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Archive for the ‘Uncategorized’ Category

Tips On Getting Killer Drum Sounds by Nick Blagona

Tuesday, June 19th, 2007

Engineers spend more time getting drum sounds than any other instrument. I’ve personally seen situations where days have been spent getting a drum sound. Kits are changed, heads are changed, cymbals are changed, heads are taped up or un-taped, mics are selected and changed, the kit is placed in various parts of the studio, head damping devices are used, mini pads are cut up and placed on heads, and on it goes. The poor drummer keeps hitting his kick, snare, and toms … by the end of this, he or she is back in rehab.

Here’s my approach for a great drum sound. My recommendations for drum mics: Sennheiser MD 421s, Shure 57, and some Neumann 87s. I like using the Neve 1081 console in Studio 1 at Metalworks, so all frequencies mentioned here are from the 1081s. I find that padding down the preamp a low as you can go with the fader up gives me the best result. Having the mic pres all the way down gives me very little leakage from the cymbals to the toms and hi-hat to the snare and very little kit in the kick drum.

The Kick
Mic the kick drum with a Sennheiser 421, throwing a sandbag in the drum helps to dampen out any overtones. The mic should be placed right at the beater. I also use a Yamaha NS10 woofer as my second mic, placed where the front skin used to be. I record this flat since it has the prefect frequency response.
For the 421, give it +3 at 82 Hz for bottom and +4 at 6.8 K for added attack.

The Snare
For the snare drum, use the Shure SM57 at a 45- to 60-degree angle about an inch or two above the head pointing it at the centre of the snare.
+2 at 82 Hz, -2 to -4 at 820 Hz, and +4 at 6.8 K for crispness. If you like the idea of miking under the snare for some rattle and hum, use an AKG 414 in a tight pattern under the stands.

Mic all three toms with the 421s set at about a 45-degree angle to the centre of the tom. I usually add
some 8.2 K.

For the overheads use U 87s. Place the mics about 16″ over the cymbals’ centres and towed out at about 45 degrees. I usually record them flat.

An AKG 451, pointing at the centre.

Nick Blagona has recorded The Bee Gees, Chicago, The Police, The Tea Party, Alexisonfire, Deep Purple, and many others. Please go to www.nickblagona.com for more details.

Everything Is An Amplifier Part I by Bryan Martin

Tuesday, June 19th, 2007

The basic building block in audio is the amplifier. When the word is mentioned most of us have the image of a power amp pop into our heads. They get all the press because they are the largest and most glamorous of the species, but what about the myriad of smaller and forgotten gain stages that occurred before the signal arrives at this last power stage? They are largely anonymous and taken for granted, but determine the quality of recorded sound.

Almost every knob on a piece of audio equipment is controlling a specific amplifier stage. And the farther we get away from the basic understanding of this simple entity, the farther we get away from knowing how to maximize its sonic potential.

The steady advancement of technology has served to obscure their very existence. Amplifiers have become so small and commonplace that they have virtually disappeared from human consciousness. Just look at the iPod nano – that thing is loaded with amplifiers, all crammed onto a little chip and powered by another sliver of technology.

The implementation of each gain stage, individually, and then as a complete amplifier, determines the sound quality of a piece of audio equipment. This fact seems to have been largely lost in the mysteries of time. Most people don’t even realize the devices they use even contain an amplifier. It just works.

In recent years, there has been an onslaught of multi-function units, recording channels, and the like. With the recording business moving from a professional to a consumer market, manufacturers are trying to offer the most features for the price. This looks great on the outside, but there is a large cost on the inside. The quantity of functions within a unit is usually inversely related to its sound quality. Fundamentally, it is difficult to design a good-sounding, multi-function unit, because every gain stage comes with the constraints of its implementation.

The requirements of a gain stage are:
1. Its gain coefficient. (With a coefficient of 10, 1 V input will give 10 V output.)
2. Bandwidth. For audio we generally want to double the range of human hearing (20 Hz to 20 kHz) so that would be 40 kHz to insure good transient response.
3. Input impedance.
4. Output impedance. (Generally we want to have the output impedance of the previous stage low in relation to the stage that it is driving to minimize the losses in the coupling between the two stages.)
5. Maximum output signal before clipping.
6. Maximum input signal before clipping
Pick up the August issue of PS for Part II.

Bryan Martin owns Sonosphere Mastering. Over his 20+ year career he has worked with David Byrne, Rufus Wainwright, Max Roach, Run DMC, and White Zombie. He can be contacted via e-mail at bryan@sonosphere.ca or on the web at www.sonosphere.ca.

Roger’s Rules of Compression by Roger Nichols

Thursday, April 19th, 2007

1: Don’t. I would rather spend the time to ride the solo or vocal to get a cleaner sound with no compression artifacts. I also prefer to manually remove pops and sibilance. You can use the volume automation in a DAW to eliminate vocal pops and sibilance problems by drawing a V-shaped notch at the center of the pop or ess. It does not have to be very wide, and it will work better than any automated de-esser or pop filter.

2: For the most transparent compression, use a ratio between 2:1 and 3:1. This will increase the apparent loudness of your vocal, but will not have that annoying pumping sound of badly adjusted compressor settings.

3: Don’t compress more than 4dB. Watch the gain reduction meter on the compressor. Adjust the input gain or threshold level until the reduction reads between 3 and 4dB, no more.

4: Use multiple compressors connected in a series if you need more than 4dB of compression. Set the attack and release settings differently and you will have more compression without sounding like you’re killing the vocalist.

5: Parallel compression works in some circumstances. You have the dry signal and the compressed signal – mix them together to get the sound you want. Make sure you compensate for any delay in the compressor to avoid phasing.

