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Thursday, April 26, 2012

Radial Increases Focus On Export And Hires New Sales Manager

Radial Engineering Ltd. is pleased to announce that Steve McKay will be transitioning from Sales Manager to fill a newly created position as Export Sales Manager. Roc Bubel joins the Radial team as new sales manager.

According to Radial company president Peter Janis: “Steve has been managing our domestic sales team for several years while also investing what time he had available on developing our export distribution channels. With the global economy shifting to emerging markets like China, India, Turkey, south-east Asia, the Middle East and Brazil, we feel that the time has come to invest more energy into these important markets. As Steve has already been working with our international distributors, the move here is natural. We feel that this will finally afford the time and focus to grow our Radial, Tonebone and Primacoustic brands in these important emerging markets.”

To fill Steve’s shoes, Radial has appointed Roc Bubel as Sales Manager. Roc previously held the position of National Sales Manager for Fender in Canada. Janis continues: “I have known Roc for nearly 30 years. We worked together back in the 1980s when I was with Fender so when Roc decided to get back into the industry it was an opportunity that we could not pass up. During his 22 years with Fender, Roc gained a tremendous reputation as being a top notch manager and very well respected by his peers. We are very excited about these changes and look forward to a very exciting year.”

Radial Engineering

In The Studio: Going For That “Vintage” Sound

This article is provided by Home Studio Corner.

 
Do you know the reason we rave over those “vintage” recordings?

Imperfection.

Forty years ago, they didn’t use tubes, transformers, and tape by choice. Nope, it was all that was available to them at the time.

The “sound” of so many of our favorite recordings from that era came from running the audio through gear that couldn’t faithfully and accurately reproduce the signal.

Each piece of gear added something to the sound — warmth, low end, smoothness, punchy-ness, even noise.

In short, it was near impossible to get a clean, accurate recording. The gear added to the sound.

Our recording heros aren’t heroes because of the tools they used. They’re heroes because of how they USED the tools.

They took this “imperfect gear,” with all its pros and cons, and they made great music with it.

I really believe we have a huge advantage today.

We have access to affordable equipment that will give us nice, clean, accurate recordings. AND we can also get the “vintage” gear sound if we want, too.

The key difference between now and then? We have a CHOICE.

Do those “vintage” albums sound great? Sure.

Does that mean we need to make our recordings today sound just like those? Heck no.

Here’s what I think.

I vote that we constantly push the envelope, and use the technology available to us RIGHT NOW to make great-sounding music, even if it doesn’t sound like the hallowed vintage recordings we hold so dear.

Don’t get stuck in the past. You don’t need a $3,000 tube compressor to get great recordings. Heck, you might even find that the big fancy comp doesn’t even give you the sound you really wanted anyway.

Use whatcha got. That’s what the old-school folks did, and it worked well for them.

For example, I’m still amazed at the different tones I can get with a simple, run-of-the-mill compressor plug-in.

You can too.

Joe Gilder is a Nashville-based engineer, musician, and producer who also provides training and advice at the Home Studio Corner.Note that Joe also offers highly effective training courses, including Understanding Compression and Understanding EQ.

Peavey MediaMatrix NION Training Seminar Scheduled For May

Peavey Commercial Audio will host a two-day MediaMatrix NION Certification training course in New Orleans during May, where AV designers, consultants, contractors and end users can learn best practices for designing, deploying and implementing MediaMatrix audio distribution, processing and control systems.

The MediaMatrix NION Certification training course instructs AV professionals on the fundamentals of MediaMatrix, the most flexible and scalable audio networking system on the market, as well as how to design and program projects in nWare; how to set up a NION processor; how to create end-user GUIs for nTouch 60 and nTouch 180 touch screens and PC kiosks; and how to integrate the XControl into a MediaMatrix installation.

The MediaMatrix NION Certification training seminar will be held in New Orleans on May 10-11, from 8:30 a.m. to 5 p.m. each day.

Each successful student will receive a completion certificate that can be submitted to InfoComm for receipt of 7.5 renewal credit hours for InfoComm CTS, CTS-I and CTS-D. For full class descriptions and registration, go here.

Peavey has educated thousands of AV system designers, integrators and end users from around the world since MediaMatrix revolutionized the professional audio industry in 1993.

MediaMatrix now offers the content of its renowned seminars as online courses here.

Completion of the online Peavey MediaMatrix Basic or Advanced course earns two hours of credit toward InfoComm’s CTS and CTS-D Certification Renewal.

Learn how MediaMatrix can be used to create virtually any audio system. For more information, go here.

MediaMatrix
Peavey

Avid Releases New Version Of Sibelius First

Avid has announced the release of Sibelius First, combining the Sibelius Student and Sibelius First notation products into a single easy-to-use, powerful, and fun music composition tool.

The new Sibelius First takes advantage of the features introduced by Sibelius 7, including a task-oriented UI specifically designed to guide the user through the creative process, along with quality sound, 64-bit technology and support for the latest operating systems.

It also offers new video export functionality, online publishing, and direct sharing to YouTube, SoundCloud, and Facebook, and more.

Sibelius First’s standout features include:

  • Sibelius First offers a refined, task-oriented user interface that makes it possible to get great results without prior experience.

  • Choose from scanning, transcribing, or MIDI import to quickly get ideas into Sibelius First—now with full MusicXML interchange for transferring music into and out of other applications.

  • Includes a premium sound library so compositions can be played back in detail on high-quality sampled instruments, and produce better quality exports.

  • Unlock the full power of a 64-bit computer, and directly address more than 4GB of RAM.

  • No engraving or notation expertise is required to get great results. Magnetic layout and dynamic parts make it easy to produce and tailor scores to suit musicians of all kinds.

  • Sibelius First scores can be added to the score library on the Avid Scorch app (sold separately).

  • Users can easily export compositions as videos and publish their work to social media websites (YouTube, SoundCloud, Facebook) along with traditional print, email, or audio exports.

Sibelius First software is available now for $119.99 USD and multi-seat 5 packs are available for $299.99. Current Sibelius First and Sibelius Student customers may take advantage of upgrade options for $39.99.

Sibelius Online
Avid

Wednesday, April 25, 2012

Differences, Cause & Effect And Consequences Of Polarity And Phase

Polarity and Phase - these terms are often used as if they mean the same thing. They are not.

POLARITY: In electricity this is a simple reversal of the plus and minus voltage. It doesn’t matter whether it is DC or AC voltage. For DC, Turn a battery around in a flashlight and you have inverted or, more commonly stated, reversed the polarity of the voltage going to the light bulb. For AC, interchange the two wires at the input terminals of a loudspeaker and you have reversed the polarity of the signal coming from that loudspeaker.

PHASE: In electricity this refers only to AC signals and there MUST be two signals. The signals MUST be of the same frequency and phase refers to their relationship in time. If both signals arrive at the same point at the same time they are in phase. If they arrive at different times they are out of phase. The only question is how much are they out of phase, or stated another way, what is the phase shift between them?

The important point to note in these definitions is that you can reverse the polarity of one signal and you can measure this change. You need two signals to measure a phase shift.

For convenience, the word “speaker” will be used in place of the more correct term “loudspeaker” in the rest of this article.

A picture is worth 1,000 words… but a few words of explanation can help.

