Tuesday, August 17, 2010

Voiceover - How it was done 10 years ago

From MIX Magazine
To the uninitiated, recording an announcer or "voice-over" artist would seem to be relatively simple compared to other things audio. But for those who have done it, it's a creative/technical task not to be taken lightly. Speech sounds are harmonically and dynamically complicated because of the way vocals are produced-by the chest; lungs; diaphragm; larynx; the oral cavity, including the tongue, hard and soft palates, the teeth and lips, the nasal cavities; and by the dynamic interaction of all those elements through time. Explosions of air bursting from the mouth, the lips and tongue can sound wet, and "sss" sounds can overmodulate a track.
Voice-over artists understand these factors, and the best know how to use them to produce their own voice character. As professionals, they can be counted on to back off for louder passages, to suppress hard plosives like P and T, and to stay a consistent distance and angle from the business end of the mic. Nonetheless, the engineer on the other side of the glass has to have a keen ear, a good technical understanding of how to capture the voice cleanly and a well-developed sense of how to interact with both the Vo artist and the clients.
Mix interviewed five people (see "The Vo Panel" sidebar) who record voice-overs and who edit and mix radio, TV commercials and long-format programs, such as documentaries for The History Channel.
What are the defining characteristics of a good voice-over recording?
Michael Mason: Vos have to be very present, which starts with the acoustics of the room. You want an absolutely dead room, and you want it large enough so that any reflected sound has had a chance to travel out a way, and then return. You want it dead because, while it's always possible to make a sound more "live," as yet there's no "dereverb" box.
James von Buelow: We're looking for people who are good storytellers and who have a voice that's not too sibilant or too dull. Mouth noises and the like can be taken care of during editing, even little clicks in the middle of words. But you've go to start with a Vo artist who has a voice that possesses clarity and a pleasant quality.
James von Buelow
Joe Casalino: It's clean and quiet, with suppressed mouth noises, not overmodulated and not overly compressed.
Wouter van Herwerden: You want as little external interference as possible- just a clean feed from the mic, which lets you treat it however you want to in post. You don't want the talent to get too close and crowd the mic, for then you risk certain distortions. These include popping, mouth noise and too much modulation of the diaphragm of the mic.
Could you describe your announce studio?
Von Buelow: It has a floating floor and walls, so it's very quiet for a New York booth. It's a prefabricated, about 7 by 10 feet with an 8-foot ceiling, so it's fairly small. The air-conditioning piped in there can't be heard when it kicks in. Acoustically, the booth's pretty dead, with fabric-covered walls. It's so dead, in fact, that I'm often surprised when I go in there just how softly the talent is speaking, even when they appear to be loud when heard through the monitors.
Casalino: It's 8 by 12 feet with a 9-foot ceiling. It was custom-built, with 6- inch multilayered walls that float, and with a floating floor and acoustically sealed door. The window facing the control room has triple half-inch panes, so I can work at reasonably high levels. There's also a window in the studio looking uptown toward the Empire State Building.
Van Herwerden: The dimensions are about 10 feet by 25 feet with acoustic panels on the walls and ceilings, as well as carpet on the floor. There are also fixed diffusors to scatter sound and suppress standing waves.
What microphone and preamp combination do you use?
Butler: On the East Coast, a typical mic configuration is a [Neumann] U87 with a hunk of foam over it. I've found this to muffle the sound, so I tend to use as little pop filtering as possible. A nylon screen is about as far as I'll go. like U87s on females and thin-voiced guys. But I prefer a Sennheiser 416, which has a lot more punch to it. It's my primary Vo mic. I hate console mic pre's. I like Focusrite preamps, The Gold Channel from TC Electronic and the Millennia.
Mason: I use, on average, three different mics—a U87, a Sennheiser 416 and a Neumann U89. Basically you want a very quiet mic that allows you to get a lot of noise-free gain out of the mic preamp, which means a gain setting of no more than 45 dB while recording conversational-level speech. With an 87, I don't use the highpass filter, leaving that to the controls on the console. I don't use outboard preamps because the mic preamps in the Euphonix console are awesome. All the EQ and dynamics are just outstanding.
Mike Mason
Von Buelow: We do about 95% of our work with a Neumann 89, leaving it flat. I use a Millennia Media Model HV3, with the gain at about 12 o'clock and no filtering.
Casalino: It's a Neumann U87 in cardioid, to a Focusrite Green preamp and compressor, patched into the 02R console, where it's bused directly into the AudioFile.
Van Herwerden: The mic we use most of the time is the Sennheiser 416. We also have a Neumann 87 here. It's very sensitive and has a wide pickup in the cardioid field, so there's more likelihood of it picking up the room acoustic than if you use a short rifle, like the 416. It's not that sonically I prefer the 416 to the Neumann, but it'll give me a cleaner voice sound. We're trying to a get a specific vocal sound that will cut through whatever else is going on in the track without having to do a lot of extra processing. We're using a TC Electronic Gold Channel [for the mic pre's]. The Euphonix CS2000 consoles we employ also have preamps, but we prefer the TC Golds. They're pleasing sonically, they have more headroom, and they have some builtin features, which are nice.
How do you position the mic relative to the talent?
Butler: The mic's capsule is right on line with the talent's mouth and parallel to his or her face. I'd say from their lips to the actual capsule is about 4 inches. The pop screen is about 1.5 inches or so from the capsule, and then it's about 1.5 inches from the pop screen to their mouth. If "P pops" are a problem, which they can be with a U87, I'll put it into figure-eight or omni. The broader the pattern, the less popping. Sometimes I'll do that if it's relatively tight-miked, quiet Vo where I won't get too much bounce around the room. If the 87 is inverted and comes in from above on a boom, you'll get a slightly brighter pickup than if the mic is used upright.
