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Rich’s Rights To Recording Electric Guitar by Richard Chycki

Friday, January 19th, 2007

I’ve been fortunate 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 no-brainer 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 micing 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-micing 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-micing, 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.

Miking The Snare Drum by Tim Crich

Tuesday, December 19th, 2006

For the best snare drum sound, using a properly tuned and professional drum kit 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.

Audio Phasing: Part I by Al Whale

Tuesday, December 19th, 2006

The 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 foot.

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.

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.

Setting Levels For Digital Recording Can Take More Thinking … Than You Might Think by Alec Watson

Thursday, October 19th, 2006

Every once in a while I get a little carried away with trying to think up some technical gem; this time, for a change, I thought it might be nice to take things back to basics – real basics – but not necessarily real simple. Sure, it’s good to know the “hows” (how to do this and that), but more importantly it is better to know the “whys.”

We all know (in digital) that the green lights on the record level meters are good and the red lights are BAD. Did you know however that there are different shades of red? Okay, well there aren’t really, but some nasty red lights on tracks are more acceptable than others…

Of course, we could simply avoid any red lights at all (and the point of this article) by setting levels really low; so why not be extra conservative when setting record levels? The reason we are trying to maximize levels is two-fold; both reasons having to do with noise. There is an inherent noise floor in a preamp that the microphone is plugged into; there is also a noise floor is present in the analog to digital converter that the preamp is plugged into. The A to D conversion process also suffers from a different type of noise, “quantization noise,” that can become an issue when recording digitally at low levels. The noise floor of a preamp or A to D converter can be heard as a hiss; whereas quantization noise is more of a digital artifact that is present when an instrument being recorded is so quiet that it is barely moving the meters. So, in an ideal world we are simply trying to get our recorded signal as loud as possible above the noise floor of the electronic circuits and digital conversion process.

Life isn’t always so simple though is it? It turns out that many mic preamps, especially the tube variety, have a “sweet spot.” There are often three amplification or electronic stages on the way from a microphone to a digital recording medium. The first stage can be found in the microphone itself. A condenser microphone will often have a “pad” where you can attenuate the volume of the incoming signal. If you find your source sounds distorted no matter how low the preamp is set, your source could be distorting your microphone.

The next amplification stage is the mic preamp. This is usually the piece of gear where you get to trust your eyes AND ears. Many mic preamps have little red lights to tell you whether they are distorting or not; sometimes a little red light on the mic-pre can make the track sound more aggressive or fatter; be very careful with this though as there is no “undo” button for a distorted signal. You might find you can add a bit of “grit” later in the digital plug-in world.

Don’t be afraid to trust your ears; no you don’t want lots of little red lights in your digital recordings, but don’t go stopping a take if “the magic is happening” and your ears say it’s okay. It is probably much better to have a flawed recording of a great performance than a technically perfect recording of something completely unremarkable (Hmmm, I am suddenly thinking of some people’s records…).

Alec Watson is a producer engineer that works from his destination studio sitting atop the Georgia Strait – not to be confused with George Straight. Visit him online at www.alecwatson.com.

USB Or Firewire?!? by Alec Watson

Saturday, August 19th, 2006

I have received some e-mails lately asking whether to go with Firewire or USB interfaces, here are some thoughts: I find that Firewire has more stability on older systems. This may, in part, be due to the fact that you have your printer, cordless mouse, keyboard, hub, external hard drive, camera, USB coffee maker … all plugged into the USB ports while the Firewire port sits empty. If I were to choose one system over another, that would be why. A Firewire port will likely be dedicated to your music interface allowing for uninterrupted bandwidth to and from your processor.

USB microphones will be a big hit this year – I don’t think I would run out and buy 12 of these and then plug them into a couple of USB hubs and then record my band – though conceivably you should be able to do this. But if you just want a simple rig for getting sound into the computer, these little techie gifts from the Computers Gods are awesome. I first reviewed the Samson USB mic last year, and although it isn’t likely to replace any of my studio mics anytime soon, value for money, it was excellent. Apparently I wasn’t the only one who thought this, as there are all sorts of manufacturers who will be peddling USB mics this year. If you just need to get an acoustic instrument or vocal into the computer, you will be hard pressed to beat the ease of use and price of these.

The greatest innovation, possibly of all time, I have saved for last (insert echo-y deep announcer voice) Roland’s V-series MIDI accordion! You think I am joking don’t you? I am, a bit. Seriously however, this thing rules! (www.roland.com/products/en/FR-7/). For all the keyboard players and programmers out there who have to twiddle, for hours, with all sorts of knobs to get their keyboards to play horns or strings, this unassuming little machine could change programming as we know it. The biggest reason a keyboard can’t recreate a realistic horn or sax sound is the fact that the keyboard is a percussion instrument. Much like a drum, the envelope of the sound attacks hard and then decays; an accordion however IS a wind instrument, one with keys. You can play horn arrangements that sound fantastic on this little machine – accordion geeks of the world rejoice! For all those years that you have been picked on and laughed at, the lead guitarist and keyboard player are now going to be secretly envious! You’re still not likely to get any action after the show, but oh yes, the lead guitarist will secretly be plotting against you, while he is getting some action!

Alec Watson is a producer/engineer that works from his destination studio on Vancouver Island. His parents couldn’t afford to get him a piano when he was a child; they got him an accordion; the rest, including his career, is history.

Don’t Shout Before You Speak by Jim Yakabuski

Saturday, August 19th, 2006

The lights go down. The dry ice creeps over the front of the stage. The crowd is frantic as a low rumble builds and builds until the ceiling tiles are falling out of the roof and people are ready to run from the building. Just as you think you can’t take it any more, the rumble builds to a deafening, throbbing crescendo and then is abruptly cut off by blinding light and a band on stage that sounds as if it is playing through a transistor radio.

Sound familiar? Hey, it has happened to me. The darned intro tape can kill you every time. And why is it that bands always want to use something that has 4 Hz in it to open the show? Go figure.

The problem that causes this discrepancy in level is usually SPL reference. During the afternoon when you soundchecked the band in an empty room the volume of the intro tape seemed quite substantial. But after an opening act and the roar of the audience as the house lights go off, you find yourself pushing the level of that intro tape higher and higher, leaving the band to come out sounding less than impressive.

You need to establish the maximum level that the intro “rumble” DAT can go before it upstages your band’s first song power level, and not be freaked out if it doesn’t sound loud enough as it’s rolling. It’s better to start out with the intro sounding a bit low and the band sounding a little loud than the other way around. I refuse to let all the frequency bands through when this type of tape is handed to me. If the bottom end of the band doesn’t usually live in the 30-40 Hz region for most of the show, then I’m going to high-pass my DAT intro tape to at least 40 or 50 Hz. You want the audience to remember the first note the band plays with an overwhelmed feeling, so let it be good and powerful. Don’t let a silly tape that was produced and mixed at Skywalker Ranch give your sub-bass speakers too much of a workout before the real deal comes on stage. Save the best for last and lighten up on intro overload.

This article is taken from Jim Yakabuski’s book entitled Professional Sound Reinforcement Techniques. The book is published by MixBooks, an imprint of artistpro.com.


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