The following figures show the differences and some consequences of polarity and phase. Figures 1 through 12 show graphs of sine wave signals. Actually it is a sine wave from one signal source split two ways. Except for figure 1, one of the splits is “processed” by reversing its polarity and/or by delaying it (phase shifting it) as described. To put this in the real world, imagine two speaker systems side-by-side, each reproducing one of the signal splits. (More precisely, the graphs show what you would see on an oscilloscope looking at the output of a mixing console with each split going to a separate input after one of the splits has been “processed”.)

The vertical scale in the graphs is in arbitrary units of -2 to +2 with lines at each 0.5 interval. If you like, consider this as -2 to +2 volts. Because phase shifts are measured in degrees, the horizontal scale in the graphs is labeled in degrees with a vertical line at each 90-degree point. One full cycle or period of a sine wave is 360 degrees.

Assume that the signals shown are 1 kHz sine waves, in which case each vertical line represents 1/4 millisecond of time. Sound travels in air about 3.4 inches (85 mm) in 1/4 millisecond so each vertical line also represents this distance. Note that in the graphs the signals all start 1/4 millisecond or more from the left so you can clearly see when each signal starts. (The importance of this will be seen in figure 9.) There is no signal along the flat line from -90 to 0 degrees.

Signals In Polarity, In Phase
Figure 1: This shows 3 periods or 3 cycles of two simple sine waves. Both are +/-1 volt high at their peaks = total of 2 volts. One is shown in blue the other in red.

Figure 1: Sine Waves in Fig. 1 Added.

Figure 2: This is what happens when the two are combined (= added together). This is exactly what would happen on a line exactly between the two side-by-side speakers. The two signal beings being in phase and in polarity add up so the peaks are now at the +/- 2 volt lines = 4 volts or twice the original signals. Acoustically this is an increase of 6 dB = 20 x log(1+1).

Figure 2: Two Sine Waves - Same Polarity & Phase.

Signals Out of Polarity
Figure 3: This is like figure 1 but the second sine wave, shown in red, has been reversed in polarity. As you can see the + and - voltage points are exactly opposite from the first sine wave, shown in blue. This would be accomplished by reversing the +/- input connection on the speaker reproducing the red sine wave.

Figure 3: Two Sine Waves - Red = Polarity Reversed.

Figure 4: This is what happens when the two are combined. Each point of the two signals being in phase, but opposite polarity, adds up to zero. Acoustically this is an infinite decrease of output. Because you can’t take the log of 0 assume the difference is actually 0.0.01 volts (the dots = 58 more zeros). 20 x log of this number is -1200 dB. That should be pretty quiet. You can’t easily hear this with two speakers because of having two ears. But using a very carefully positioned microphone to measure this in a place with no sound reflections, you would find almost no signal.

Figure 4: Sine Waves in Fig. 3 Added.

Signals Ot of Phase

Figure 5: The second sine wave, shown in red, starts 1/4 millisecond later (90 degrees later) than the first one, shown in blue. Put another way, the second signal has been delayed by 1/4 millisecond.

Figure 5: Two Sine Waves - Red = Phase Shifted 90 Degrees.

Figure 6: This is what happens when the two are combined and it’s pretty interesting. First notice that the peaks are almost at the +/-1.5 volt lines. The value is actually +/-1.414 volts. This is a 3 dB increase. This would be like listening to two speakers but the one reproducing the red sine wave is 3.4 inches (85 mm) further away from you than the other. The first thing you hear is only from the speaker reproducing the blue sine wave. The black line starts when the sound from the second speaker is heard and this line is the combined signal of both speakers.

Figure 6: Sine Waves in Fig 5 Added

Suppose the speaker reproducing the red signal were only 2.25 inches (57 mm) further away. The signals would be shifted by only 60 degrees. The increase for the combined signal would be about 4.5 dB. So the amount of phase shift is important.

The second thing to notice is what happens at 1/4 millisecond or 90 degrees after the blue signal starts when the second signal “kicks” into the picture represented by the line turning black. There is a distinct change in the waveform.

The third thing to notice is that the entire waveform after the “glitch” is shifted in time compared to figure 7 about 45 degrees = average of 0 and 90 degrees.

Signals Out Of Phase And Polarity

Figure 7: The second sine wave, shown in red, is a combination of the sine wave in figures 3 and 5. The signal not only has its polarity reversed but it is shifted in phase by 90 degrees compared to the first signal, shown in blue. In this case the speaker reproducing the red sine wave has its +/- input connection reversed in polarity and is 3.4 inches (85 mm) further away from you than the one reproducing the blue sine wave.

Figure 7: Two Sine Waves - Red = Phase Shifted 90 Degrees & Polarity Reversed.

Figure 8: This is what happens when the two signals are combined. The picture is similar to figure 6 with two important differences. First the “glitch” at the point where the second signal starts is different. This is the point where the line turns black. Second is that the entire waveform is shifted by 45 degrees again but this time to the left of the original signal.

Figure 8: Sine Waves in Fig. 7 Added.

The “Glitches”
The glitches in figures 6 and 8 give an indication of what happens during the onset of a signal. While the so-called steady state portion of the combined signal (shown by the black portion of the lines) looks the same except for the amplitude change, these glitches will affect the transient attack of sounds. This is not to say that either will sound horrible, but a phase shift between otherwise identical replicas of a sound WILL make a difference in the sound of the initial transient attacks, depending on the frequency and amount of phase shift.

This is exactly the kind of phenomena that can occur in the crossover region of a speaker. This is because the distance from each driver to the listener is usually different and the crossover itself shifts the phase of the signal between the drivers. Speaker designers are often faced with a choice between something like what you see in figures 6 and 8. Neither is “correct” so a designer can only choose the one that “listens” better. Just looking at these two, I would bet the waveform in figure 8 might sound better and the choice would be to reverse the polarity of one of the drivers. These crossover “glitches” occur only over a small range of frequencies where both drivers reproduce the sound. It is well accepted by designers that this kind of “improvement” is sonically more significant than the fact that frequencies above and below the crossover point may be out of polarity.

Signal Phase Shifted 180 Degrees
This is where many get into trouble in thinking that phase and polarity are the same thing, meaning that it is often assumed that a 180 degree phase shift and reversing the polarity are the same.

Figure 9: In this figure each sine wave lasts for only 2-1/2 cycles. The second sine wave, shown in red, is shifted in phase 180 degrees from the first shown in blue. This is what would happen if the speaker reproducing the red sine wave were about 6.8 inches (170 mm) further away from you than the one reproducing the blue sine wave. You can see that between the 180 and 900 degrees the signals LOOK like they are simply out of polarity but they are NOT. It is VERY important to note that if you could not see the beginning or the end of these signals you could not tell whether they were out of polarity or 180 degrees out of phase. Too often this is what causes confusion between a polarity reverse and a 180 degree phase shift.

Figure 9: Two Sine Waves - Red = Phase Shifted 180 Degrees & Polarity Reversed.

Figure 10: This is the result of combing the two signals. Unlike figure 4 where the signals are simply out of polarity, and completely cancel, there are clearly two positive halves of a sine wave visible before and after the two signals cancel along the black line between 180 and 900 degrees. The first is from the blue sine wave in figure 9 that occurs before the start of the red sine wave. The second is from the red sine wave in figure 9 that continues after the blue sine wave stopped.

Figure 10: Sine Waves in Fig. 9 Added.

