crews switch in and out, they all need to have receivers in the same blocks as the transmitters on the actors; one transmitter, but multiple receivers. They want the sound to transmit to the cameras. They want those wireless on certain blocks so if they move camera with sound, everything works together.” Topham says that’s why wireless manufacturers are now putting out wireless mics that cover three blocks, or 12 UHF frequencies.

TIME CODE
Mozegear Tig Q28 Time Code Generator
Topham says time code slates and other boxes have become more important, especially with the newer cameras. “There just aren’t a lot of cameras out there with solid clocks. From an F300 to a Lexa to a RED, the time code is no good. The Ambient clocks inside the Sound Devices gear, Zaxcom, Denecke and Mozegear clocks are all very reliable. That’s why you have Lockit boxes on the side of all these cameras. Zaxcom makes an IFB that carries audio and time code, so you can feed both into a camera from the same box. Camera operators like this because it’s one less box hanging off of their cameras.”

Many of the new, less expensive digital cameras don’t even have time code connections, but Trew adds that while there used to be three types of time code connectors, “Now there are connectors that are sometimes unique to each camera. So we have to find out what devices are being used and what interface cables are needed, and then help people understand how to use them.”

Timecode Buddy :mini trx
In addition to Topham’s time code gear list, Trew adds Betso and Timecode Buddy. “Betso had some consistency problems when they first came out, but they seem to have fixed them. Timecode Buddy is a very nice system that incorporates metadata, slates, and script supervision. Mozegear is the newest, but seems spot on in quality and utility. Industry leaders Denecke and Ambient continue to be excellent.”

Sound Devices 633
The two major players for recorders and mixers are Sound Devices, who followed their 788 and 664 with the 633 and 688 compact mixer/recorders, and Zaxcom who recently introduced their own Nomad and Max compact mixer/recorders. Having said that, people coming in for rental appreciate the Zaxcom gear that uses their “NeverClip” circuitry that lets you scream into a mic without overloading it. Wireless systems suggested by the rental houses include Lectrosonics, Zaxcom, Sennheiser and Wisycom. Boom mic choices continue to be Schoeps CMC641 and CMIT, Sanken CS3e and Sennheiser MKH 50. Lavs mentioned range from DPA 4061 to Sennheiser ME2 and MKE-2, Sanken COS-11, Tram 50 and Countryman B6.

When renting from any shop, some price by three day weeks and some by four day weeks. For example, three day means, you pay for three days but get the gear for five days. Some shops have monthly rates, others just stack weeks to make a month. 

This story is a digital reprint from the June 2015 issue of Markee Magazine. You may read the entire June issue here.

Technique, Inc. Copyright 2015 All Rights Reserved

Contact Ty Ford at www.tyford.com

Sennheiser MO 2000 Industrial Optical Analog Microphone

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Sennheiser MO 2000 Optical Microphone
This is not a review of a studio toy. It is, however, about audio; industrial audio. I had been hearing about a Sennheiser optical mic. Was this for studio use? Is it analog or digital? How does it work? I reached out to Sennheiser and within a week it was on my door step.

The Sennheiser MO 2000 system consists of a smallish half-rack-space box, a two-way fiber optic cable capped with an omni element. The system is powered by a wall wart power supply. There are both coarse and fine gain controls on the front of the chassis and a simple, lighted Off/On button. The omni mic element of the MO 2000 has a frequency response of 20Hzto 40kHz (+/- .6dB). 
MO 2000 Rear View

The analog balanced output impedance is 600 Ohms, unbalanced 33 Ohms. Analog outputs, both balanced XLR and especially the unbalanced BNC connector, convey that this is some sort of industrial gear. With an SPL of 134 dB, the environment can be at the pain threshold. With a S/N ratio over 50 dB (A weighted), the selfnoise of the microphone will prevent it from being used for critical applications in a recording studio, but that’s not its intended application.



The omni transducer works by shining a lightfrom an LED source generated in the box up one run of the dual fiberoptic cable onto a non-conductive diaphragm made of composite plastic. Sound moves the diaphragm and that movement varies the amount of light that is reflected back down the other fiber optic run. The variations in light intensity are received by a photo diode back at the box, converted into a voltage swing and there’s your audio. 


When would you use a noisier than studio standards microphone system with no metal parts? Maybe as a talk-back mic in a recording studio that pumps pure oxygen in to elevate the mood of the musicians? Inside or near a hydrogen filled dirigible? In corrosive environments that would corrode the metal? In high magnetic fields where the metal may become dangerously projectile; like in a particle accelerator or in an MRI. 

Non-conductive omni microphone element
An email to Sennheiser’s Vanessa Jensen, Sr. Product Specialist, Integrated Systems quickly got to the point. Yes, the MO 2000 system is designed for (but not limited to) deployment in hazardous environments where the absence of electrically conductive components in the transducer avoid the risk of creating sparks or generating heat. The MO 2000 can operate safely in otherwise explosive environments such as for the acoustic monitoring of gas drying plants in natural gas production. In this case, Jensen reports, “The microphone can ‘hear’ slow leaks, which, due to their small size, cause neither a pressure loss nor an alarm message in the other monitoring systems.” So, yes, industrial uses, provided the leaks are noisier than the selfnoise of the MO 2000. The MO 2000 microphone itself is not submergeable in water or presumably in other fluids.

In medical applications, the MO 2000 system can be used in magnetic resonance imaging (MRI), for example to maintain communication with the patient during MRI scans or to achieve active noise cancellation in MRI. The optical microphone does not disturb the imaging process and is itself not influenced by the strong fields inside magnetic resonance imaging equipment. In measuring technology, the microphone is ideal for use in EMI/EMC laboratories, as it does not influence the electric field.

Optical Connector for MO 2000 Microphone
Further, the MO 2000 complies with the requirements regarding intrinsically safe optical radiation according to EN 60079-0:2004 and EN 60079-28:2007. It also meets the requirements of the ATEX Directive on explosion protection, which was documented with the Type Examination Certificate TÜV 07 ATEX 553824 and TÜV 07 ATEX 553825X. This certification allows the MO 2000 to be used in potentially explosive atmospheres of Zone 1 (e.g. gas processing plants). The MO 2000 has the degree of protection IP 54 (5=protection against dust deposits, 4=protection against splashing water from all directions) and is therefore suitable for outdoor applications.

Technique, Inc. © Copyright 2013 All Rights Reserved

Reach Ty Ford at www.tyford.com











Audio-Technica AT5040 Quad Diaphragm Cardioid Condenser Studio Mic

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Four capsules are visible
inside the two-layer grill
For years, Audio-Technica has brought solid, economical workhorse mics to market for broadcast, video/film and recording. If they have any cross to bear, it's that they frequently have stayed away from the spotlight. Yes, they get street, live and studio cred for many of their mics as solid performers; just not a lot of of icing on the cake. I think those days are over.

The new Audio-Technica 5000 Series has begun. First out of the chute is the AT5040 electret cardioid condenser mic and accompanying AT8480 mic clip. It's an electret. If that makes you wince, it might help you to know that on several occasions over the years, I was assured by people who had been making top-shelf mics for major companies, that there was no reason that electret mics couldn't be made as good as externally polarized condenser mics.


