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Audio Phasing: Part I
by Al Whale
Distance Between Mics
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Image A
he speed of sound is approximately 1,130' per second in air, depending on the actual air temperature. Therefore, if you have a 1,130 Hz tone it will complete one full cycle in one 1'. Now suppose that you have a tone generator feeding a speaker, with two microphones as shown below (image A). Distance affects the phase. If the mics were both the same distance from the speaker, they would be in phase and would add together. The resulting tone would be twice the level (6dB) of either tone (image B). Similarly, if the second mic was 1' further away from the speaker, the two sources would still be in phase and would again add together. If the second mic was only 6" further away from the speaker than the first mic, the two sources would now be out of phase. This would cause the tones to cancel (image C). For the next section return to the previous setting, mic #2 is 1' further away from the source than mic #1. Frequency also affects the phase. At 565 Hz (1,130 hz/2) the tone will now complete a full cycle in 2'. As seen in the following example, the two tones now arrive out of phase and thus cancel (image D). At 1,695 Hz (1,130 Hz x 1.5) the two tones also arrive out of phase and cancel. However at 2,260 Hz (1,130 Hz x 2) the two tones arrive in phase and thus add. This effect, known as comb filtering, can be shown to repeat all the way up the frequency band. The following graph shows the resultant gain verses frequency (image E). Note that when the two signals are equal, if they are exactly in phase they add 6dB, but if they are exactly out of phase, they totally cancel. In an actual situation, the effects would probably not be as pronounced, since the levels from the two mics would seldom be exactly equal. One good example of this situation is when two mics are (mistakenly) placed on each side of a lectern, with the idea that they will pick up the audio regardless of which way the speaker turns. This will result in poor sound quality. As the speaker turns his head, one mic can be closer than the other, thus introducing the comb filtering. Comb filtering will produce a hollow, diffuse, and thin sound. Pick up the February issue of PS for Audio Phasing: Part II.
Image B
Image C
Image D
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.
Image E
Miking The Snare Drum
by Tim Crich
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or the best snare drum sound, using a properly tuned and professional drumkit is paramount. Whether the band is Death Metal From Saskatoon or The Polka Pals ‘n’ Gals, the drums will be the backbone of the recording. Start with a dynamic mic, as it can handle the high transient levels of the snare drum and a solid, stable mic stand. Position the mic off-axis with the rest of the drums to minimize leakage. Aim the mic directly at the point of impact – where the tip of the stick makes contact with the drum. Look down the barrel and line up the placement. Of course, place the mic where the player can’t accidentally whack it. Expecting a drummer not to hit a poorly placed mic is like asking a record producer not to order sushi; sooner or later, it’s going to happen. It’s your fault if the drummer hits the mic with the drumstick, not his. For more crack, maybe place a second mic with a different quality, such as a crisper high end, alongside the first. Keep these two mic capsules as close together as possible because two mics on any one source can create phasing issues. Perhaps add a third (switched out-of-phase) mic underneath the drum aimed up at the snares. Get the best sound using mic choice, placement, and level before reaching for the equalizer. If possible, record the individual snare drum tracks on your digital recorder, and analyze the sound waves. Work on moving the mics around so, when recorded, all the drums are in total phase. Good luck!
Tim Crich is a recording engineer/writer living in Vancouver. His credits include The Rolling Stones, John Lennon, Billy Joel, Bon Jovi, KISS, and lots more. Watch for Tim Crich’s Assistant Engineers Handbook 2nd Edition coming soon. Reach him at tcrich@intergate.ca, www.aehandbook.com.