Signal Phase Shifted 180 Degrees And Reversed In Polarity

Figure 11: This is the same as figure 9 but the polarity of the red signal is reversed from figure 9.

Figure 11: Two Sine Waves - Red = Phase Shifted 180 Degrees & Polarity Reversed.

Figure 12: This is the two signals in figure 11 combined. Between the 180 and 900 degrees, the signals add much like in figure 2. However there are significant differences in the overall 90 to 1080 degree signal. The first 1/2 sine wave of this signal is only from the blue sine wave from figure 11. The last 1/2 sine wave is only from the red sine wave in figure 11. You can clearly see that both of these 1/2 sine waves are only 1 volt at the peaks. This is a clear difference from figure 2 where all the peaks reach 2 volts.

Figure 12: Sine Waves in Fig. 11 Added.

The reason is that the two signals in figure 11, even though identical, are offset by 180 degrees. They add together only between 180 and 900 degrees when both are being heard. More importantly, during this time period DIFFERENT parts of the same signal have added together. For example you can see that between 180 and 360 degrees it is the second 1/2 of the blue signal’s first complete sine wave that adds to the first 1/2 of the red signal’s first complete sine wave.

Real Audio Signals
Sine waves are easy to look at to dramatically show the difference between polarity and phase. Armed with this knowledge you can look at figures 13 through 18 that show something like a real audio signal where the effects of polarity and phase are more difficult to see.

The signal shown in these figures was a generated by a mathematical algorithm that produces something close to a pink noise signal. Pink noise contains all frequencies with an equal amount of energy in each octave band. Real audio signals don’t look much different than pink noise (but one would hope they sound better!). The scales on these graphs are arbitrary. You can look at the vertical scales as +/-3 volts if you like. However, because of the way the signal was generated, there was no way to define absolute time or degrees along the horizontal scales. Suffice it to say that the phase-shifted signal used in these figures was shifted by one data point out of the 240 data points that make up the signal lines.

There is one important thing to understand about phase shift. The amount of time one signal is delayed from another will have different effects at different frequencies. Assume there is a 1 millisecond time difference between two identical signals. At 500 Hz the result will be as shown in figure 10 because at 500 Hz the 1 millisecond time difference is a phase shift of 180 degrees. The signals are offset by 1/2 a cycle. At 1 kHz the signals will be offset by 1 complete cycle. In other words you would hear one cycle from the first signal then both combine then you’d hear the one cycle from the second signal after the first stopped. This is similar to what is shown in figure 12 (which shows only 1/2 cycle) but is not the result of the same conditions that were used to make figure 12. At 250 Hz the effect would be as shown in figure 6 because a 1 millisecond time difference corresponds to a 90 degree phase shift at 250 Hz or an offset of 1/4 cycle. At lower frequencies the phase shift would be even less and the signals would tend to add as in figure 2, approaching but never quite reaching the 6 dB increase shown in that figure.

Contrary to phase, polarity affects all frequencies the same way. It makes the positive portions negative and the negative portions positive. Put another way, it simply flips the signal over the same way at all frequencies. With these things in mind, examine figures 12 through 18

Effects of Polarity and Phase On “Real” Audio Signals
Figure 13: This shows a pink noise signal generated as noted above.

Figure 13.

Figure 14: This shows both the original signal in blue and what happens when an identical but phase shifted signal is added to it, as shown in red. The red signal is similar to the combined signal shown in figure 6. Note the increases in signal level and the changes in the waveform (many glitches). However you can also see the combined signal follows the original fairly closely.

Figure 14.

Figure 15: This shows both the original signal in blue and what happens when the phase shifted signal is also reversed in polarity and combined with it, as shown in red. In this case there are huge differences between the original and combined signal.

Figure 15.

Figure 16: To better understand what is going on, this figure shows an averaged or integrated version of the pink noise signal in figure 13. This is basically what would you would see if you graphed the readings from a typical SPL meter for the signal in figure 13.

Figure 16.

Figure 17: This shows the averaged signal from figure 16, in blue, and the averaged combined signal from figure 14, in red. Note that there are primarily level differences (mostly increases). Otherwise the two lines look very similar.

Figure 17.

Figure 18: This really shows what is going on in figure 15. The blue line is the averaged signal from figure 16. The red line is the averaged signal from figure 15. The red line shows that the out of polarity and phase-shifted signal approaches a straight line. Because you are looking at a broad frequency range, you are seeing a severe cancellation of the lower frequencies due to the polarity reversal. However, unlike the low frequencies, the upper frequencies do not completely cancel due to the phase shift. The red line contains primarily high frequency energy. In the blue signal the higher frequencies are the small “bumps”. These can be clearly seen in the red signal and most of them correspond to those in the blue signal.

Figure 18.

Figure 18 is a prime example of what you would hear if you stand exactly between two speakers playing the same signal (i.e. mono) with one speaker out of polarity. The bass will disappear. But, there will always be a difference in distance between you and the speakers due to the spacing of your two ears and probably a slight overall difference in distance between you and each speaker. A difference in distance means a difference in the time arrival and thus there will be phase shifts between the sound from the two speakers. The amount of shift will vary with frequency. Because of the shorter wavelengths at high frequencies, the phase shifts allow most of the highs to be heard. They may be out of polarity but the effect is like what is shown in figure 8. Also, in a room you would also hear sound reflections from the floor, walls, and ceiling. You would only hear something like the red line in figure 18 outdoors away from any reflective surfaces or in an anechoic chamber.

Figure 19.

The small distance between your ears and any small difference in distance from you to each speaker do not cause appreciable phase shifts at low frequencies. This is because of the considerably larger wavelengths. The difference in your distance from each speaker might be only 1 inch (25 mm). However, the wavelength of even a 1 kHz sound is roughly 1 foot (300 mm) and at 100 Hz roughly 10 feet (3 m). At the lower frequencies the polarity difference predominates because the phase shifts due to the difference in your distance from the speakers is very small compared to the wavelengths of the low frequencies. Thus the lower frequency signals, being nearly in phase but out of polarity, will cancel like in figure 4. The lower the frequency the less the phase shift between the two speakers and the greater the cancellation.

A Polarity / Phase Field Trip!!
(As with all physical exercise, check with your doctor first, who might not recommend you do this for some reason.)

Find two railroad tracks, lie across them, and wait.

Two trains, one on each track, come along. Both are right side up and both hit you at exactly the same time. The trains are in polarity and in phase.

The same thing happens again and both trains hit you at exactly the same time. However, this time one train is upside down.

That is a polarity reversal.

The third time both trains are right side up but one hits you first and the other hits you shortly after the first. That is a phase shift.

The last time the second train is upside down and hits you later than the first. That is both a polarity reversal and a phase shift.

Summary
So there you have it. Although this has only touched on a few areas concerning phase and polarity issues, it is hoped you better understand the difference between the two and a few of the effects of each. Remember that the audio frequency range covers wavelengths of over 30 feet (10 meters) at the lowest frequencies to less than an inch (under 25 mm) at the highest frequencies.

While a reversal of polarity will affect all frequencies identically, a difference in time arrival between two otherwise identical signals will have very different effects on the phase between them. The amount of phase shift will be different at different frequencies and this will depend on how much time difference there is between the arrival of the two signals.