QUICK NOTES: Application: Studio and booth recording Features: Exceptionally quiet and extremely sensitive Electret Cardioid with four rectangular diaphragms Price: $2,995.00 with custom suspension mount

FEATURES

Designed with the aid of two anechoic chambers, one at AT Japan and one at AT US, the AT5040 has a frequency response of 20 Hz to 20 kHz. Partially because it has no output transformer, response rises gently a few dB below 80 Hz. There’s also a gentle presence boost of 2dB that begins to rise at 1 kHz, achieves 2 dB about 3 kHz to 4.5 kHz and dips back to zero at 5 kHz. There’s a slight wiggle and then a short 1dB plateau between 9 kHz and 11 kHz. By 20 kHz the frequency 
response is down 2 dB. This curve works 
very well on male and female voice, among other things.

Open circuit sensitivity is a walloping -25 dB (56.2 mV) re 1V @ 1Pa. Audio-Technica makes the point that because they were after “sonic purity” before anything, including manufacture cost, there are no switches, pots or transformers to degrade the output.


Rectangular diaphragms have been used in other microphones; notably Sanken, Pearl and Milab, so this is not a first. The combined area of the four diaphragms is the theoretical equivalent of a round diaphragm of about an inch and a half in diameter.

Audio-Technica AT5040 from the inside

A single round diaphragm 1.5 inches in diameter would be problematic because of size and mass. If you could make a usable round diaphragm that big, there could be two benefits; low selfnoise and high sensitivity. The selfnoise of the AT5040 is 5 dB-A. That makes it one of the quietest mics on the planet. The increased diaphragm size also makes it one of, if not THE most sensitive mic on the market, requiring less preamp gain than probably every other mic out there. It's 9 dB more sensitive than a Neumann TLM 103 and just as quiet. 

Where would you use that extra sensitivity? Where the source and ambient sound are very quiet, this mic will shine. Also, because the AT5040 has no output transformer, the output has a wider bandwidth. I could see the low frequency components on the waveform when I zoomed in on the timeline that I have never seen with any other microphone. 
AT8480 Mic Clip

The mic clip for the AT5040 is the AT8480. By itself it's a work of art that one might find in MOMA. It holds the mic gently but firmly, while allowing the mic to be turned for positioning. 

IN USE
Let’s look at the AT5040 as a booth mic. Even though the AT5040 is as flat as it is, there's enough sparkle to do very nice things for male and female voice. The AT5040 has a rich clarity on male voice with no harsh edge and a slight, chesty thickness that reminds me of days when I used to smoke half a pack a day (or more). It has sort of a ribbon mic quality on the low end,  while remaining smooth and bright on the top end. Here's my first recording of the AT5040 into a Sound Devices 744T recorder, recorded in my living room.

I work regularly with VO talent Molly Moores (www.mollymoores.com). I record her for a flight of radio and TV VOs every month. Molly has a great voice, but with the wrong mic, her sibilance can peek out a bit too much, especially when rushing to get all the copy in and compressing/limiting to increase the punch. We tried the AT5040 on her as well and found that we didn’t have to use any EQ. I might have nudged 125Hz up slightly, but the AT5040 was very complimentary to her chest tone and again, no edginess. Listen to a raw Molly Moores voice track with no EQ on SoundCloud.

I was concerned that off-axis sound across diaphragms this large would result in scattering and messy phase response. I worked the mic from each side, top and bottom in search of some sort of smeariness or beaminess, but found none. There is a fairly narrow angle of acceptance for high frequency response. Anything more than 20 degrees either side of the centerline and the high frequencies begin to roll off. The rolloff is well-behaved.

Depending on the abilities of you VO talent, you may not need a pop filter, but it’s not a bad idea to have one for talent who haven’t learned how not to pop a mic. Also because this mic actually hears some very low frequencies, it might catch some breath eddies. The AT8480 shock mount is exceptional in design. It gives the mounted mic a very finished look as well as being highly functional and very easy to use. The non-reflective finish of both the mic and suspension mount would make the pair a likely candidate for the desk on some upscale TV talk show.


Drummer Mark Ayers and Bassist Al Page
Next I tried the AT5040 with a Greg Hanks BA-660 tube mic preamp on a kick drum during a music recording session. I was concerned that the AT5040 diaphragms might be damaged, but got the go ahead from Audio-Technica to use it on kick. 

Drummer Mark Ayers plays with both heads on his 20" Dominion Duo Fade ddrum with Evans EMAD beater head and doesn't care for "click" on the kick, so, as shown in the picture I tried setting the mic up aimed across the hole in the front head. We cut a track and Mark liked what he heard. We moved on. Bassist Al Page plays a 1966 Gibson EB2.

Here's Russ Beaumont, guitarist for the above group on a Les Paul > Fender Amp > Audio-Technica AT5040 > Greg Hanks BA-660 preamp > RME ADI-8 DS > Protools.


SUMMARY
It's very difficult to improve on the best existing microphone designs. While most mics on the market these days are the result of a mad rush to be insanely affordable, having the vision to innovate a top-shelf mic that is unique as well as outstanding takes a lot of guts. 

Combining four rectangular diaphragms effectively is a major feat of engineering. The AT5040 is the first in Audio-Technica's 5000 Series. It will be very interesting to watch what they do with this grand new effort. At $2999, the cardioid-only AT5040 will be out of reach for some buyers, but so are Volvo, BMW and Mercedes Benz. The AT5040 makes a statement and repositions Audio-Technica as a high-end mic manufacturer. As people find out more about what the AT5040 brings to the party, I expect it will gain even wider acceptance. 

Ty Ford has been reviewing professional audio gear for over 20 years. Find out more about him at www.tyford.com.

Technique, Inc. © Copyright 2013 All Rights Reserved



Greg Hanks BA-660 Tube Microphone Preamp aka "Monster"

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Greg Hanks BA-660 Preamp/Limiter


OK, here's the deal. I've been communicating with Greg now for about a year and we have had many deep conversations, parts of which make my brain hurt. 

Yes, I do know a reasonable amount about audio engineering and manufacturing and I have good ears. Enough to hear some very subtle differences and ask questions that sometimes embarrass me and sometimes embarrass the person I'm asking. 

The only reason I ask questions is because of the reader; the person reading what I've decided to write. If I can't convey the information properly, I'm wasting my time and your time. I say this because I want you to know I'm thinking about YOU as I try to make sense of every piece of equipment.

When Greg first sent me the BA-660, there were some issues. There was something strange in the output section, the meter was a little wacky and there was fan noise from the cooling fan. He found solutions very quickly. The fan thing was a request from me. I told him that many recording environments today don't have separate control rooms and studios. Even if they do, I find  fans distracting in a control room. So, there’s a new multi-speed, temperature controlled fan that, at its lowest speed, I can't hear unless I put my ear within four inches of the BA-660. When planning where to put the BA-660, leave a little extra breathing room and the fan will run slower (and quieter).

The BA-660 is very different for a number of reasons. Yes, it's a tube preamp and limiter; nothing too different there except that it's very, very quiet and very, very clean. Greg chose tubes not for color but for capacity and how well he can get them to do what they do. For example, the preamp path has a frequency response of 7Hz to greater than 28kHz (+0/-3). 

