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Sound Advice

Location CD Recording: Miking & Mixing Techniques – Part 2 by Earl McCluskie

June 19th, 2009

When recording, for example, a contemporary choir backed by piano, bass, and drums in a natural hall environment, the backup will be picked up by the choir mics, and will sound boomy and unfocused if not carefully controlled. Fortunately, most halls have a built-in solution: reception spaces and other rooms with doors opening into the hall. The drums and bass can often be located here. If the bass is acoustic, some sound will enter the hall, but considerably less than if the instrument was in the hall. Communication for the instrumentalists, and conductor if necessary, can be achieved with headphone fold-back and video monitors.

Typically in such a space, one would mic the piano by putting mics inside the piano, closing the lid, and perhaps even encasing the piano in packing blankets. This produces a distinctive sound, but does not take advantage of the natural piano sound in the room. Instead, position the piano with as much distance from the choir as is possible, and balance its pickup with the leakage into choir mics. During sound check, experiment with your post-production plans (EQ, compression, etc), as getting the right balance has to happen now.

Miking technique alone will not give you the kind of control you need to produce a full “studio” sound. Once you have captured a full choir sound, you will find that the room characteristics will define the choir sound as being in a natural acoustic space, and this will not balance well with the drums and bass.

An old trick used for “fattening” up the sound of a guitar involves double tracking the sound source, applying compression and expansion to one track, and then mixing the two together. The choir pickup will have unneeded bottom end from the omni mics, so in the processed track, much of this can be rolled off. You can also narrow the stereo width of the processed version of the choir, using the unprocessed original to create a sense of depth and width. You can delay this track as well, although care must be taken with possible phase cancellations, leading to an unnatural choir sound. Also, any processing done to the choir sound will impact the piano sound, and vice versa.

The sound of the hall has now become an integral part of the choir sound, and can be blended with the backup ensemble tracks, sweetened with appropriate reverb.

Earl McCluskie is a producer/engineer and Owner of Chestnut Hall Music, a music production company based in the Waterloo region of Ontario. The company specializes in location CD recording, both live and session. Recent projects have included Vancouver-based composer Timothy Corlis with the DaCapo Chamber Singers, Montreal’s violinist/composer Helmut Lipsky and soprano Suzie LeBlanc, and the Guelph Symphony Orchestra.

In-Ear Monitors: Tips & Tricks by Keith Gordon

June 19th, 2009

Picking up where we left off last issue, there is another important psycho-acoustic effect to remember when working with IEMs. When IEMs are used in both ears, there is a “stereophonic” effect (stereo mix not required, just both ears used) known as binaural summation which yields a perceived 6dB increase in volume without any change in the level of either ear’s input volume. This means the left and right IEM are each outputting 90dB SPL, but when both IEMs are inserted, our brain sums them together and we hear an equivalent 96dB SPL, yet without the hearing damage associated with those extra 6dB SPL.

You can try this experiment yourself by turning on your MP3 player, setting a level, and putting in one earbud. When you add the second, you will notice a substantial jump in level. The practical upside of this is to always use both IEMs and not just one like so many performers I see on TV. It makes me cringe to think of how much louder they are blasting their IEMs to get the same volume. Even worse, if they are using floor wedges in an attempt to get the “best of both worlds,” they will be blasting their open ear too.

I learned another trick for wireless IEM users from Mike Prowda, monitor engineer for Nine Inch Nails and David Bowie. Prowda likes to use a compressor and limiter before the wireless transmission stage. Wireless systems have fairly narrow bandwidths in which each channel operates, so to best exploit what is available, it is important to aggressively compress and limit the signal before the wireless stage to keep it from overloading while at the same time not leaving any dynamic range unused – and therefore wasted.

This is similar to the approach radio stations take with their transmissions, using multi-band compressors and limiters to deal with different frequencies separately so that the overall energy level is controlled while not making the music sound overly squashed. At the time, Prowda was using Aphex Dominators, though there are similar units that can also handle this multi-band compression approach. For those of you without access to advanced tools such as this, try experimenting with whatever compression you do have before the wireless stage to see if you can find improvement. If you have any questions, please drop me a line.

Keith Gordon is a veteran audio engineer who helped develop a DSP-based hardware/software IEM system (inearsounddesign.com) in conjunction with Westone Laboratories. He can be reached at keithgordonca@gmail.com.

Location CD Recording: Miking Techniques by Earl McCluskie

June 19th, 2009

Location recording of non-live events has its pros and cons. On the pro side are natural acoustics, a unique sonic character that can give the recording a distinctive sound, prestige from the name of the facility, and sometimes lower rental costs. On the con side are external noises, little or no control over the early reflections and reverberation, difficulty isolating musical elements, and less than ideal control room monitoring conditions.

If the cons can be overcome, or ways to successfully deal with them found, good recordings can be made. These recordings do not have to be limited to just classical recordings, which typically are recorded in natural acoustics, or “live performance” environments. As an example, a 40-voice choir backed by piano, bass, and drums singing contemporary jazz-influenced music can be successfully recorded in a natural ambience.

The choir sound that one would naturally pick up in a church or concert hall using mic techniques associated with classical choral recording would have a significant amount of ambience and depth, suitable for that style of music, but not with the sort of warmth and presence that is associated with a contemporary “pop” sound. A good hall acoustic has a life and character that only the best studios can emulate, and so it is often worth finding a way to capture this sound.