Ashly Audio Gear Showcased At The Detroit Institute Of Art

One of the primary event spaces at the Detroit Institute of Art (DIA), the second-largest municipally-owned museum in the nation, is benefitting from a sound system upgrade that centers on an Ashly Audio PEMA integrated processor/amplifier and FR-16 network fader remote control.

“The walls, floors, and ceiling of the event space are solid stone,” observes Dennis Phillips, owner of Phillips Pro Audio, the company that designed and installed the new audio system. “It’s an art institute, after all. But solid stone makes the space tremendously reverberant.”

The previous sound system solution was to bring in temporary speakers-on-sticks for events that required music and/or microphone support, but because the room is 40 feet by 50 feet, the direct sound was often overwhelmed by the indirect sound. Moreover, requests for sound system support began to arrive with ever-greater frequency, making the set-up/tear-down cycle an ever-greater burden.

“They had passed a tipping point and wanted something simple, permanent, and intelligible,” summarizes Phillips.

The combination of the Ashly Audio PEMA integrated processor/amplifier and the Ashly Audio FR-16 remote fader met that criteria. PEMA packs full, open-architecture Protea processing and either four or, in this case, eight channels of reliable amplification into just two rack spaces.

The FR-16 (and its smaller sibling, the FR-8) provide user control via sixteen (or eight) integrator-assignable faders and buttons, plus a “master” (also customizable).

The FR-16 and FR-8 receive power and communicate via Ethernet, which simplifies integration. Phillips designed a custom wall recess in which to house the DIA’s FR-16, along with an iPod dock and a CD/DVD player.

Because of the Ashly FR-16’s implementation, user control of the new system is simple. Each fader controls the volume of an input source, and each associated button toggles between mute and unmute.

Inputs include two wireless microphone systems, the museum’s central paging system, and various music/video sources.

“It’s exactly what the DIA was hoping for,” says Phillips. “They have all the control they need, but no more. The faders are simple and intuitive. On our end, the system was painless to set up and worked as expected from the start.” Atlas FAT-series recessed ceiling speakers now distribute sound across the space, which dramatically improves the direct-to-indirect ratio, and thus intelligibility, throughout the event space.

Phillips was alerted to the integrated Ashly solution through the work of Mike Somerville, MI sales rep with McFadden Sales (Westerville, OH). “Mike is great,” Phillips says. “He keeps us up to date on the latest technology, and he provides with excellent recommendations when we request them. With the Ashly PEMA and FR-16, Mike came through again.”

Because the PEMA and FR-16 are competitively priced, Phillips was able to provide this powerful solution while staying well within budget.

Ashly Audio

In The Studio: Tips For Mixing Rap Vocals, Part 3 - Compression

This article is provided by the Pro Audio Files.

 
Check out Mixing Rap Vocals Part 1 and Part 2 before reading this article.

Compression is a difficult subject because there is a lot you can do with it.

So let’s look at the main reasons to grab a compressor before getting into some of the more intricate uses.

Quick Macro-Dynamic Control

Macro dynamics refer to words and phrases. These are the clear dynamics you can hear as “this part is louder, that part is softer.” The most transparent way to get things sounding even is to actually automate the vocals manually.

But sometimes time doesn’t allow for this approach. So if you aren’t automating, a light ratio, slow attack, slow release, just catching the louder moments with the threshold is a good way to even things out.

Micro-Dynamic Control

What volume automation might not catch is the very quick dynamic changes – loose spikes at the fronts of words. These spikes aren’t heard so much as “volume” but more as an overall quality to the vocal.

The issue with these spikes is two fold – first, they eat away at your headroom pretty quickly– second, they will trigger any compressors you are trying to use for purposes besides micro-dynamic control.

It can be useful to dedicate a compression stage toward pulling back these vocal spikes. Generally a fast attack and release, and a light ratio does the job. The light ratio is to retain the articulation of the word and minimize frequency skewing.

The key is to set the threshold low enough to catch as much of the peak as possible while effecting the body of the signal as little as possible. I try to avoid using limiters for this purpose. I like the Empirical Labs Distressor for this (especially for controlling peaks while tracking), as well as digital style compressors such as the Logic or Pro Tools stock compressors or the Waves C1.

The attack setting is very important – it’s usually between a number of nano-seconds and two or three milliseconds in the digital world, and on the faster side of things for the analog world (totally varies unit to unit).

Getting A Vocal To Stay Audible Through A Mix

The power of compression is that you can make something louder while not actually raising the peak volume of the signal. This becomes extremely useful for making something cut through a dense mix or to come forward. This is probably where the majority of compression work for rap vocals come in.

Rap is generally an in-your-face, visceral style of music. The kick is physical, the snare is physical, subtlety isn’t really the overall goal. And the vocals are paramount. I’ve mixed a number of rap records where the vocals are lower in the mix, but never have I thought it was a good idea.

Generally I want the vocals to be equally as strong as the drums or stronger, and I want them as “forward” as possible. Compression is usually a part of that equation.

Optical Compressors

The smoothest way to get those vocals forward is through optical compression. The rounding quality of the attack and the unique shape of the knee in an optical compressor makes them ideal for vocal work.

Examples of optical compressors would be the CL1B, the LA2A, LA3A, your stock Logic compressor has an optical mode, RComp has an optical mode – and don’t quote me on this but RVox has an “optical” sound to it, as does the “smooth” setting on the UBK-1.

One of the advantages to opticals is that they tend to have easy access. Many have just one knob to control the degree of compression.

Attack And Release Time

Of course you’re not limited to simply optical compressors or fixed time settings.

Many other compressors work very well for rap vocals – in fact, any decent compressor can yield great results if set properly.

The key is setting the attack and release times appropriately. People will suggest milliseconds or time ratings for the best attack and release for vocals but the issue is that 300 ms on one compressor might give you the same results at 75ms on another.

So, I’d rather advise your compression technique based on the expected results. Your goal is to pull up as much of the “sustain” of the voice – the weight of it – while minimally affecting the articulation. Taking notes? – It’s about to get heavy.

When dealing with the articulation of the words, you’re primarily gauging your attack time. There’s generally a substantial range of attack speeds that work for vocals. What you don’t want to do is set the attack too short, or the shaping of consonants will be blurred. Nor do you want to set the attack too long, because you’ll allow the consonants to poke through too hard. So you want to find a middle ground.

A good way to experiment is to temporarily pull the threshold down a little farther than you normally would and find your attack setting that way, as the effect of the attack time will be more exposed with the lower threshold.

With the release time, my goal is to pull up as much of the body of the voice as possible. So I’m going to set the release on the faster side. I don’t want the voice to distort or become unnatural sounding, but I want as much body as possible before I get to that point.

In terms of both the compressor ratio, and the release time, I tend to be a little more aggressive with rap vocals than “softer” music. The “integrity” of the vocal sound is not really as important as the prominence of it. For a more relaxed, natural sound I might do a medium release and 3:1 or 4:1 ratio.

For a rap vocal I’m going for a pretty quick release, and I’m doing 4:1 up to 8:1. Rap isn’t really supposed to be “pretty,” so I don’t worry if the compression becomes a bit audible.

Thicker Vocals

Another great use for compression on vocals is to make the vocal sound thicker – particularly in rap. Rap is frequently recorded in home studios, even by big name artists. And home studios rarely produce the thick, full vocal sound that one can get at a professional facility.