The Line In path (so you can use the BA-660 just as a limiter) has a frequency response of 6 Hz to 36kHz. You may never need it because your mics and other sources may not have that kind of response, but it's there, waiting like a impeccably-dressed chauffeur in a yet to be imagined super car, ready to take you somewhere special.

Phantom Power (well, voltage really). While the BA-660 is designed to provide 48V DC, it can also crank out 300V DC @ 5mA for your Bruel and Kjaer mics. The BA-660 can handle balanced from minus 90 dB to +40 dB. This is possible due to a very complex input stage with relays, pads and transformer that are employed as needed to keep things under control.

BA-660 Rear Panel
The rear panel is simply laid out; IEC power connector, signal ground, chassis ground, balanced line level out via XLR, a -10 balanced Insert Send that can also be operated unbalanced, an Insert Return that runs at +4 and follows the preamp and line in circuitry, a Side-chain Input that runs at +4 balanced, the input to the dynamics control system linking multiple BA-660s, a balanced Class A floating solid state output capable of +34 dBv and will drive loads as low as 150 Ohms.

The front panel (see above) is populated by a wide-ranging input selector covering mic and line levels. The wide range is achieved by a combination of relays, resistive termination and a transformer. An input overload LED activates when the output of  either the first or second input stages exceeds +34 dBv internally. 

An Insert Bypass switch allows manual bypass of the Insert Points accessible on the back panel. These allow you to hard-wire bypass anything you may have plugged into the Inserts instead of making you crawl around behind the rack. When the jacks are unused the Insert circuitry is automatically bypassed. A continuously variable Input Trim pot allows +/- 10dB of range to establish proper gain staging. After that, the fun begins.


A Polarity Reverse switch is provided for the Mic and Line Inputs and is implemented between the Insert Returnand Gain Cell. 


There are two (Three?) metering systems with the unit; LED Type Audio metering which measures the audio signal at either the input to the gain cell or at the output of the unit as selected by the input meter select described below. This meter combines VU type metering which shows up as a ‘dancing dot’ around the threshold control and Peak type metering which uses the same LED’s as the VU but shows up as a moving bar and measures in 10 db increments. 


There is also a mechanical (analog) Gain Reduction meter in the center of the front panel. A Meter Zero adjustment allows for calibration of the Gain Reduction meter. A Limiter Tip In adjustment is provided to accommodate different vacuum tubes properly. 


The Input Select switch toggles the audiometer between Gain Cellinput and the finaloutput. A Gain Reduction Bypass button provides a hard bypass of the Gain Cell responsible for the limiter.  


Not too well-versed with compressor/limiter settings? No Problem. The BA-660 has both fixed and variable presets, five of each. According to the simple but well-written manual, these presets emulate a Fairchild 660, LA-2, Neve Console Bus Limiter, SSL Quad Limiter and RCA BA-6. You can trimthe attack, release and ratio of the variable presets or use the fixed values and make changes with the input level, threshold control and make-up gain control with 20 dB of gain.


The biggest dial on the front panel adjusts the threshold of the limiter from -40 to +20. There is aGain Cell Clip LEDto the right of the threshold knob. If you see it blinking and don't think the audio sounds bad, congratulations, you're a punk outlaw with no regard for fidelity. That's not a defamation, just an observation. 


Although you can adjust the attack timefrom 3ms to 35ms, it'll sound pretty gnarly at the fast settings. So for the second time in a few sentences, I'm saying it's not particularly difficult to make your audio sound really bad. It's a choice!


Release time varies from 45ms to 1.8 seconds. Faster is louder and pumps more. The Ratio control varies from 1:2.1 toa little over 9:1. A maximum of 18dB of Gain Reduction is possible before the circuit behaves like a fixed attenuator. At that point many of your clients who don't care much about dynamics or fidelity will be thrilled...and probably flattened by the sheer density of the sound. This is where you, as the recording engineer, leave the room for a few minutes to protect your ears, turning up the monitors as you go, to let your clients bathe in their own reverie. Actually, it's healthier if you have a remote control for the monitors so you don't have to be in the room when you GO LARGE with the volume.


Is the BA-660 too much of a beast for you? Concerned that you're too much technology? Greg has written elegantly simple setup directions and posted them on his web site. Here they are. Not as simple as a dbx 160, but not so scary either.


Here are some links to audio clips recorded with various mics and the BA-660.


AT5040 > Greg Hanks BA-660 Kick Drum


Gibson Les Paul>FenderAmp>AT5040>Greg Hanks BA-660

D28S Martin>TLM103>GregHanks BA-660

You won't find the BA-660 at Guitar Center. As of this posting, there are BA-660s at DSP Doctor, Calistro Music and Vintage King Audio


For more details, reach out to Greg Hanks at Greg Hanks Design.

Technique, Inc. © Copyright 2013 All Rights Reserved


Rode Reporter Dynamic Interview Microphone

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The first mental image I get of a dynamic Omni stick mic is of NBC’s Al Roker trying to remain standing for a live standup during a hurricane. We always hear him just fine, even in 60 mile an hour winds. It’s been a while since anyone has taken a shot at this category of mic. How well would the new Rode Reporter compare?

FEATURES
The Rode Reporter is a classic hand-held, end-address, dynamic omni stick mic in a long-handled form factor. It comes in a designer box with zipper pouch and mic clip. The 10.7 inch shaft is three inches longer than the industry standard ElectroVoice RE50, allowing a few more inches of reach for inquisitive video journalists. A two-sided clip-on flag holder is provided for station logos. Unlike square flag holders, this requires the user to make sure the mic is held with the flag broadside instead of on edge to the camera so the flag can be seen. 


Rode Reporter Comes With Flag
That’s where the Reporter’s unique shaft shape comes in handy. It’s flattened on one side. While on-camera talent is trying to make their point to the camera, all they have to do is hold the shaft so that their thumb is on the flat side and make sure the flag is turned so that it faces the camera. 

The body is made of a hardy die-cast aluminum alloy coated in a matte black, non-reflective finish. The headgrille is comprised of three layers of metal mesh, also non-reflective. The inner most screen is almost half an inch below the outer two, providing a nice space for plosives to slow down before encountering the final screen. Because the headgrille is easily removable by simply unscrewing it, you can dunk it in disinfectant periodically during flu season to prevent contagion from expressing itself from one roving reporter to another. 

The capsule itself, has a final, thin foam screen set into the handsome brass housing. It seems not easy to remove. The spec sheet quotes 150 Ohm impedance, a 70 Hz to 15kHz frequency response and sensitivity at -56.0 dB re 1 Volt/Pascal (1.00mv @ 94 dB SPL) +/- 2 dB @ 1 kHz. 

The classic RE50 is at -55 dB, the RE50 N/D is at -51 dB. In side by side tests, the Reporter was, in fact, just over 5 dB less sensitive than the RE50 N/D. The Reporter weighs in at just over half a pound, slightly less than the RE50 N/D. Here's a link to Reporter RE50 comparison files.

On the upper end, the frequency response of the Rode Reporter has a lift that begins to rise at 3 kHz and hits +5dB at 6 kHz. That will pull the voice nicely out of the mud. It stays there until 11 kHz before dropping down across 0 dB at about 15 kHz. The low end is flat to about 120 Hz, and then slopes off to be -10 dB at 40 Hz. That will push the mud a bit farther down without taking the bottom off of most voices. 