Roger Nichols is a recoding engineer and producer and has won seven Grammy Awards, the 2001 TEC Award, and received 11 Grammy nominations. He is on the Board of Governors for the Miami Chapter of NARAS and lectures at Berklee School of Music, Musicians Institute, Recording Workshop, Full Sail, Vancouver Film School, and University of Miami. Visit www.rogernichols.com.

5 Tips For Stalking, Managing, & Capturing Rogue Sounds With Traps & Baffles by Russ Berger

Thursday, April 19th, 2007

Employing Sound Traps and Baffles is much like hunting.

1. Know your hunting grounds: Before the hunt, know and understand your acoustical environment. Once you bound a space with walls, a floor, and a ceiling, you’ve committed acoustics. The boundaries of your space define the low frequency modal response and set limitations for the ambient decay time. Wonderful programs and countless texts have been written that clearly describe the process for analyzing, predicting, and managing acoustical boundary conditions.

Once you understand your environment you will better know how rogue sounds behave in the space; you can better identify where problems might lie and devise a trap to capture the problem.

2. Put the traps where the beavers are: Place traps to capture rogue sound much like you’d place traps for beavers. Placing beaver traps on the ceiling will do you little good, just like placing acoustical traps where the sound you want to capture doesn’t exist. Beavers pretty much live their lives along the floor plane. But rogue sounds live in the three dimensional world, so successful hunting can be achieved if the traps are placed in proximity to boundaries and intersections.

3. Be sure your passive trap is big enough to capture your game. Lower frequencies require larger and deeper traps to control and manage long wavelength rogue sounds.

4. Know how many you want to trap: Trapping one beaver vs. an entire colony will require different methods. The effective trap absorption efficiency is proportional to the area of coverage.

5. Conceal the trap: A good looking studio always seems to sound a little better. Integrate your traps into the architecture and along with those rogue sounds you’ll catch new clients.

Bonus Tip #6: go to www.RBDG.com – Russ Berger is Owner of Russ Berger Design Group (RBDG), which is a design and consulting firm that combines expertise in acoustics, architecture, and interiors to create technical environments and buildings for recording studios, broadcast facilities, creative production spaces, and home theaters.

Grounding, Shielding, Hums, Buzzes, & Things That Go Zap! In Your Sound System by Neil A. Muncy

Thursday, April 19th, 2007

Noise susceptibility (or the lack thereof) in audio systems is a function of two principal factors: shielding, and the “pin-1 problem.” The endless conversations concerning this matter inevitably involve earth “grounding,” a subject which has been around for so long (200+ years) that it has devolved into a sea of confusion, misinformation, and mythology, even though it is completely dictated by easily understandable basic physics.

Conventional grounding mythology would have one believe that electronic systems of all kinds must be robustly connected to earth ground in order to properly function – audio signal processing systems in particular. The grounding reality is that airplanes, motor vehicles, laptop computers, blasters, etc. seem to work just fine without connections to earth ground. Nevertheless, A/V systems of all kinds are considered exempt.

According to the conventional mythologists, “noise in audio systems must have something to do with grounding, what else could it be?” The bad news is that the short answer to this question would fill up this entire issue many times over. The good news is that on the Professional Sound website www.professional-sound.com, a long list of reference material will be found. In addition, the June 1995 issue of the Journal of the Audio Engineering Society, entitled “Shields and Grounds,” includes seven papers which directly address this matter. Go to www.aes.org, and look up “Special Publications.” It’s available as freeware to anyone for $15 US, less if you’re an AES Member … it may also be downloadable. It won’t take you long to realize that the conventional mythologitsts just might be wrong!

Neil Muncy has been around since the days when recorded sound was analog mono and vacuum tubes ruled the audio landscape. He has been a consultant in the audio field for many years, and can be contacted by email at: nmuncy@allstream.net.

Audio Phasing: Part II by Al Whale

Monday, February 19th, 2007

Comb filtering, which produces a hollow, diffuse, and thin sound, will occur with one microphone receiving the same sound from two sources. A common example of this is shown in image E. If the microphone had been closer, the difference in the direct path and the reflected path would have been greater, thus the reflected path’s reduced level would have had less effect. Also the reflected source volume would have been less if the floor had been carpeted.

Methods of correction:
1. Keep the vocal audio mix low into the monitor.
2. Handhold or place the microphone closer to the singer.

While the monitor helps the singer, as the monitor’s gain is increased, the resulting vocal will be more muffled. Many professionals use in-ear monitors to eliminate this effect. Although not popular with the performers, using music only on the monitors (no vocal) will also minimize comb filtering. Often, the house audio suffers when trying to improve the monitoring for the performers.

This article was prompted after I attended several concerts in which the music was excellent, however the dialogue was difficult to understand. Most of the production crews knew the script so well that they were unaware of the problems. If you asked the audience, they would probably say that they thoroughly enjoyed the music. If you were more specific and asked them about the script, they probably would be unable to answer. The comb effect of excessive use of stage monitoring would mush the dialogue so that the audience (which doesn’t know the words) would be unable to understand them. If the concerts are trying to tell a story, they basically miss the goal and only provide enjoyable music.

Ideas to reduce comb filtering:
· Reduce the number of paths from the same audio source.
· Fewer microphones.
· Reduce the possibility of reflections.
· Reduce the relative amplitude of the additional paths.
· Increase the difference in path lengths, thus the secondary path will have more attenuation.
· Use absorbent material.
· Use directional qualities of the microphones.

The following sites assisted in this article: Calculations of attenuation over distance www.mcsquared.com/dbframe.htm; calculations of distances www.pagetutor.com/trigcalc/trig.html.

Al Whale is a Broadcast Technologist and Assistant Chief Engineer at CHBC-TV. He has also set up and operated sound systems and taught sound in many church settings. Reach him at awhale@chbc.com.


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