So being able to thicken and give weight to a vocal is an extremely important skill. In order to do it right though, you need a little more than compression. You need an EQ to make sure the vocal is as even and smooth as possible. Then you need some “saturation.” Saturation is just a nice name for friendly distortion. Saturation moves and enhances the harmonics of the vocal. Over saturating will sound like crud, but just the right amount gives the impression of a richer sound.

So the formula is – get the vocals sounding clean, saturate to get the vocal sounding richer, and compress that signal. You have to tread carefully though as over EQ’ing, over saturating, or over compressing will make your vocals sound horrible. Unless you do all of that in parallel!

Parallel Compression

Parallel compression basically means making a copy of the signal, compressing the snot out of it, and then blending that parallel signal back in with the original signal.

The advantage here is that you can get really liberal with the effects, and just blend it in to where it doesn’t sound unnatural. This is great is you are trying to fill-out frequencies that weren’t really there in the original recording, because you can really saturate and compress the parallel signal and generate some very consistent dense harmonics.

Then just blend that in until just before it starts to sound too effected. One of the reasons I really like the UBK-1 for vocals is because it gives you a saturation stage followed by a compression stage and the ability to blend both in parallel.

Distortion Free Equalization

Lastly, compression can be used to tame frequencies without the artifacts from EQ. Often with vocals you’ll have moments where the vocalist changes their tone.

A common example is vocalists will often become more midrangy as they project because they tighten their neck and push more air through their nose. If you simply notch out some midrange – that might work, but it might also take away some of the energy of the overall performance, or some of the frequency information you need to make the vocal stand out.

A good alternative is to use a compressor with either an adjustable side chain, or an external side chain input. The side chain signal is what the compressor reacts to – so if you can EQ the side chain to target problematic frequency areas, the compressor will “intelligently” react to those tones and pull them down.

Conclusion

Compression is a powerful tool that many people struggle to fully understand, so try to get your hands on one and start experimenting. As always I’ll keep an eye on the comments in case there is anything that needs clearing up. I also encourage you to share your own compression tips here!

Matthew Weiss records, mixes, and masters music in the Philadelphia, New York, and Boston areas. Find out more about him here.

Be sure to visit the Pro Audio Files for more great recording content. To comment or ask questions about this article go here.

Tuesday, April 24, 2012

Making It Bigger: Understanding Effects & Adding Them To Your Mix

On rare days off from running sound, I get the opportunity to go to several local clubs to check out the work of other mix engineers.

This is really cool because I pick up tips that can help my own work, in addition to experiencing a wide range of gear.

One thing in particular that I pay close attention to is effects, and I find that in most every case, some type of effect is applied on almost every instrument and vocal.

Which leads me to the point that particularly in club sound, the proper use of effects can leave the audience with the impression that they’ve heard a full-blown concert.

Reverb, simply defined, is the result of many decaying reflections of sound in a room.

Reverb units simulate reflections of different types of rooms, and most units are outfitted with standard pre-sets for long or short reverb times, reverse gates, gated reverb, delays, flange, chorus and so on. These settings can usually be adjusted and saved for specific

More specifically, some adjustable settings include:

Reverb type. Common choices include room, hall, plate, and vocal.

Reverb delay. The time between the initial signal, such as a vocal, and the start of the actual reverb effect.

Reverb time. How long the reverb can be heard after the input signal stops.

Gate time. The amount of time before the response is cut off. A gated reverb, typically applied for drums, cuts off the sound of the reverb more abruptly than the usual more gradual decay.

Reverse gate. Usually a gated reverb, but with a twist. Instead of simulating reflections that become quieter after the initial signal, a reverse gate simulates reflections where the sound gets louder over time, and then abruptly cuts off. Again, usually applied to drums.

In contrast to reverb, delay is simply a number of very distinct added repetitions to sound. These repetitions will usually be the same (or very close to the same) volume of the original signal, although many delay units now allow setting both volume and the decay of the repeating signal.

On most units, the timing of these repetitions can be set from milliseconds to several seconds. The user can also define “feedback” - the number of times it will repeat.

Delay can be used on almost any instrument, but most commonly on vocals. A short delay time used on a vocal can “fatten it up” nicely, whereas a long delay can create an echo effect.

Both delay and reverb run in stereo can also be used to create even bigger sounds and/or “ping-pong” effects. Some units also provide the ability to set different reverb or delay times for each side of a stereo system.

I always recommend running any effects in stereo in order to gain the biggest advantage from them.

Running effects in stereo versus mono is like comparing night versus day – the difference is quite substantial.

Now let’s switch to the application side. When adding effects to instruments and vocals, keep in mind that in most situations, it all should be done in a complimentary manner. (Except in very rare cases.)

For example, if mixing a slow ballad with a long reverb applied on vocals, then chances are that a short-gated reverb on the drums will sound awkward.

I’ve also heard different delay times and reverb settings on lead and background vocals. Again, for the most part this is not a good idea. If all vocals are intended to be part of a unified pallet, which is almost always the case, then they should be presented as unified.

In other words, adding a random setting to each vocal - without any specific artistic purpose - will result in vocals that are a jumbled-up mess.

I once found this out the hard way. Experimenting with a new, original song, I set up different delay times for the background vocals.

Everything sounded great - until all four vocalists started singing a certain passage together. Then it became three badly matched background voices topped by a lead voice that also didn’t remotely fit.

It suddenly sounded like a different song had started! (By the way, if you ever find yourself in this position, slap the side of your rack and act like the gear is causing the problem, not you.)

A very common question with effects: how much is enough? I think we’ve all come across mixes that were compromised or even ruined by the overuse of effects. (On the other hand, I’ve also heard some so-so bands that were saved by a good mix and heavy doses of effects!) 

Particularly in a smaller environment like a club, certain elements of a mix tend to stand out: guitar lead, keyboard lead, thundering kick drum, and so on, while most of the time, the effects are not intended to stand out.

The audience should notice when there is no reverb or delay on the vocals, but not perceptibly notice when these effects are applied. Enhance with effects, don’t over-compensate.

Of course, there are exceptions to every rule. A good example is the gated reverb on the snare drum in a lot of Phil Collins songs.

Another can be found on “Bohemian Rhapsody” by Queen, which has a heavy flange on the vocals. And that same band’s “We Are The Champions” offers a reverse gate application that lends a great sound to the drums.

So particularly if mixing the same band regularly, make it your business to learn the songs inside and out.

This includes going to band rehearsals, where you can calmly evaluate where effects might add something special, and also talk with the band about it as well. (This might seem obvious, but I’ve talked to so many mix engineers over the years that never bother to attend rehearsals.)

The routing of effects is another key piece to the puzzle. If at all possible, they should be sent from an aux send, a pre-fade effects send, or a monitor send (if not running monitors from front of house). The reason is control.

The band I regularly work with plays in a different club every night. This means that every show I readjust my gain structure, fader levels and EQ in light of the specific club parameters and specified volume limits.

By running pre-fader, I’m able to adjust the input to the effects unit and leave it the rest of the night without worrying about clipping when I push up a lead.

I also return all my effects back to a channel on the house console - again for more control. This provides the ability to EQ the effects and control the output easily with the faders, and I can also turn off the effects when the band isn’t playing.

Running it back to a channel also allows me to easily add an effect in the middle of a song, and to more easily and effectively add it to the mix.