IN USE
I didn’t have those winds available for testing, but I did spend some time actively trying to pop an RE50 N/D and the Reporter. When you blow directly into either, it’s obvious that you are no longer talking, but apart from that, they both survived equally well. 

I manhandled each mic barehanded to generate handling noise. The RE50 N/D was slightly less susceptible, or rather, the frequency of what it heard was lower than that of the Reporter. Not a big difference, and with gloves on, both mics quieted down. 

SUMMARY
Well-made omni mics are supposed to sound pretty even, with no on or off axis frequency lobes. I spent some time moving around the capsule in search of beams, peaks and nulls but found no axial anomalies. The Rode Reporter is a lightweight, but solid piece of gear. To back that up, it comes with a twelve month warranty that extends to a ten year warranty when you register after logging onto www.rodemic.com

Ty Ford has been review professional audio gear for over 20 years. Find him at www.tyford.com.

Technique, Inc. © Copyright 2013 All Right Reserved

FAST FACTS

Application
ENG/EFP audio pickup

Key Features
Sturdy construction, light weight, well priced

Price 
List Price $299, Street Price $199 USD

Contact
Rode Microphones


Audio-Technica BP4025 Stereo Mic - In A Field Of Its Own

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Audio-Technica BP4025
You don't think about it much, but good stereo ambi - ambient sound - can add a lot to the feeling of a video sound track. But how do you capture it? There are many stereo mics and the cost range varies significantly; from several hundred to several thousand dollars. 

The Audio Technica BP4025 is a relative newcomer. B&H lists it for $649. It's a professional X/Y patterned mic and requires phantom power. It has a five-pin XLR output connector, 10 dB pad and low frequency roll-off. The BP4025 comes with a five-pin to dual three-pin XLR Y-cable that plugs into any professional mic input. 

A simple foam pop filter is included, but for serious outdoor work, you'll need something more wind resistant. The significant difference between the BP4025 and other stereo mics is that the BP4025 has relatively large diaphragms; about an inch in diameter. These larger diaphragms grab sound with less selfnoise than mics with smaller diaphragms. Selfnoise exhibits as high frequency hiss and can really spoil the sound. 
Audio-Technica BP4025 Grille Removed

My first test with the BP4025 was a simple walk around the backyard. I plugged the BP4025 into my Sound Devices 744T audio recorder grabbed my headphones and headed out. It was a great Spring day with birds tweeting everywhere. 

Using the headphones to help me find a good spot. I held the mic in my hand and hit RECORD. The BP4025 is so quiet that the selfnoise was well below the noise floor of my quiet suburban neighborhood. I heard absolutely no hiss. 

As I stood listening to nature in stereo, I spotted one of my neighbors walking her dog. She was about 150 feet away from me. As I watched her approach, I head a sharp clack. She's used to seeing me with audio gear hanging off me and as she got closer, she stopped to see what I was up to...THIS TIME. 

When she got within comfortable conversation range I noticed she was chewing gum. I asked her if she had clacked her gum as she was walking. Yup. That was the clack I had heard from about 150 feet away. 

The recording proved very useful a few months later when I was posting "Hot Flash", a double award-winning indie short I produced last year. There was a backyard scene in which Diana Sowle, wearing an electric dog collar, walks aimlessly across the lawn, only to be stopped by an invisible electric pet fence. The boom mic was obviously not in stereo, nor did it hear much ambience. It was not a nice Spring morning and there were no birds chirping. In fact, there was a rather noisy air conditioner across the street.

"Comes With" Accessories
The stereo ambi track fit perfectly. I chose a particular section so the scene opens with a particularly nice bird tweet. No one has ever questioned me about the sound. No one knows, until now, that the ambi was from a different neighborhood on a different day. 

You can hear for yourself at the HotFlashMovie web site. I've always felt the most important quality of great sound is that is doesn't draw attention to itself. Since then I have continued to record stereo ambi and have created a small but growing archive of great sounding stereo ambi tracks; all done with the AudioTechnica BP4025.

If you can't afford the BP4025, there's always the lesser AT8022.

Technique, Inc. © Copyright 2013 All Rights Reserved.

Contact Ty Ford at www.tyford.com

Sennheiser SK 5212-II and EK 3241 Wireless - Less Is More

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Sennheiser SK 5112-II and a nickel
Shrinking spectrum and challenges to hiding a body pack transmitter are driving location and studio audio people to find better solutions. One of these solutions is the Sennheiser SK 5212-II transmitter. 

The retail price is $2,349, for just the transmitter and another $2,100 for the EK 4231 mono receiver, for a total of $4,400. 

I found a $3,750 Bundle Price online for both transmitter and receiver that also includes the AA battery power kit for the EK 4231 receiver (batteries not included). 

There's a big price difference between these pieces and a Sennheiser G3 transmitter and receiver kit for $629.95 at B&H. Read on to discover why.



Sennheiser still sells the older SK 5212-I beltpack transmitter. They have less bandwidth; 
450 - 638 Mhz 
602 - 798 MHz
762 - 960 Mhz

The newer SK 5212-II has five ranges; 
470 - 638 MHz
606 - 790 MHz
614 - 798 MHz
614 - 697 MHz 
776 - 866 MHz. 


Not all of these frequencies are usable in the US, but I've included them because this blog receives global attention (thanks and thanks for mentioning it to others!).

So what's so special about the SK 5212-II other than its size that warrants the price? More flexibility and a better sounding compander. The transmit frequencies are adjustable in 5 kHz increments. This allows more transmitters to be operating in the same area. The SK 5212-II has three power levels; High (50mW), Low (10 mW) and LoI (10 mW, I for intermod). 


Sennheiser's Ben Escobedo
I reached out to Ben Escobedo, RF Services and Field Support Engineer at Sennheiser for more information about the LoI (low intermod) power setting. According to Ben, "The low intermodulation mode (LoI) adds a special isolation circuit in the RF signal path which reduces intermodulation, however the output power is still clamped at 10mW. The difference between LoI and regular "low" power lies in the extra power consumption of 70mA vs. "low" power mode. In fact, LoI has the same power consumption as in HI mode @ 50 mW. 

The effects of LoI are seen on a scope and it does a great job reducing intermod spikes (3rd and 5th intervals). It is most useful where there is limited free spectrum and when one is running many transmitters in close proximity to each other."



Wig Pack for SK 5212 and SK 5212 II
Theatrical & News
You'll find the Sennheiser 5000 series on Broadway fitted into opera singers' chest plates and wigs. The WMB Wig Pac features pro mesh fabric tabs that allow the Wig Pac to be securely pinned to any hair piece or wig cap. 

The hook and loop closure secures antenna and audio connectors in place and protects from slight pressure and moisture. The Wig Pac is made with fabric covered neoprene, which is durable, moisture resistant, machine washable, and latex-free.

WMB makes belt and thigh mounts from sixteen to fifty-two inches and shoulder mount packs in small, medium and large.

For "no fail" broadcast applications, the talent uses two SK 5212-II, on the left and right rear hips with a wireless IFB receiver between them in the center of the back. 