Close-miking the choir would defeat the advantage of the hall by suppressing its natural attractive acoustic. Even the best cardioid pattern mics have significant colourations resulting from their uneven off-axis response, and these often do not compliment the room acoustics. A carefully-placed array of three or four omni mics over the choir can produce a natural-sounding pickup.

Make sure that choir members are as equally distant from the mics as possible, with the lower voices singing directly on-axis to the mic, and the higher voices projecting slightly below the 0-degree axis of the mics. The distance between the mic array and the choir will also depend on the ratio of direct to early reflection balance that sounds best. Use two additional omni mics placed behind the choir to pick up the warmth of the choir, and give additional boost to lower male voices, which tend to be more omni-directional.

Earl McCluskie is a producer/engineer and Owner of Chestnut Hall Music, a music production company based in the Waterloo region of Ontario. The company specializes in location CD recording, both live and session. Recent projects have included Vancouver-based composer Timothy Corlis with the DaCapo Chamber Singers and the Guelph Symphony Orchestra.

Mastering Pet Peeves by George Graves

April 19th, 2009

1. The majority of vocalists don’t know how to use a mic, and what is worse is that a lot of engineers don’t know how to teach them to use a mic. As you’d imagine, the outcome sounds pretty bad. That’s the most irritating scenario for me. I find sibilance is very harsh, and it can be easily tamed in the early stages using the proper mic and technique for a certain vocalist. Sometimes you need a de-esser in conjunction with a compressor/limiter, and of course, using the best-sounding A/D converter possible is key. I know a lot of these lower-price rigs don’t give you that capability, but as a sound engineer, you need to strive to get the best you can possibly afford – and in some situations, less is more. Adding a lot of EQ after the fact rarely helps the recording.

2. Another thing with vocalists is pops. I find that because the vocal is usually one of the loudest instruments in a mix, you have all sorts of odd sounds that are normally filtered through the air when someone speaks without a mic that get captured during recording. It doesn’t help that the mic is often placed at the worst position – the jaw – so all of these bad sounds are directed directly into the diaphragm.

There are other things I could mention, but for time’s sake, I’d say those are two dreaded instances for mastering engineers.

George Graves is a Mastering Engineer at Toronto’s Lacquer Channel Mastering carrying over 40 years of industry experience.

Accurately Measuring Distortion by Wayne Jones

April 19th, 2009

There are two main areas of distortion measurement. The most common is total harmonic distortion plus noise – that’s what most audio distortion analyzers will characterize. It’s a good measurement of performance, but the area where it falls apart is measuring distortions at high frequencies, especially in band-limited devices (and so many components are band-limited). All digital systems are limited to half of the sampling frequency, so they’ll automatically be limited with the anti-aliasing filters to 20 kHz or so. That means it makes no sense to measure THD above around a third of that frequency, so the THD readings you do above 7 kHz don’t mean anything. They’ll give you a number, and it’ll probably look really good because the filter is rolling off the harmonics, so it’s really just measuring noise – not distortion.
That doesn’t mean there’s no distortion at high frequencies. Your ear will certainly tell you that there’s indeed distortion. How do you characterize that? In a band-limited medium, such as analog tape in addition to new digital systems, intermodulation distortion measurements are a way to characterize higher frequencies. One type of IMD measurement is the so-called “Twin Tone,” where you take two high-frequency signals (15 and 16 kHz, or 18 and 19 kHz) and you look for the difference frequency component at 1 k. That will give you a true, accurate, and usable characterization of high-frequency distortion – right up to the band edge limit. If your system cuts off at 22 kHz, you could measure 21 and 22 kHz and get a true characterization of distortion at high frequencies.
This method was discovered in the ’40s while measuring optical film soundtracks on which all film sound was done. It was an optical track on the edge of the film, before it started being striped with a magnetic coating for magnetic soundtracks. The problem there was that the upper frequency limit of an optical soundtrack was 7 kHz, so all of those early films from the ’40s, ’50s, and even the ’60s stopped at 7 kHz. People in the film industry and SMPTE recognized that total harmonic distortion measurements above 1 or 2 kHz were meaningless, so they came up with the SMPTE Intermoduldation Distortion Method which used a 7 kHz and 60 Hz signal and measured the inter-modulation products developed from that. It ended up being a realistic, accurate, and useful characterization of the distortion of an optical film system.
My advice is that if you’re looking at a band-limited device, as most things are now, be careful measuring THD above a certain frequency, and use other techniques to get a better characterization of what’s really happening.

Wayne Jones has almost 40 years of experience in the pro audio and audio test and measurement fields. He’s served on various standards and has been a consultant to companies like Intel, Microsoft, and SigmaTel in recent years.

Following The Golden Rule by Andy Hermant

February 19th, 2009

Andy Hermant believes a mix of analog and digital recording techniques is the ideal way to work. PS asked which stages should be performed in which domain.

I call it my Golden Rule. First, you’ve got to have a great song. Nothing starts without a great song. Then you need to have a great performance. If you don’t have the song or the performance, you’re still nowhere.

But after that, grab the best microphone you can find, the best preamp, and the best converters. Then comes the all-important mic technique. Once you have that pure source material, you can bring it into the digital domain and go crazy and manipulate the recording to get the most out of it.

If you don’t have the goods going in, you won’t have the goods going out.

Andy Hermant founded the Manta Sound Company, Canada’s first digital multi-track studio, Duke Street Records, and 1:2:1 Recording. He was manager of Post Production at the CBC for 13 years and has served on the boards of CARAS, CIRPA, FACTOR, and Roy Thomson and Massey Halls.


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