For example, let’s say there’s an echo that needs to happen in the middle of a particular song. I set it up and have it ready, then when the right time comes, slide up the appropriate fader and then back down, and it’s on to the next effect.

Four effects units reside in my own club rack - one for delay on vocals, one for reverb on vocals, one for reverb on drums, and one that I use to achieve certain other things.

This unit is programmed ahead of time with different delay times, echoes, flanges and so on that I want for that night’s show.

Again at risk of stepping into the land of the obvious, I always name these effects according to the song they’re to be applied to. This way I don’t have to try to try to remember different setting names in the heat of the mix.

The extra unit also means I can leave my main reverb and delay settings untouched while adding the extra effects where and when I want them.

Many of the more common and affordable effects units now on the market come with the ability to pre-program several different effects on one channel, eliminating the need to purchase several units.

What effects units does Tim use? Top to bottom: TC Electronic M-One on vocals; Alesis MidiVerb 4 for a quad chorus on sax and some special effects on guitar, drums and vocals; Yamaha SPX 990 on drums; and an “old reliable” Roland SDE2500 for certain vocal reverb as well as delay on certain songs. (click to enlarge)

Keep in mind, though, that this can present some limitations.

To bring up the volume of the reverb, but not the delay, requires doing so in the effects unit settings rather than just pushing up a console fader.

And there is access to only the effect “sounds” and features/options provided by the given manufacturer of the unit. Still, the majority of units offer plenty of effects options.

One key difference is sound, which is where you come in, because as we all know, sound “quality” is a subjective evaluation.

Experiment by listening to various units to find the one (or ones) best for your operational needs and sonic situation.

If possible, rent a unit for a week or two to check out before buying, and never be afraid to ask the advice of others.

Tim Andras is a mix engineer with well over two decades of experience working with sound.

Friday, April 20, 2012

Metric Halo’s ULN-8 Centers Solar Powered Mobile Recording Rig

The video for Karmic Juggernaut’s new song, “Oo Wah Hoo,” documents the band’s inspiring and utterly fresh take on the recording process.

Piloting a Subaru Outback topped with a plane of solar panels and freighted with a bank of batteries, the band toured its favorite locations in the great state of New Jersey to record each instrument in the glory of the outdoors.

In the video, what you see – acoustic guitar on the beach, drums in the forest, wailing solo on the mountainside, and more – is what you actually hear.

In order to keep the power consumption low and the fidelity high, the project relied on the one-rack space, eight-channel, Metric Halo ULN-8 preamplifier/converter/interface for all studio functionality, save for microphones and a computer.

JR Skola, a fellow engineer and filmmaker who now heads Brooklyn-based Dawn of Man Productions, produced the video and provided the Metric Halo ULN-8 to make the recording happen. But it was the band’s drummer, Kevin Grossman, who first hit upon the idea of creating a solar-powered mobile recording rig.

“This mode of recording combines all of the things that I love to do: hanging out with friends, being outside, and making music,” Grossman explained.

And to allay any suspicion that the video is first and foremost a green technology PR stunt, it’s worth noting that it was only after highlighting the glories of recording outside that he said, almost as an afterthought, “and while I could have run the whole thing from my car engine with a power inverter, I thought, ‘why not let the sun do it.’ Solar may not have the lowest carbon footprint yet, but it’s worthwhile to promote alternative technologies.”

The solar panel technology consisted of a “standard off-the-grid setup” of three 15-watt solar panels, a battery bank with protection against over- and under-charging, and a power inverter to generate the AC power required for the gear. In general, the solar was enough to record acoustic instruments indefinitely, but the band’s vintage tube amps required both the batteries and the panels and thus enforced a finite session recording time.

Although you wouldn’t know it by watching the video, “lighting, weather, and our mere 100 amp-hours of battery life made recording the amplified instruments a challenge,” according to Grossman. A MacBook Pro running Logic Pro was power-light, as was the efficient Metric Halo ULN-8. Although the band had a large collection of mics at its disposal, workhorse Shure SM57s and AKG 414s captured most of the tracks.

The locations featured in the video are all places where the members of the band and their friends hang out. The crew trudged through knee-deep water to beat high tide on their way to Sandy Hook Gateway where they recorded bass, but only via the Metric Halo’s DI. “We recorded the bass on top of one of the old munitions bunkers at the abandoned Fort Hancock, but it was mostly for the inspiration and the shot,” Grossman admitted. “However, we did have birds flying all around us, and I pointed that fact out to everybody. I marveled, ‘this is THE recording studio!’”

Next, they drove to Monmouth Battlefield. Grossman and McCaffrey played a vintage Yamaha console organ outside the site’s old farmhouse. The next day the band decamped to Belmar Beach for the acoustic guitar recording that opens the video.Later that same morning, the crew returned to Pat’s 30 Acres to record McCaffrey’s electric guitar. A month later, the team reconvened at the Delaware Water Gap near the Appalachian Trail to record guitarist Randy Preston’s blistering solo.

Similar sessions at Allaire State Park for drums and Asbury Park Casino (a cavernous abandoned space) for vocals rounded out the track. The sound of the track makes Karmic Juggernaut extremely happy. “We’ve recorded songs and soundtracks in multi-million dollar studios,” he continued, “but with just the Metric Halo ULN-8 and the acoustic beauty of un-walled space, ‘Oo Wah Hoo’ outshines them all.”

In the future, expect Karmic Juggernaut to pack enough solar panels to cover an entire band. “We’re working toward a full live performance using only power from the panels,” Grossman beamed.

Metric Halo

Thursday, April 19, 2012

Meyer Sound Raises The Bar At Jay-Z’s 40/40 Club

New York’s recently renovated 40/40 Club, owned by hip-hop superstar Jay-Z and partner Juan Perez, has recently undergone a multimillion-dollar renovation which included a complete sound system upgrade. Multiple Meyer Sound loudspeaker systems have been installed throughout the 12,000-square-foot nightclub from the main floor and the VIP lounges to the bathrooms and hallway.

Fort Lee, N.J.-based JD Audio & Video Design (JDAV Design) handled design and installation of an A/V package that also includes four 3-by-3 video wall arrays—each measuring 165 inches—and over thirty 55-inch LED flat-panel HD monitors.

According to JDAV Design President Gabriel Karlis, 40/40 Club’s proprietors had previously installed a sound system that featured equipment from various manufacturers. “I kept telling them that Meyer is a much clearer-sounding system,” he recalls. “You can’t come close to the overall clarity.”

“Even out of the box, pre-tuned, the system was blowing people’s minds,” adds JDAV Design’s VP of engineering, Kevin Nellen. “Then we tuned it, and I was blown away individually, section by section. When we time-aligned everything and turned it into one big system, it was really pretty amazing.”

Perez is impressed with his customers’ response to the upgrade. “Both our clients and friends have been blown away with the clarity of the sound,” he says.

The main floor of the 40/40 Club features two UPA-1P and two UPQ-1P loudspeakers, five MM-4XP self-powered loudspeakers, two Meyer Sound 600-HP subwoofers, three MM-10 miniature subwoofers, and a Galileo loudspeaker management system with two Galileo 616 processors.

The mezzanine level houses two UPJunior VariO loudspeakers, three MM-4XP self-powered loudspeakers, three 500-HP subwoofers, and two MM-10 miniature subwoofers.