In Use

After rigging the EK 3241 receiver for battery-powered bag use with the AA battery option, I used the SK 5212-II with a Sennheiser MKE-2 Gold lav on a talent playing the part of a game show host. I wish there was something remarkable I could report, but the system just worked the way it was supposed to, no problems. 


After powering up, the transmitter screen remains backlit for a short period. In that mode the screen is very readable. You can easily toggle the main control to display frequency, name or audio level, audio sensitivity, EQ, output power and channel. 

When powered up, there's a small LED inset in the top of the transmitter that glows red. It also acts as an audio peak display blinking much brighter if the audio exceeds the proper level.

Walk Test
I've been doing walk tests to determine range around my neighborhood for a decade. I hang transmitters on a certain branch of the Mount Fuji Cherry on the front lawn and take a hike until the reception begins to drop. I tried the SK 5212-II in both 10mW and 50mW at 621.000 MHz and found they both began to fall apart at almost the same spot; about 116 yards from the tree. This is consistent with the idea that, despite the power difference, there isn't that much difference with coverage at the fringe of the signal. 

Checking the very helpful online Sennheiser Interactive Frequency Finder, I noticed a Construction Permit for WMJF LD a 5kW just five miles down the road. Never heard of them, but the station is part of Towson University. John MacKerron, who hired me to teach there as an adjunct decades ago, is now the Chair of the TV/Film Department. A long telephone call later, WMJF LD is dark, so my test accurately reflects what the potential capabilities of both power settings are. 

Golden Sound
With the Sennheiser MKE-2 Gold, the overall sound of the wireless was very smooth and no hiss to the point of sounding perhaps rolled off a bit on the top relative to a Sennheiser G2. I have a 24-bit WAV file using the MKE-2 Gold, SK 5212-II and EK 3241 available for download from my DropBox account - HERE.  Battery life for the single AA battery was very good. As we approached 5 hours, we still had juice in the alkaline battery, with the transmitter running at 50 mW.

The 3000 and 5000 Series Sennheiser wireless use HiDyn Plus companding. (Sennheiser uses HDX on their 2000 and Evolution Series.) During my use of the SK 5212-II and EK 3241, I didn't hear any artifacts. The frequency response of the system goes down to 50 Hz. The Sennheiser G3, shows specs that roll off at 80Hz. 

The SK 5212-II operates over an impressive 184 MHz bandwidth. Its input may be adjusted from -30 dB mic level to +40 dB line level in 1 dB steps. For acoustical environments with problematic low frequencies, it has a 120Hz High-Pass filter. I found menu navigation very easy on the SL-5212-II


EK 3241 Receiver with
one of four external battery options
EK 3241 Receiver
The EK 3241 analog, single channel receiver uses two antennas. It's True Diversity, meaning there are not just dual antennas, but dual antennas and dual receivers. The EK 3241 is compatible with a number of Sennheiser transmitters. Currently, there are three different versions; 

450-626 MHz
590-834 MHz
798-960 MHz. 

When you order the receiver, you specify which 36 MHz bandwidth within each of the three versions you want. You also want to be aware of what frequencies are legally available and where there are likely holes in your area before buying any gear. Like the SK 5212-II, the EK 3241 receiver is also tunable in 5 kHz steps. 32 custom frequencies have already been programmed within the switching bandwidth of 36 MHz; in addition, you can store another 20 frequencies in steps of 5 kHz into the user bank. 


EK 3241 Battery Kit
Base plate with XLR and External DC jack
Optional Kits
Although the EK 3241 receiver has optional kits for Thompson, Sony, Panasonic or Ikegami cameras, it can also be used in the bag, running on two or three AA batteries. AA battery operation requires optional kit parts, but provides a very handy operation. 

You just remove two screws from the bottom plate of the receiver, remove the bottom plate, slide the kit shell on and re-attach the bottom plate, locking the shell into place. A smaller clip shown below holds two AA cells that slide into the shell and a three battery clip is also available. There's a small LCD strip on each clip that indicates battery life.

At one point the last LCD power mark on the battery display began blinking, signaling that power loss was imminent. The receiver continued to work, but later on, after having turned the receiver off, it would not power up, even though the LCD battery meter was still blinking. I installed new batteries and the receiver started right up.
EK 3241 B50 Battery Box
with LCD battery meter.

In addition to an assortment of four battery boxes, the EK 3241 also has an external DC power jack and cable for larger, longer lasting DC supplies. There are also a headphone output and QuadPack options for four receivers. 

The manual suggests not operating the transmitter and receiver closer that five meters apart to avoid overloading the receiver.
Menus
Menu navigation on the EK 3241 is tougher than on the SK 5212-II, due to the small window, no back-lighting and menu structure. It's not a deal breaker and most adjustments are not that difficult, but I wouldn't take the receiver on a gig without getting to know it. One very handy feature is being able to read the transmitter battery level from the receiver window.

Technique, Inc. © 2013 All Rights Reserved.


TASCAM DR-60D Four-Track Digital Recorder for DSLR and other uses

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As long as DSLRs continue to be designed by video people and photographers, audio problems will probably remain. 

I had heard that the Canon 5D Mark III audio was OK and proved it here with the DR-60D by sending audio from the DR-60D to the 5D Mark III. 

The trick was using very little gain on the Mark III and using the line output gain on the DR-60D to do the heavy lifting. 

4/2014 Update: We also found that you could get a better feed into a BMCC camera this way. Line out from the DR60 to the mono 1/4" inputs of the BMCC. Set the BMCC to MIC instead of line and set the BMCC input level to 15% and the Line out of the DR60 to 3. Faint noise but more than adequate for a scratch track. 

TASCAM, and parent company TEAC, to their credit, have always empowered people to be able to do creative audio production at a fair price, especially at the pro-sumer level.


They are one of the original Architects of the Revolution in the Audio/Video Industry.
It was with a TEAC Model 2 mixer and TEAC 3340S back in 1973 that I began my freelance VO and music recording career. I don't recall what I paid for that, but it was thousands of dollars.

At $349 the TASCAM DR-60D is a four-track digital recorder with two balanced XLR/TRS mic inputs and an unbalanced 1/8" stereo mic input that records 44.1, 48 and 96kHz at 16 or 24-bit WAV files to SD or SDHC cards. That's frankly amazing. What's missing? Not much as the feature set bullet list I've included at the end  this review shows.


If you decide not to read the manual, your success with and full use of the DR-60D will be limited. At some point, it occurred to me that the DR-60D and a Rubicks Cube or maybe a PlayStation console have a lot in common. A lot of capability in one handful, but getting the right combination proves a bit tricky until you get the knack. Then it's a snap.



MORE UNDER THE HOOD
The menu system is fairly good and easy to navigate. In it you'll find a few less obvious features. RECORD MODE contains controls for how many channels you want to record, some muting possibilities, and Mid/Side recording options for both channels 1 and 2 as a pair and 3+4 as a pair. There's an obvious yellow light on the main control panel that lights brightly to remind you whether you're in 2-channel or 4 channel mode. 

The Dual Mono and Dual Stereo selections are very neat. After you choosing the DUAL mode (you can choose either inputs 1 and 2 or 3+4 as the source), you can scroll down to DUAL LVL where you can choose from 0 to -12 dB to set at what level the second track or tracks will record. Even though there are limiters on all inputs, when you're a one-person shop, this feature may save your butt when unexpectedly loud moments occur and you're not paying attention to the audio...as you should. Nice! (10/11/13 update: Steve Oakley asked for my opinion of the preamps. They were quite usable and very clean with my mics. I didn't mention them earlier because I didn't hear any problems.)