Mason: On both a U87 and 416, I'll mount the mic on a boom coming in from above. They're generally about 6 inches away. I prefer to have the mic capsule's lower edge on line with their mouth but just above their upper lip. Using the 416, you've got to back up a little more, because it's a shotgun. I use a nylon pop filter to avoid pops. If that doesn't work, I'll angle the mic off to the side to suppress popping, though you have to be careful off-axis, because that does change the sound.
Von Buelow: In long-form work, because they're sitting down, it's maybe 6 to 8 inches from their mouth, and for commercial work, it varies. It will go from very close-3 to 4 inches-to maybe a foot for a very loud speaker. For the latter, I would tend to use the mic's pad to protect the front end of the mic.
Casalino: It's to the talent's side, turned toward the talent, at about a 40-degree angle from straight on to the mouth. They're not talking directly into it, which can help with "pops." Generally, it's about 6 inches away. I often use a nylon pop filter they work right up against.
Tim Butler
Van Herwerden: You try to come in from the side and get it reasonably close without intruding too much on their space. I angle it about 20 degrees from their mouth axis and place it 2 to 4 inches away. I get good presence that way, without excessive danger of pops. Since the acceptance angle of the 416 is probably about 20 to 30 degrees, the artist has to stay "on mic" to maintain consistent results, but that's not a problem with pros.
How do you set up the gain structure of your system when recording a Vo track and when doing the mix?
Butler: Because I want to preserve headroom, I record significantly lower-probably 10 dB lower-on recording a Vo than I do on the final mix. Voice actors tend to become popular because of a unique harmonic structure in their voice, and part of that package seems to be a fair amount of transient information. That stuff can get clipped off or distorted rather easily. So I tend to record relatively low, with peaks 15 dB below 0 VU, which I could never have done with tape because of noise.
Van Herwerden: During the first rehearsals, you'll get an idea of what kind of signal you're dealing with and adjust the headroom accordingly. The dynamic range isn't all that great: You're working with a 2, 4 or 5dB range. We operate here, like anywhere else, at a +8dB peak. We don't record Vos anywhere near that level, because we don't need to with digital systems. As long as we get it down cleanly into the system, we can deal with the odd peak or shout as long as it doesn't exceed that +8 level.
What kind of signal processing do you use during initial recording of a voice-over?
Butler: I tend to always record flat; if I record 60 people a month, I'd bet that 59 would be flat. In the mix, I may add just a sprinkling of EQ, but if you've got your mic placement and choice right, you shouldn't need much. once in a while, I'll go through a highpass if there's some kind of problem. And if the talent is sibilant, you've got to change the mic. I don't EQ that out, and I almost never use a de-esser. But sometimes the talent has such a wicked "s" that I'll apply some in post, though I never use it while recording. [For dynamics,] I might sometimes run just a stitch of limiting, just a smidgen. Down 2 or 3 dB, max. And I set attack and release times by ear, using theconsole compressors.
Mason: Overall, an 87 is a little too dull, and it needs some brightness generally in the 5k range, with some cut at 300 Hz or so. often, I'll use a highpass filter to get rid of some of that subharmonic stuff beginning at 80 Hz, because it just eats up headroom without being heard. Concerning compression, I find it's better to compress a Vo at 2-to-1, both when recording and when mixing, than to compress it once at 4-to-1. I prefer a fairly fast attack and a slower release, because I think a fast release tends to be heard. Regarding de-essing, during the mix I'll use the algorithms set up in my Euphonix con- sole dynamics. or if it's really nasty, I'll throw it into the Pro Tools and use some of the plug-ins to deal with it.
Von Buelow: I plug it into a channel on an 02R, which I use to boost 3 kHz and 10 kHz about +2 dB to brighten it.
Casino: On the 02R console, I'll dial in a very steep highpass filter at 94 or 105 Hz and below, so it just goes away. I don't use a whole lot of compression, 2 or 3 dB at the most. But I don't change it a lot and try to concentrate on consistency of microphone position and sound in the booth. I only EQ at the mix stage.
Wouter van Herwerden
Van Herwerden: In the normal day-to-day recording sessions, we don't apply any processing at all, for the simple reason that if we had to continue on another day, in another room, or with someone else, that the voice will sound the same from session to session. Later, during the mix we'll do processing and EQ, but during initial recording absolutely nothing gets added, other than maybe a little compression or limiting.
During the finished mix, how do you handle EQ and compression/limiting of the Vo track?
Butler: One of the ways that commercial clients judge the mix is how loud their spot is perceived to be compared to others on the air. I have several stages of compression to achieve this. There will be a small amount of console limiting on the Vo input of the mix. Then I might have an 1176 compressor/ limiter on an insert, as well. I'll also have just a little bit of bus compression. And last, I'll patch in a TC Electronic Finalizer Plus, which is a 3-band compressor. With that I can give 4 or 5 dB more level to the DAT. I'm trying to maintain levels that don't exceed -7 or -8 on the meter of a Sony 7030 DAT, while achieving an average mix level of +1 on a VU meter. If you can achieve both of those goals, you've got a pretty hot mix.
Von Buelow: You tend to end up with 3 to 5 dB of boost at 3.5 kHz, or that area, and then a little boost at 8 to 10 kHz. That midrange and high end really seems to do the trick on television.
Casalino: I only EQ at the mix stage. I'll start at 60 Hz and pull that back to avoid "tubbiness." I'll start lifting the top at 5 kHz maybe, on up to 8k. But I'll avoid 2 to 3 kHz; that can be a little nasty.
Van Herwerden: The talent have a particular kind of vocal quality you're trying to maintain in the mix. With Pro Tools, you can save EQ setups, so I can recall them for the artists they were created for. Specifically, I'd be using a TDM plug-in within the virtual mixing page input channels. The same thing goes for compression and de-essing. My processing is "virtual," not hardware, and the nice thing about that is if the mix has to go to another room, as long as it has the plug-ins, too, they can just load the entire session from our backup CD-R and re-create everything I've done in the original session.
CAN WE ALL AGREE?
Common Technique in Voice-Over Recording