Upstairs are five VIP lounges available for private events, each outfitted with Meyer Sound equipment. The elit Lounge houses four 48 V, DC-powered UP-4XP loudspeakers and a UMS-1P subwoofer, while the Corzo Lounge features two UP-4XP loudspeakers, one miniature MM-4XP loudspeaker, and a UMS-1P subwoofer.

The Owner’s Suite and Player’s Lounge each include two UP-4XP loudspeakers and one UMS-1P subwoofer. Last but not least, the Jay-Z Lounge houses four UP-4XP loudspeakers and one UMS-1P subwoofer.

Even the bathrooms and hallway boast Meyer Sound in the form of six Stella-4C installation loudspeakers. And housed in the rack room are two more Galileo 616 loudspeaker management processors, as well as two MPS-488HP and one Stella-188 power supply units.

As for the club’s video, a Crestron DM-MD32x32 DigitalMedia matrix switcher provides control of the video wall arrays and flat-panel monitors.

“It’s a very complex system,” says Karlis, “but with all its capabilities, it ended up being very simple for the end user.”

Meyer Sound

Tuesday, April 17, 2012

PreSonus Releases Studio One Free Version

PreSonus is now shipping Studio One Free, a new entry-level version of its DAW for Mac and Windows that is intended for beginners who don’t yet need the advanced features in Studio One Artist, Producer, and Professional.

Studio One Free provides all of the recording and editing features needed for basic music production—and as its name implies, it’s free.

To get Studio One Free, download the Studio One installer from the Studio One website, install it, and then choose to run it as “Free” when the activation dialog comes up.

Once the application is downloaded, it can be used immediately. No Internet connection or user account is required, no product key is involved, and Studio One Free will not time out.

This new version of Studio One has some of the features originally found in Studio One Artist, including the single-window work environment, the content browser with Search, drag-and-drop functions, Control Link MIDI mapping, sidechain routing, automatic delay compensation, real-time audio timestretching and resampling, unlimited audio and Instrument (MIDI) tracks, unlimited effects channels, and unlimited channel inserts and sends.

Native key commands can be utilized, or users can choose key-command sets from Pro Tools, Cubase, or Logic. The included editor also allows users to create their own key commands.

Studio One Free offers some of the new Studio One 2.0 features, notably multitrack MIDI editing, single and multitrack comping, and Track Transform (advanced track freezing). And it also includes the same 32-bit sound engine found in Studio One Artist and Producer.

Eight Native Effects plug-ins come with Studio One Free - Beat Delay, Channel Strip (which includes dynamics processing and EQ), Chorus, Flanger, MixVerb (a mono/stereo reverb), Red Light Distortion, Tuner, and Phaser - plus the Presence virtual instrument with more than 100 presets.

Like Studio One Artist, Studio One Free does not support ReWire and third-party (VST, AU, etc.) effects plug-ins and virtual instruments.

Studio One Free does have limitations compared to Studio One Artist. For instance, a number of Studio One 2 features are not available in the Free version, including Folder Tracks, Transient detection and editing, groove extraction, and PreSonus Exchange integration.

Studio One Free presents a very easy to upgrade to from Free to one of the advanced versions. At any time, even after the demo times out, it can be converted into Studio One Free with a single click.

Studio One Free is available now from the PreSonus Studio One website.

PreSonus

Monday, April 16, 2012

Harman Offering Regional HiQnet System Architect & JBL HiQnet Performance Manager Training Courses

In association with rep firm Robert Louis Associates, Harman’s System Development and Integration Group (SDIG) has announced in-person training sessions for HiQnet System Architect and JBL HiQnet Performance Manager software in Pennsylvania and Ohio.

The training will be lead by Harman product application specialist Emilian Wojtowycz, an expert in the implementation of System Architect and Performance Manger for installed and live sound applications.

The training sessions will take place at the following dates and locations:

April 30 – May 1: HiQnet System Architect, Pittsburgh, PA

May 2: JBL HiQnet Performance Manager, Columbus, OH

May 3 – May 4: HiQnet System Architect, Columbus, OH

HiQnet is a communications protocol that enables all the compatible devices in the audio signal path, from mixing consoles to loudspeakers, to seamlessly communicate with each other. 

System Architect 3 is the software used to set up and configure a HiQnet system, by means of a series of control panels that are displayed on a PC and can be customized by the user.

System Architect enables designers to configure and control an installed sound system, using the HiQnet communications protocol that enables all the compatible devices in the audio signal path, from mixing consoles to loudspeakers, to communicate with each other seamlessly. 

System Architect 3 features a system design philosophy centered on workflow and the use of a diagrammatic representation of the installed or live sound venue. Devices are arranged by both their physical and logical placement allowing the designer to ‘educate’ System Architect about how they are to be used. In return the software is able to provide automation of many of the laborious system design tasks for free.

Attendees of the System Architect courses will learn:

·    Design workflow modes

·    Overview of Ethernet AVB technology and AVB routing

·    Comprehensive design workflow and system tools

·    Custom and master panel creation

·    Going online and networking

·    Day-to-day operation

Both System Architect courses award attendees with 5.5 InfoComm CTS RU credits.

JBL HiQnet Performance Manager is a software application derived from the System Architect core code but tailored for live sound operation. It is designed to configure networked audio systems within performance venues such as theatre, house of worship, and corporate and other performance sound events.

Attendees of the Performance Manager course will learn the way in which Performance Manager guides the configuration of system design workflow:

·    Working with array templates and the JBL Line Array Calculator II tool

·    Adding passive or powered speakers automatically

·    Adding and associating amplifier racks automatically

·    Simplified drag-and-drop networking

·    Using the built-in test, tuning and calibration control interfaces

·    Running and monitoring the system with the dedicated show mode

“System Architect and Performance Manager are two powerful tools that make the design, setup, and tuning of audio systems faster and more efficient than ever before. These training courses will provide attendees with the knowledge to harness these resources for their everyday use,” stated Adam Holladay, market manager, Harman SDIG.

To apply for a course, go here on the Robert Louis Associates website. 

Harman HiQnet
Harman

New Version 4.7 Software For Studer Vista Consoles Brings Graphic EQ To All Channels

The latest release of software for Studer Vista digital consoles brings 30-band graphic EQ functionality to all channels on the desk, whether they are inputs, buses or outputs, in addition to the existing full 4-band parametric equalisation.

Assignment and selection of the graphic EQ is via the Vistonics touch screen, and the control of the frequency bands is then placed on the channel faders, or via the rotary controls in the Vistonics section.

On Vista 9 consoles, the FaderGlow illuminates red when the faders are used for graphic EQ, with the channel name screens indicating the band’s frequency. The start band can be placed at any suitable fader position on the console for convenient user location.

A combined EQ curve is also displayed in the Vistonics section.

The software is available as an upgrade for all Studer Vista consoles which utilise the SCore Live DSP core, and is dependant on DSP configuration.

Studer
Harman

Friday, April 13, 2012

Church Sound: The Art (And Necessity) Of Compression

A while back, I got the rare opportunity to work with the youth band at our church.

These guys have an incredible heart and passion to worship and have loads of raw talent which translates into a powerful time of worship.

When they lead, as a worshipper I feel free and emboldened to praise God the way He created me too.

When they lead, as a sound operator I have to work as hard and quick as ever to create a decent mix to help facilitate that worship.

More Compressors Please

As is the case with most youth bands and even many churches, they’re not using state of the art or high dollar gear for their services.