RECORD MODE is also where you'll find the Mid-Side options. Choose REC to decode while recording in mid-side and MONITOR to record in mid-side that will be decoded later, or for playback of mid-side files that were recorded without decoding.

THE GOZINTAS & GOZOUTAS
On the left side of the DR-66D are the combo XLR/TRS input jacks. Extra points for locking XLR connectors! Phantom Power is applied only to the XLR inputs, not the TRS. Channels three and four are only accessible via the 1/8" mini-TRS jack. One use of the inputs, would be to have separate mics going to the 1/L and 2/R XLR inputs and an unbalanced stereo mic. Maybe you're at a sporting event and you have two announcers, each on a separate channel, and a stereo mic picking up crowd noise.

You could also use a Y-connector with 3-4 and plug two unbalanced sources in there; maybe two wireless mic receivers. The problem there is only one knob to adjust 3 + 4. If you need to adjust one or the other, you have to do it before it gets to the 3-4 input. You could also setup a small mixer with many sources and mix them to the 3-4 input, hoping that running unbalanced would not allow noise or interference to sneak in. There's the envelope. How far do you want to push it?

The CAMERA IN and CAMERA OUT ports are poorly named. CAMERA IN is an audio return from the camera. CAMERA OUT is a DR-60D output that you send TO the camera.

THE OTHER SIDE
The left side of the case is populated by the I/O power switch, the 1/8" PHONES OUT jack and volume control, the 1/8" LINE OUT and volume control and a rubber covered port that conceals the USB 2 jack and SD card slot. There's also a HOLD button. It may be one of the handier buttons. Slide it to the up position and the controls are deactivated. Do that and then hand it to your friend who NEVER reads the manual and ask him to re-route something or record something for you. 

FACE TIME

The front of the DR-60D is deceptively simple, but there's a lot going on. The amber screen has an adjustment for contrast, but I didn't find adjusting it away from position 5 or 6 helpful. There's a setting that determines how long the screen remains backlit. Backlighting is essential for low light operation, but the unlit display works well in normal lighting and in full sun. A backlighting intensity control for low light conditions would be helpful and might save some battery power.

A relatively large spin and push data wheel to the lower right of the data screen adjusts easily. The MONITOR SELECT button allows you to select a full mix, 1-2, 3-4, CAMERA IN, or 1, 2, 3 or 4 individually. The MIXER feature allows independent control of each of the four output pans and levels.


TASCAM DR-60D and Canon 5D mk iii
THE WARTS

The DR-66D is not positioned to replace Sound Devices or Zaxcom recorder in the bag, but it appears to be a viable option for many lesser tasks; DSLR cameras with punk audio sections top the list.

No SMPTE, which begs the question of how well the audio will sync when you're trying to double record (using camera audio for a reference to lock the better audio recorded by a  second recorder). Although no SMPTE is not necessarily a deal killer in this day and age, if that's important to you, then stop here.


Amid comments from some that, "they don't need no stinkin' SMPTE", the next question is how long are your shots? Each camera and audio recorder has its own digital clock. If you're shots are pretty short, any slight sync drift may be anywhere from immeasurable to tolerable to fixable. However, the longer the shot, the greater the possible drift. You might get lucky and have two devices that run very close to the same speed. You might not.

There is some zipper noise in the mic input pots circuitry. Not as much as I've heard in others systems and not as much as the first release of the TASCAM HS-P82, which subsequently got fixed. If you are in a noisy enough environment, such as a busy office or on a city street and you adjust slowly enough, you probably won't notice the little clicks, but you might notice the jumps in level.


Some videographers I do sound for don't realize that even when there's only one person talking, I'm still riding gain. If the background is quiet enough (think of inside interviews) and the person speaking begins each sentence with a fresh breath but doesn't project, their first few words are usually a lot louder than the rest of the sentence.


If I can, without bringing up the ambient noise floor, I manually adjust for that so when the project goes to postproduction, the voice is much more level and there's a lot less work to do. As they are inhaling, I rotate the pot back a bit to a lower level. As I hear them run out of that first burst of air, I turn the pot back up a bit to keep their voice at the same level.  That would be noticeable in very quiet situations with the DR-60D.



The gain knobs on the DR-60D are a little small and too smooth to adjust with great confidence, but if you don't operate as I do and don't expect to do a lot of gain riding or mixing, then neither the small knobs or slight zipper noise matter. Or, as I just suggested to a small film/video company out of Washington, D. C. who like working with DSLRs, put something like a Sound Devices MixPreD or 302 mixer in front of the DR-60D. Control the levels more finely and with no zipper noise.

BAD MEMORIES

After inserting a 16GB SanDisk 30MB/sec Ultra SD card and hitting FORMAT, I waited patiently watching the reels spin on the screen. After 10 minutes, I ejected it and the screen informed me that the card was invalid. It would have been nice to know that without waiting ten minutes. I guess I snuck past the gates because the DR-60D was on when I plugged in the SD card. The manual says put the card in, then turn the unit on. When I did that I was informed that the card was invalid.

Next I tried a PNY SDHC Optima 4GB card. When I powered up after putting it in, the screen prompted me and I said go ahead. Within seconds, the system said I was ready to go. I decided to also try the deep formatting that had hung earlier, and it hung again on this card spinning wheels, so I ejected the card, turned the DR-60D off, inserted the card and powered the DR-60D back up. This time it said the card that it has quickly formatted before was invalid.


I powered down the DR-60D again and inserted a third SD card. Bringing the power back up the screen asked if I wanted to format the card. I hit YES and five seconds later the main screen came back up. I took the first SD card, plugged it into the card reader attached to my Mac and deleted all the files from the card. I found a WAV file that I had recorded on the DR-60D but could not access on the reader, so I dragged it to the desktop and it played. I put the newly erased SD card into the DR-60D and powered it up. It asked to initialize the card, I agreed and we were good to go. Observe protocols and you should be OK.


I reached out to TASCAM's Dan Montecalvo who forwarded my message to Tom Duffy. Tom pointed me to the chart below, saying, "We have a continued project for media testing on all our products, so the lists are updated 1-2 times a year depending on the product. The lists do emphasize brands that are available worldwide, potentially passing over high quality brands that might only be easily available in the US. The chart below is valid as of 10/10/13. For updates, please check the TASCAM DR-60D Tested Media List link. 

BATTERY LIFE
The DR-60D requires four AA cells and has no port for external power. My first set of alkaline batteries didn't last quite as long as I expected; a little over two hours with two Phantom Power Schoeps mics running. They went a bit longer on the second set, but I wasn't using two Phantom Powered mics.


TASCAM suggests trying Sanyo Eneloop rechargeables or, for longer shoots, the BP-6AA, which, as its name suggests, holds six AA batteries of your choice and powers the DR-60D via a USB connection. Yes, it's bulkier, but the BP-6AA attaches to the bottom of the DR-60D and has a threaded hole on its lower side for camera stand mounting. (10/11/13 update: Apparently the bolt on the bottom of the BP-6AA is not long enough to thread into the bottom of the DR-60D, so you may have to McGyver a velcro strap-on to your tripod. 