Each panel member has his own distinct approach to VO recording, but there are a few fundamentals that all can agree upon.
The main difference between short-form (commercials) and long-form (documentaries, audio books, etc.) is the total amount of compression used. There’s much more in the case of commercials, so as to make them “loudness competitive” with adjacent spots. Long-form readings are usually done with the talent seated, but there are exceptions because of personal taste (and endurance). Commercial spot VO artists usually stand; the body English makes for a better performance. Generally, standing while reading results in better vocal control because the diaphragm is free to move. Headphones were used by everyone, although Tim Butler felt they contributed to the talent being too concerned with the sound of their own voices.
Scripts are almost always placed on a music stand that’s padded and angled to avoid reflections back into the microphone. Usually the goal is to place the active part of the script high enough to avoid the talent looking down at it and getting off mic.
There were several other areas of complete agreement between everyone interviewed.
All record to the hard drive of a digital workstation with a DAT backup.
• All have a console with digitally stored control settings to enable recall of session parameters.
All edit most of their own material, and all perform final mixing on projects.
All are of the opinion that women’s voices are more likely than men’s to offer sibilance problems.
All the participants had several monitoring options, using Tannoys or Genelecs for large monitors (especially useful for revealing low-end thumps, pops, etc.), NS-10s and Auratones as small speaker references, and some kind of 2- or 3-inch television speaker as the final test of what works on the air. Wouter van Herwerden had some illuminating comments about this last piece of equipment, which he calls “Mr. Crappy.”
“My driving force is narration, so I’ll use him to help establish an EQ for the VO that gives me the sound that I want out of a 2-inch speaker,” he says. “Once I’ve set that, I’ll go to the NS-10s or bigger speakers and start doing my first pass on the mix, referencing everything to my narration track. While doing this, I’ll keep referring back to Mr. Crappy because he’s the final arbitrator of all the stages of our work here, that is until 5.1 takes hold much more widely. At the end of the day, a 2-inch speaker is what it all comes down to.”