Challenges In Recording 5.1
by Michael Nunan
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s the great bulk of television production begins to make the transition to HDTV, and casts a wary eye on the notion of 5.1 surround sound – we’re faced with a quality versus quantity dilemma. On one hand, virtually all of the cameras that we’re likely to employ in the acquisition of HD pictures feature excellent digital PCM audio recording. That’s a “check” in the quality column. On the other hand, even the most expensive HD field cameras have a disturbing number of audio connectors on the back … two! Even on cameras that utilize videotape formats that natively feature four (and more) audio channels, the default configuration of the camera usually features only two inputs! That leaves us with a significant quantity problem. All other things being equal, I can readily suggest several microphone techniques, or microphone systems which will do a great job of capturing multi-channel sound. Even after accounting for the requirement that these systems be rugged, simple to use, and field-portable – there’s still a number of options to choose from. Alas, in the world of factual TV production – all is not equal. While the last few years have seen an increasing number of multi-track loca-
tion recorders appear on the market, their use assumes as a prerequisite that doublesystem recording is easily accommodated by the production. Unfortunately, most television production (that isn’t dramatic in nature) still relies on a single-system workflow (where the camera is the only recording device on location, responsible for recording both picture and sound) and therein lays the challenge. Until producers are adequately convinced of the merit of authentic surround production, most will be unwilling to undertake the cost and complexity of changing the way they normally work. The alternative is for manufacturers to start making camera systems which have native multi-channel audio capabilities, which will allow us to honour the tradition of single-system production, while still satisfying our requirement for higher track counts from the field. Either way, the thing we need most is for the audio community to become much more vocal in lobbying for change in the world of TV. Those of us who love sound and understand the crucial role it can play are already convinced … we need to stop preaching to the choir, and start educating everyone around us.
Michael F. Nunan is the Post Sound Supervisor at CTV Television Inc. Reach him at mnunan@ctv.ca.
Rich’s Rights To Recording Electric Guitar
by Richard Chycki
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’ve been fortunate enough to record a number of legendary-status guitar players like Aerosmith’s Joe Perry and Rush’s Alex Lifeson. Watching them work is truly an inspiring and educational opportunity; artists like these have accrued a wealth of real-world experience in manifesting instantly recognizable guitar tones. Being the captor of these tones, I’ll share some tips about recording electric guitars. Right tools for the job: This is a nobrainer but is a common miss. Select gear and tone that works for the song and put your individuality into it. Want to get the right tone? Listen to it. Really. That means pointing the speaker right at your head, not blowing across your knees while you stand in front of a half-stack. Off-axis settings are brittle and don’t sit well in a mix. Right mics: While there are a myriad of possibilities for miking an amp, I’ve had
great success with a few favourite mics. First is the venerable Shure SM57. I’ve tried the Shure Beta 57 and, while it sounds similar, the polar pattern is so tight that finding the sweet spot in front of the speaker can be quite a mission. Other mics I commonly use include the Sennheiser 421, the Sennheiser 409, and the Earthworks SR30. Special mention goes to the Royer 121 ribbon mic. This workhorse mic sounds amazing for almost any electric guitar purpose from country to metal and the specially designed ribbon element won’t fry from the high SPL of close-miking an amp on 11. Right place at the right time: Personally, I prefer to record guitars in more of a dead environment, although I’ve been known to track in extremely live environments (Joe Perry’s tiled bathroom for one) for effect. In all situations I have the amp lifted well off the floor to avoid troublesome reflections, and I don’t use anything hollow
that could resonate (like a roadcase). Right phase: For multi-miking, it’s important that the phase relationship between the mics remain consistent. Liberal testing of phase using the console’s phase flip button is a necessity when blending mics. For mics placed at various distances from an amp, comb filtering can result from the phase shift due to the longer time the sound takes to reach the more distant mic. Fortunately, a small company in the Los Angeles, CA area called Little Labs has a device called an IBP (In-Between Phase). It can shift the phase to any degree from 0 to 180 so it’s a simple task of dialing the mics into exact phase. Happy recording!
Richard Chycki is currently recording a new CD for Rush and has worked with Aerosmith, Mick Jagger, Seal, Pink, and many others in the past. Reach him at info@mixland.ca.
Audio Phasing: Part II
by Al Whale
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omb 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 below. 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 performances 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 performers 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.html; calculations of distances www.pagetutor.com/trigcalc/trig.html.
The reflected source is -3dB of the direct source. The comb filter effect will be present.
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.
SOUND ADVICE
GroundinG, ShieldinG, humS, BuzzeS, & ThinGS ThaT Go zap! in Your Sound SYSTem
by Neil A. Muncy
MICROPHONE DEVICE-1 HYDRO DEVICE-2
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oise 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
H N G Schematic diagram of generic audio signal processing system showing interconections between equipment, building power, and ground. Copyright by Neil A. Muncy all rights reserved.
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 mythologists just might be wrong!
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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.theaters.
Tips
by Russ Berger 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.
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
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.
sound advice
Everything Is An Amplifier Part I
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by Bryan Martin – 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.
he 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
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ngineers spend more time getting drum sounds than any other instrument. I’ve 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..."
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