According to Duffy, the BP-6AA extends operation to about 7.5 hours using Phantom Powered mics using Eneloop batteries. I'm guessing that's for 16-bit, two channel recording.

Additional external power supplies include; 


BatteryGeek Personal Power Bank 8000

iSound - Portable Power MAX

Aluratek - APB04F


There's also the PS-P515U AC power adaptor for less remote use. It's a line lump with a USB port. According to Duffy, "For tethered operation, we have the PS-P515U AC to USB Power Adapter, but if you buy say a 15 foot USB to USB mini cable, you have to watch that you don't get one with too thin of gauge, i.e. the voltage might drop too low to operate the DR-60D. I'm in the process of qualifying which long cables are good or not (I'm currently suspicious of the monoprice ones).
The REMOTE jack on the DR-60D accepts the RC-3F tripedal remote control that provides
TASCAM RC-3F Remote Control
a host of extras as shown in the table below.




And finally, although not as potent as the RC-3F, the TASCAM RC-10 hard-wired remote control allows basic control of the DR-60D.


If aiming, focusing and shooting a DSLR is too tame for you, the added audio features of these TASCAM DR-60D accessories should liven things up a bit.


For more information on the TASCAM DR-60D, Click Here.



Technique, Inc. © Copyright 2013 All Rights Reserved


Ty Ford may be reached at www.tyford.com.


DR-60D FEATURES
  • Record to SD/SDHC card (Up to 32GB)
  • Simultaneously record up to 4 tracks
  • Record Mode:MONO, STEREO, DUAL MONO, DUAL STEREO, 4-CHANNEL
  • Recording format:44.1/48/96kHz 16/24bit (WAV/BWF)
  • TASCAM original HDDA microphone preamps
  • Recording levels can be adjusted independently for the 1/L, 2/R and 3-4 inputs
  • Two XLR/TRS inputs support +4dBu line level input and phantom power (24 or 48VDC on XLRF)
  • Plug-in power and high-output mic input supported on inputs 3-4
  • CAMERA OUT connector for output from the DR-60D's mixer (adjustable gain)
  • CAMERA IN connector for sound monitoring from the camera
  • Independent adjustable LINE OUT and HEADPHONE jacks for high-quality sound output
  • 50mW/ch headphone output
  • Tripod mounting socket (bottom) and DSLR mounting screw attachment (top)
  • Handles protect the screen and can be used to attach a shoulder strap
  • Soft-Touch Rubber Keys for silent operation
  • HOLD switch to prevent accidental operation
  • QUICK button feature for easy access to various functions
  • 128x64 pixel LCD with backlight
  • USB 2.0 connection for high-speed transferring of files
  • Mini USB cable included
  • Operates on 4-AA batteries, an AC adapter (sold separately) or USB bus power
  • Can extend battery life with BP-6AA battery pack (sold separately)
  • Dedicated remote control jack for the wired RC-10 remote control or RC-3F footswitch (both sold separately)
  • Internal mixer: PAN and LEVEL controls
  • Low cut filter(40/80/120Hz)
  • Limiter (1/L and 2/R can be selected for link-operation)
  • Delay function for microphone distance adjustment (up to 150ms)
  • M-S decode function
  • Slate tone generator (AUTO/MANUAL)
  • Selectable duration of slate tone from four positions (0.5/1/2/3 sec, when auto-generated)
  • Selectable slate tone generation. 3 positions: OFF/HEAD/HEAD+TAIL, when auto-generated
  • File name format can be set to use a user-defined word or date
  • Dual-recording function allows two files to be recorded simultaneously at different levels
  • Auto-record function can automatically start and stop recording at set level
  • Pre-recording function allows the unit to record a 2 second sound buffer before recording is activated
  • Self-timer function for solo recording
  • New file starts recording automatically without interruption when maximum file size is reached
  • Track incrementing function allows a recording to be split by creating a new file when desired
  • Jump back and play function
  • Equalizer function for playback, and level alignment function to enhance the perceived overall sound pressure
  • Resume function to memorize the playback position before the unit is turned off
  • MARK function up to 99 points per audio track
  • DIVIDE function

Neumann RSM 191 Stereo/Shotgun Mic - Going, Going, Gone!

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Neumann RSM191
The Neumann RSM191 is on the endangered species list because the factory only had fourteen of them left a few weeks ago, when I asked. After they are gone, Neumann says they have no plans to make more. The RSM191 will then fade into history as one kick-ass, remote controllable stereo/shotgun microphone. If you want one, it will probably be a special order. Why bother to post this review? It's just a great sounding microphone system.






Neumann MTX191A


Recording in the field is always a challenge. You hope to come back with the good stuff. The stuff you go out with normally determines how good the stuff is that you bring back. 

In this case, the good stuff is the Neumann RSM 191 stereo/shotgun (10/31/13: Originally $4,550 for mic, MTX191A power supply/remote control, road case and cables, I just saw one for $6000 on Ebay). B&H shows the mic as discontinued, but does list windscreen, foam, cables and mounting accessories. 

Inside the RSM 191 are three capsules mounted within an inch of each other; a small-diaphragm front-directed cardioid capsule with a short interference tube and two small side-directed cardioid capsules. (Note: A stereo/shotgun mic works in either of two modes; stereo mic or shotgun mic, but not both at the same time.) 


RSM191 Multipin Mic Connector
A multi-pin cable connects the mic to the MTX191A power supply/pattern box. The MTX191A is a sophisticated and powerful part of the system. Two rotary switches on the front allow for the selection of -M/S, M/S, -X/Y and X/Y operation. 

When in the M/S modes, the second rotary switch adjusts the Side gain across a range of -9dB to +6dB. When in X/Y modes, the second switch adjusts the width of the pattern for 60 degrees to 170 degrees. 




Neumann MTX191A Power Supply/Remote

Other details include a battery test/battery on switch and a small door which covers the receptacle for a standard 9 VDC battery that powers the system when Phantom Power is not available.

On the back of the box are the multi-pin jack for the mic cable, a 5-pin XLR for the output, a 10dB pad and a switch offering two bass roll-offs. A Y-cable attaches to the 5-pin XLR, splitting the side and front capsules. 

My first gig was to record "Larksong", a madrigal group, in several churches and in a recording studio. Getting six members of a madrigal group together is a logistical feat within itself, so I also looked for other opportunities to find the "boundaries" of the mic.


Neumann RSM191 Side Capsule Response
I had recorded "Larksong" before, using a beyer MC833 stereo mic and a pair of Audio Technica 4050s in Blumlein array. All the early recordings were done in churches. One of the RSM 191 sessions was recorded in one of the same churches we had recorded in before. In all cases, I used GML mic preamps and recorded directly to a Panasonic SV-3900 DAT. 

While the early recordings were always technically very good, the RSM 191 brought something to the table that the others didn't. I would describe this a coloration or a finish. Normally I steer clear of coloration as much as possible, but this was different. Except for minor pan adjustments, the RSM 191 sessions sounded more like a finished production when I played them back over the studio monitors. 