Wednesday, August 11, 2010

New Microtech Gefell Transistor LDC

New from Microtech Gefell is the M 1030, a large diaphragm studio condenser microphone featuring a cardioid pattern and ultra-low noise performance. Designed for vocals or narration, as well as instrumental recording (solo or in stereo pairs) on guitars, piano, percussion, wind and string instruments, this side-address design exhibits a smooth frequency response with a slight presence boost in the 8k to 14k Hz range.
The M 1030's cardioid polar response exhibits a high degree of rejection for sounds impinging from the rear or sides of the microphone, providing excellent isolation from nearby instruments or noisy sound sources. The M 1030 operates on standard 48VDC phantom voltage and has a green LED on the mic’s front face that indicates the mic is powered up.
The studio condenser microphone M 1030 combines modern large diaphragm capsule technology with the latest in semiconductor circuit topology. The size of the microphone housing is optimized with regard to the expectations of a large diaphragm microphone for studio applications. The micro- phone is specifically designed to meet the needs of professional and semi-professional users who demand the highest performance.

The microphones are ideally suited for universal miking applications in broadcast and sound studios. Applications include vocalists, announcers, dialog pickup and as spot microphones for recording guitars, keyboard, percussion, wind and string instruments.

The pick-up pattern is perpendicular to the direction of the microphone axis (side addressed). The model number and pick-up pattern symbol mark the front of the microphone. The green light- emitting diode (LED) inside the protection grid operates as optical ready indicator.
Under the grille, the large-diameter capsule feeds an electronic impedance converter based on a newly designed circuit topology. This transformerless design reduces the noise floor to a low (7 dBA, DIN EN 60 651) while raising the maximum output capacity, resulting in a 135dB dynamic range. RFI susceptibility is very low. To reduce sensitivity to mechanical impact and structure-borne noise, the mic capsule and electronics are elastically suspended inside the compact metal housing. 
gefell-M-1030-web.jpg M 1030 ships with a wooden case and swivel stand mount. An optional EA 92 elastic suspension (shown in photo) with A 93 adapter is available to further attenuate noise in extreme situations.

Tuesday, August 10, 2010

Shure Model 55- Still Cool After All These Years


prod_img_super55bcr_l.jpg
AFTER MORE THAN 70 YEARS IN PRODUCTION,Shure’s Unidyne 55 series easily takes the prize for the longest running, most successful product in audio history. The story begins in 1937 with Shure engineer Benjamin Bauer looking for a single-capsule approach to creating a unidirectional microphone.

