Monday, September 7, 2015

DPA Boom Mics - 4017 and 4018 with MMP-B and MMP-C Preamps


DPA manufactures a wide variety of microphones for live sound, studio recording and location audio. I had seen some recent comments about the DPA 4017 short shotgun and 4018 supercardioid for location audio. I reached out to the US office in Longmont, CO, just north of Denver, and arranged to have them sent in for a look and listen. 

These are part of a modular system of electret mics. Don’t be scared off by the word “electret.” Science and technology have proven that electret mics with internal polarization, can be designed to sound as good as externally polarized “true condenser” mics. I think the term “true condenser” was created by a marketing department to differentiate its own externally polarized mics and cast a bit of shade on the competition. if you're not a true condenser, what are you? A false condenser?

DPA stands for Danish Professional Audio. According to James Capparelle, DPA Marketing Coordinator and Product Specialist for the US office, "In 1992 two employees took the Brüel & Kjær measurement technology and started DPA to make microphones. B&K still exists. DPA still orders the original omnidirectional 4006 and 4007 "German Silver" membrane capsules from B&K." Both the 4006 and 4007 capsules can use any of the modular preamplifiers.

Mikkel Nymand
James Capparelle
"German Silver" actually contains no silver at all. According to DPA tonmeister Mikkel Nymand, "We have a few other capsules in our portfolio using an alloy called “German Silver”, consisting of copper, nickel and zinc. These are used in the legendary recording mics 4006 and 4007. The diaphragms in the 4017 and 4018 mics you have for test are mylar material with a gold layer. This design is used in the majority of our mics.

Nymand says other differences in DPA designs help to differentiate their mics from others. "Higher polarization voltage means higher sensitivity, lower self-noise and also higher distance between diaphragm and backplate, resulting in bigger dispersion of the diaphragm, and that equals higher SPL handling. You can’t achieve all this with a regular external P48V phantom power supply! The pre-polarization voltage is more than you have in your power outlet in the wall, it’s a company secret exactly how much, but I can say we are talking several hundreds volts."

The DPA modular system is comprised of three “preamps.” If you’re new to the field, these preamps do not replace the preamps in mixers or consoles you need to get the signal to line level for mixing. In this case, the preamp converts the very high impedance of the capsule into something a mic preamp wants to deal with. 
The DPA MMP-A, which I was not sent, is a .75” x 5.8” tube containing a transformerless preamp with a switchable 20 dB pad that requires 48 V DC Phantom Power and a power consumption of 2.8 mA. It has an 5 V RMS output voltage. It has a fixed -3dB cut at 8 Hz. 138 dB of dynamic range. The specs also mention that operation is stable up to 90% humidity. 

DPA MMP-B Preamp
The DPA MMP-B is a .75” x 3.9” tube containing a transformerless preamp that requires 48 V DC Phantom Power and a power consumption of 4.5 mA. It has a 1.8 V RMS output voltage. It has a nicely designed low cut and a high boost EQ. The low cut is a first order filter at 120 Hz and a fixed -3 dB filter at 50Hz. The high boost is a +4 dB with a corner at 8 kHz. 128 dB of dynamic range. It has the same humidity figure as the MMP-A.

DPA MMP-C Preamp
The DPA MMP-C is a .75” x 1.6” tube containing a transformerless preamp. At just over an inch and a half long and only 1.4 oz in weight, it’s perfectly suited for long boom extensions where weight quickly multiplies. It requires 48 V DC Phantom Power and draws 2.8 mA. It has a 4 V RMS output voltage and a fixed 3dB cut at 15Hz. 136 dB dynamic range. It has the same humidity figure as the MMP-A. DPA says the MMP-C has a slightly softer or warmer character than the other preamps. I could hear a slight difference. Having read the data sheet that describes the difference as "softer", I'll go with warmer, but subtly so.

Please note that all of these mics require 48 V DC Phantom Power. There are some mixers that only provide 12 or 24 V DC Phantom Power. They will struggle to power these mics and the downside is typically distortion because the mics are starved for power. Some mixers' specs indicate that they provide 48 V DC Phantom Power, but their total amperage capability may not be sufficient to properly power more than one or two mics. A setup consisting of a dozen MMP-B preamps (@ 4.5 mA each), for example, would draw considerably more current than a dozen MMP-C preamps (@ 2.8 mA each). So, Phantom Power is about voltage and current, not just voltage.

Most of us have probably heard stories of how high humidity causes noise to be generated by condenser mics. Sometimes, but not always, this is due to moisture that condenses on the microphone diaphragm, either due to very high humidity, or by moving the mic from a very hot and humid area into a much cooler area where moisture condenses on the diaphragm and creates a path for voltage to leak across the diaphragm. 

Noise can also result when dirt builds up on the contacts between the capsule and the preamp. Summer gets pretty humid in the Mid-Atlantic US. I found that leaving the capsules connected to the preamps instead of unscrewing them and storing them separately pretty much eliminated the noise problems on my Schoeps modular mic system. 