Neumann RSM191 Mid Capsule Response
Our best venue was St. John's Church in Ellicott City, MD. We set up in the empty church with the singers standing on parquet flooring in the chancery, facing out to the pews. Choosing the X/Y pattern, I adjusted the MTX191A to get the right angle based on the distance of 8-10 feet from the group. 

The distance was determined by the tempo of the song and the natural reverberation of the room. I moved back a bit on slower pieces to let more room in and moved up on quicker pieces to keep the room from muddying the phrasing. Decisions were made using an old pair of AKG 240 headphones; designed before they put in a big low-end hump.

In the past, I had pretty much let the singers arrange themselves in an arc, in whatever order they were comfortable with. There was a member change since those sessions and it seemed to throw the balance off. I ended up putting the two most powerful voices -- a soprano and baritone/bass -- at the ends, and moving the others around a bit until the voices started to gel. 

In further experiments, I moved the singers with the most prominent parts of a song to more centered positions. Finally, for "The Little Drummer Boy", I moved the men and their forceful "rum, rum, rum" behind the women, who were singing the lyric. In all cases, the "finished" quality of the recordings was apparent. 

Next was a stop at Flite 3 in Baltimore. As expected, the singers didn't enjoy the experience of singing in an acoustically-damped room. We tried a pair of KM 86s and U 89s in X/Y and coincident omni, but found the RSM 191 to be more open on the top. In a return visit to Flite 3, engineers Louis Mills and Mark Patey and I found the stereo spread of the RSM191 to work extremely well in the studio as a single-source mic for stereo drama. 


Neumann RSM191 Polar Patterns
Set at 170 degrees, the stereo image was extremely smooth and stable. In one test, two of us walked around in the studio while a third in the control room, with closed eyes, listened to the control room monitors and pointed out our positions with a great degree of accuracy. 

In another test, we crumpled up a plastic bag and tossed it across the room. The crinkle made by the bag in flight as it expanded was captured in remarkable detail. After adjusting distances from the mic for individual voice power, we were able to record a very acceptable stereo commercial voice track.

In Studio B, Flite 3 has a Yamaha grand piano. On this particular occasion, I used Great River mic pres and an API lunchbox.With the top open "full stick", I positioned The RSM 191 about three inches inside the piano case, in the middle of the curve and over the longest spoke of the metal frame. I angled the mic slightly to the left, so that the stereo spread would cover both ends of the keyboard. 

The Great Rivers yielded a very natural, full sound. The API preamps were edgier. Next I tried micing a Martin D28S. Placing the RSM 191 about a foot to two feet out and shooting it right into the sound hole resulted in a large natural sounding acoustic guitar sound that filled the stereo spectrum without being so wide as to be fakey or contrived. It should also be noted that, through all of the stereo applications, there were no mono compatibility problems.  

SHOTGUN
For shotgun operation, you just use the front-mounted cardioid capsule. That capsule is related to Neumann's KMR 81 shotgun. It has a 4dB peak at 8kHz that starts at 3.5kHz and returns to zero at 12kHz. 

The RSM 191 has about the same output as a Sennheiser 416. The capsule in a 416 is in the middle of the tube. In the RSM 191, it's at the bottom of the tube. 

If you're close-working the mic, that can make a difference. The 416 self noise was more noticeable partly because it was higher in frequency than that of the RSM 191. The actual level of self noise  of the RSM 191 was slightly less. The RSM 191 was more natural sounding, with not as much low end sensitivity and not the upper midrange peak of the 416. The 416 had a tighter pattern and more reach.  

IN CONCLUSION
I keep coming back to the "finished" sound of the RSM 191. It's not so apparent when listening to a single voice or simple instrument, but when listening to a group of voices or a more complex instrument such as a piano, the resulting sound is very musical. Although that 4dB rise at 8kHz might suggest some undue brightness, I never heard any while using the Great River or GML mic pres. Again, the RSM191 is not long for this world. You may have to make an extra effort to order one of the remaining new ones, but I don't think you'll be disappointed. 

By Neumann's kind permission, here's a 17-page tutorial on Mid/Side recording and the RSM191 written by Neumann's Stephen Peus.

Ty Ford can be reached at http://www.tyford.com.
Technique, Inc. © Copyright 1990 All Rights Reserved

Audacity - Clunky, But Extremely Helpful!

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I’ve been using Pro Tools for over a decade and a half. I make a good part of my living with Pro Tools on any number of Macs. And yes, I admit it. I have not been very kind to Audacity. I tried it and found it clunky. Yesterday, however, changed my opinion of Audacity.
Download Audacity Here - http://audacityteam.org
I received an email from a Google account executive in California. It wasn’t a pitch. He had found me by googling some words about restoring or saving audio. I guess that means that my present SEO strategy is working.

He had recorded audio during several meetings on his cell phone; a great way to keep track of what was said. Unfortunately, as his email explained, on these three meetings something happened and the .wav files could not be opened or even played. He said he would pay me to open them. I told him that I would try, but with no promise of success. 

Import Raw Data
I opened a folder in DropBox and sent him the link. He does not have a DropBox Account. I tried the “collaborate” link first but he said this made DropBox ask him to join. I then sent him the simple “share” feature. That worked. He forwarded the files. I tried opening them with all of my apps; Pro Tools, QT, VLC, Soundtrack Pro and Audition. Even Sound Devices Wave Agent was powerless. Then I remembered that Audacity had an “Import Raw Data” feature. 

I didn’t have Audacity on my computer. I downloaded the latest version (2.1.1) from the web site, http://audacityteam.org. Using the Import Raw Audio feature, I was easily able to bring up the first file in stereo to the time line. One track was solid noise. The other looked like it might have been someone talking. 

Choosing A Different Sample Rate
I used the Audacity menu option to split the stereo track into two mono tracks. Then I clicked on the “M” mute button on the track that looked like noise. 

When I hit play, I head voice, but at the wrong speed. I used the sample rate menu to change the playback to a different sample rate. A short trial and error period later, I found the correct sample rate and the voice track was ready to export. 

I swept my cursor over the track and used Export Selected Audio. A window popped up with half a dozen or so fields to plug values into. I ignored them, but chose the same folder as the original files for a destination. I then added “fixed” to the file name and hit return. Within seconds the file popped up in the folder. When I clicked on the fixed file, the iTunes opened the file and the audio was there. Great!

The second file was even easier. It came up as a stereo file with both tracks playable, but at the wrong speed. I adjusted the sample rate and quickly found success. I appended the name of the file, exported the selected audio and checked the file in iTunes. Another success.

When I used the Import Raw Data on the third file, it came up as a mono file but, again, the wrong speed. A quick sample rate adjustment and it sounded fine. Append name, Export, Check iTunes, Done!

I then moved the three fixed files back to the Dropbox folder and called my new client. He was very grateful. I sent him a request for payment from my PayPal account and even though he did not have a PayPal account he was able to respond and two hours later I received an email notice from PayPal that I had been paid.

I also forwarded a link to my review of the Sony M10 pocket recorder; a very easy to use, compact recorder that will work even better than his smart phone.

So. Thank you Audacity for having a feature that allowed me to go audio fishing in your pond and make an additional billable hour and thank you Google for finding me in a vast universe of people who do things with audio. 

Technique, Inc. © Copyright 2015 All Rights Reserved
More at http://www.tyford.com








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