He experimented with capsules having front and rear openings that allowed sound waves to reach the diaphragm. Partially blocking the rear openings created a short phase delay that effectively cancelled the sounds from the rear. Varying the rear port resistance created various directional patterns—cardioid, hypercardioid and supercardioid—and the Unidyne was born. In 1939, Shure announced its model Unidyne 55.

The model 55 was an immediate hit. With its clear sound, high feedback resistance and rugged dynamic capsule the 55 became accepted as a standard for decades to come. Numerous improvements followed, and with its popularity in the early days of rock, the 55 eventually took on the nickname “the Elvis mic,” even being immortalized on a portrait of The King on a U.S. postage stamp in 1994. With retro looks fully in vogue, Shure rechristened the mic as the 55SH Series II in 1996, bearing the model 55S body introduced in 1951, but with a modern SM48-style element. And like the 1939 version, the 55SH Series II was also a hit, with the showing up on stages, music videos, movies, TV shows—just about everywhere.

Super 55ENTER THE SUPER 55

Now, the 55 enters its latest generation as the Super 55, which keeps the cool chrome-plated, die-cast zinc body, but updates the mic with a new supercardioid capsule based on Shure’s successful Beta 58A. The Super 55 has a sensitivity of -53 dBV/Pa, resulting an output that’s approximately 5dB hotter than that of the 55SH II for increased gain-before-feedback. As another plus, the frequency response extends out to 17kHz, providing some extra air and articulation. At the other end, the Super 55’s bass response is smoother and seems more controlled, while the mic retains the 6k to 7 kHz presence boost that helps vocals cut through the mix. One change I appreciated was the omission of the on/off switch, which more often than not, would inadvertently get switched off, leaving vocalists starting a show with no audio. I’m definitely a fan of the “no switch/no problem” approach.

With its blue internal foam windscreen, the Super 55 retails at $311 ($249/street) and the original SH55 II remains in production at $199. And just last week, Shure announced the Special Edition Super 55 ($299 retail) a version with striking black body and red windscreen and offered exclusively through Guitar Center. But whichever model you pick, Shure is definitely making it easy to stay cool this summer.

Sunday, August 8, 2010

HOSA INTRODUCES TRACKLINK MICROPHONE TO USB INTERFACE



Park, CA – July 2010… Hosa Technology announces the new TRACKLINK Microphone to USB Interface. Designed to provide easy, direct access to software-based audio applications, the TRACKLINK is a 10-foot, XLR3F to USB Type A interface cable that enables one to make a single point-to-point connection between the microphone (XLR output) and a USB port on a personal computer. Fully functional with both condenser and dynamic microphones, the TRACKLINK Microphone to USB Interface makes it easier than ever to record speech, vocals, sound effects, and more with today’s most popular software.

t240_USX110_Hi.jpgAs a plug and play device, the TRACKLINK requires no drivers when used with one’s favorite application. The TRACKLINK is compatible with most Windows® and Apple Macintosh® audio software, including Cakewalk’s SONAR LE (for Windows PCs) and GarageBand™ on Macintosh computers. For those interested in podcasting who may not necessarily be musicians, the TRACKLINK is an exceptional tool as it simplifies the process of getting audio into the computer. TheTRACKLINK even provides visual confirmation of a secure connection and recording status via its onboard indicator lights.

The new Hosa TRACKLINK Microphone to USB Interface incorporates switchable 48 V phantom power for condenser microphones and adjustable gain for dynamic microphones. With its 16-bit/48 kHz sampling rate, this easy-to-use interface delivers CD Audio sound quality. It’s the perfect solution for home studio users looking for a convenient means of connecting a microphone to their computer.

System requirements for using the Hosa TRACKLINK Microphone to USB Interface are minimal. The TRACKLINK runs with most Digital Audio Workstation (DAW) or similar audio recording software on Windows PCs with Windows 98 SE, 2000, Me, XP, Vista (32-bit systems), or Windows 7. Macintosh computers must run OSX V10.3.9 or later. The TRACKLINK is a USB 2.0 device.