There are also the DPA MMP-ER (rear cable) and MMP-ES (side cable) Modular Active Cables. They are for hanging mics or low-profile table or instrument mics. Both use the same circuitry as the C preamp and feature a split into a 12mm housing and a 3-pin XLR connector. These components attach the capsule to a cable at a length of 3 meters or optionally up to 30 meters. 
IN THE STUDIO
Please check out the videos and subscribe to my YouTube Channel. I made these videos so you could hear what these mics and features sound like and compare them. As you watch and listen to the clips, remember that this is compressed Youtube audio, not full 24-bit, 48 kHz audio. Also, lap top or smart phone speakers aren’t a great way to make judgements about audio. To better understand what's going on, please use good headphones or studio monitors.

My studio is quiet and well damped. That means that any mic sounds as good as it can because of fewer reflections. It does however allow a very good environment to hear what each mic and combination sounds like. With the MMP-B, MMP-C, 4017 short shotgun capsule and 4018 supercardioid capsule in my studio, I was very happy with the sound of all the combinations. With all of my switching and swapping, the resultant combinations still sounded like they all came from the same family. 
I particularly like the design of the EQ controls on the MMP-B  The low cut and high boost EQ are well designed, easy to read and somewhat subtle, but effective. The MMP-C has no controls and neither do the 4017 short shotgun and 4018 supercardioid capsules. 


IN THE GREEN ROOM
For the comparison between the 4018C and CMC641, I chose a more typical location with some room reflection. 



To my ears, the MMP-C preamp and 4018 supercardioid capsule, seen online as the DPA 4018C sounds very similar to the Schoeps CMC641 and at this moment, it’s priced slightly higher than the CMC641. $1,710 for the DPA versus $1,622 for the Schoeps.

Making a supercardioid capsule is easier than making one that has as small a back lobe as possible and that has uniform frequency response as the sound source moves off axis. These changes in frequency response can be heard as slight irregularities or beaminess of certain frequencies. My Schoeps CMC641 is very good. As a person speaking moves off axis, it sounds like someone turned the pot down. Their voice fades away very evenly. The DPA 4018 supercardioid responds similarly. 

I asked Nymand about the axial response of the 4018 supercardioid capsule. "The tighter you want the side response, the more you move towards the figure-of-eight characteristic (including that the rear lobe will be in opposite phase). But exactly how much this rear lobe is present is where the R&D hours are spent. The secrets lie within the physical/acoustic design within the capsule and the rear entrance port. Do remember, any cardioid/supercardioid may not be cardioid/supercardioid at all frequencies. it’s very easy for manufacturers not paying the attention to detail as we do to claim they have a certain type, but it’s not always the case."



DPA 4018 Supercardioid Capsule


Next is the DPA 4017 short shotgun. Like the 4018, the capsules and preamps are light weight, making booming a lot easier. As demonstrated in the video, the 4017 holds it's own among the other shotgun mics and the CMC641.

With interference tube mic, you have to balance the increased distance from the sound source due to the sheer length of the interference tube against what benefit it might provide. The interference tube and capsule of the DPA4017C  account for only 4.5 inches of the total 6.1 inch length. 

While that's short enough to mount on many DSLR cameras, please don't expect great results with any camera-mounted microphones. Shotgun mics do not allow you to crop out the noise the way a zoom lens allows you to zoom in. When using microphones, getting as close as possible to the intended sound source is always your best strategy. 

Interference tube mics are great in controlled sound environments, (Think QUIET ON THE SET!) especially when that controlled environment also limits or covers hard, flat, reflective surfaces. As directional as interference tube mics are for high frequencies, they are not so directional at middle and low frequencies. If, for example, you have ambient noise such as traffic, a busy retail space, a kitchen with cabinet doors and counter tops with a lot of hard flat surfaces like pavement, buildings, walls, ceilings, counters or windows, the better choice may be the supercardioid 4018.

DPA 4017 Shotgun capsule
Another thing I like about these capsules is that there's just enough lift in the high frequencies to add clarity and intelligibility, but not enough to cause that irritating "skritch" heard so frequently with cheaper mics.

All combinations of preamps and capsules combine well and produce very high quality results. In addition they are very light weight. If you do a lot of booming for location audio work, you'll appreciate this immediately and at the end of the day.

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

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Thursday, August 6, 2015

Audacity - Clunky, But Extremely Helpful!

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








Thursday, June 25, 2015

Location Audio Rental Houses Speak Out About Gear And Changing Markets


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

Monday, May 25, 2015

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

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. 

Friday, May 1, 2015

Zoom Q8 - Like A Q4, But More


 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. 

Thursday, April 16, 2015

Rode NTG4 and NTG4+ Shotgun Microphones - New Tools

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


Wednesday, February 4, 2015

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


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.

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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