Now don’t get me wrong, they are not operating with the bare minimum. The system includes an Allen & Heath console, JBL loudspeakers and subwoofers, and solid system and signal processing. But what I longed for that night was individual compressors.

Keeping The Vocals On Top

Maybe this never happens to you, but in a mix including three vocals, an un-caged drum set, two electrics, acoustic, bass, and keyboard, I had a hard time keeping the vocals out on top to lead the group in worship while keeping the music strong.

The vocalists on the team are gifted in leading worship, but for a variety of reasons (key of the song, dynamic range, mic etiquette, etc) their volumes were all over the place that night and the second I took my finger off their faders I would either lose them or have way too much of them.

With seven stage monitors, acoustic drums, three amps and a very small stage, I was dreaming for a few compressors to help me layer the mix the way I wanted.

Most Useful Tool

The compressor is one of the sound man’s most useful tools - yet I’m always surprised how few people seem to understand and know how to effectively use this critical piece of gear. I would like to help a few more of you get comfortable using compressors.

What Is Compression Really?

The clearest definition of compression that I’ve ever seen is this: “Compression is the art of making louder parts of a composition appear softer, and conversely, the softer parts appear louder.”

That night, if I would have left the lead singer’s fader in one spot for the entire night his volume alone would have ranged anywhere from 85 dB to 120 dB. Alright that might be an exaggeration but he got loud.

When he was closer to the 85-95 dB volume he could barely be heard over the drums and guitars. Neither end of the spectrum is really acceptable in a good mix, so compression comes along and makes it possible to narrow down that volume range to make things more mixable.

Example: How A Compressor Works

Let’s say I have a 20 dB range between a vocalist’s quiet singing versus their loudest singing. With a compressor I can take that 20 dB range and make it as small as a 1 dB range, but since I don’t want to eliminate the artistic dynamic range that the singer is using to create the mood or feel of what they are singing, I can get that range down to a very manageable 5-8 dB that will make mixing significantly less complicated but still leave some of that dynamic in place.

So how do we get our compressor to do that? With some understanding of the compressor’s settings you can be on your way to a smoother sound and a less stressful time behind the board.

Control Elements

Threshold. In simple terms the threshold is the point where the compressor starts to do its thing.

Since there is a wide range of compressor and mixer brands I’m going to talk about these settings more generically as opposed to using the numbers on the knob. If the input meter on your console (let’s say negative infinity to +15 dB) matches that of your compressor, things will be a little clearer as the numbers will match.

You must first set the gain (or trim) of your channel on your mixer (on my regular console that is around +3, or typically where the green lights first turn to yellow or maybe the yellow light just starts to glow on the meter).

Now if your numbers match, and your vocal meter is showing signal between -5 and +10 dB, I’d start with my threshold set close to 0 dB. If your numbers don’t match, once your gain is set turn your threshold knob and find the area where the gain reduction knobs just come on.

Begin with your threshold there and if you find it’s not compressing frequently or soon enough you can lower the threshold from there to make it kick in sooner.

Ratio. This one is a relatively simple concept. The ratio simply says for every “X” dB the source goes up in volume, the compressor will only let the output go up “Y” dB.

For example, if you set a vocal mic with a 3:1 ratio, for every 3 dB the vocal increases coming into the board, the output will only increase 1 dB.

You can think of the ratio as setting the size potential of the source. If you want it to be able to go bigger, you can leave your ratio smaller. If you want it to stay a little smaller, or perhaps be more under control, you can set your ratio higher.

I tend to start with a ratio of 3:1 for most vocals and guitars, and often times I will go 4:1 or even 5:1 on drums or very dynamic guitars.

My preference is to start low and if you need more compression (less range) you can always increase the ratio. The reason it is my preference is simply this, I don’t want to take away any more control from the musicians than is absolutely necessary to make the mix work well.

If I start it at 5:1 when 3:1 will do and don’t adjust it down, I may be holding that source back. If I start low and it’s still too big, I can easily adjust my ratio up.

Attack. The attack is how quickly the compressor responds to the volume change. A slower attack will sound a little smoother, rounding out the sound of your source a little bit and in essence making it sound a little “fatter.”

A slower attack will generally be less noticeable which can be good for vocals and some thin or scratchy guitars. Setting your attack to a faster setting can be great for instruments such as drums or any other very aggressive instruments.

A faster attack will give an instrument more of an aggressive, pumping feel, and potentially bring out more of the high end edginess. The ultimate decider on where to set this is by listening. I tend to set vocals a little slower, guitars in the middle, and drums faster to start.

From there, if you need a little more aggressiveness or snap you can speed up the attack, and if it needs to be a little smoother or fuller you can slow it down. As in all things with sound, let what you hear guide your settings and adjust until you are happy.

Release. The release is the back side of the attack, and sets how quickly you want to release the compression once that loud burst is over. As with the attack, a slower release will sound smoother and less noticeable but could end up taking some of the aggressiveness out of aggressive instruments by compressing them when they don’t need to be. I again will tend to start a little slower for vocals, middle of the road for guitars, and faster for drums.

You’ll want to again experiment with where to set this by listening to the sound. If the source sounds like it is pumping a little bit, slow the release down to help even it out a bit. If it feels like you might be losing something on the next note/beat, you likely need to speed the release up a bit.

Again, let the sound of the source guide you to where it should be set. Listen and adjust until it sounds right to you.

Output. Most compressors have an output to help boost the volume of the end result, and here’s where I tend to see a lot of mistakes made.

Now that you’ve taken that 85 to 105 dB vocal and compressed it down to a manageable 85 to 93 dB, you may need to increase the output a little to get it over those guitars and drums. Instead of reaching for the gain or trim knobs (which would then bring more signal into the compressor and would change how you’ve set your compressor), if you add 5 dB of gain to your output you just took that 85-93 dB and made it 90-98 dB.

Especially Useful in Worship Environments

Compressors are a huge help to the sound man and used right they will help you get great sound out of your sources and give you the ability to get the mix where you want it. They’re especially useful in the worship environment where the voice of those leading the worship must always be present but not piercing, where more and more guitars are being used to lead the music but can’t overtake the vocals, and where many churches use acoustic drums.

Wrap Up

I truly believe that no one setting is right for any vocal or instrument. If you start with a lower basic setting and then adjust based off of what you are hearing, your compressors can give you a great edge to get your mix balanced and layered according to plan. Just remember, you don’t want to compress something more than you need to. If you’re having trouble keeping a source in it’s place in the mix the compressor is the tool to help you make that happen.

Duke DeJong has been involved in live production for over 15 years, has spent 10-plus years in full time ministry, and in 2011 began serving as the church relations director for CCI Solutions. You can find him online at dukedejong.com or on Twitter.

 
Check out more from David McLain at the Church Soundguy blog.

PreSonus Adds New Features To Studio One 2

PreSonus is now shipping Studio One 2.0.5, a free update to all versions of the company’s DAW software for Mac and Windows that adds several important new features and fixes a variety of issues.

Among the new features:

—Any Studio One command can now be assigned to any MIDI CC message. Use any MIDI control surface with Studio One, even if it isn’t natively supported.

—Ampire XT amp models have been reworked to sound significantly better.

—Markers can now have a Stop flag that stops playback at the marker.

The new Macro Toolbar is an extension that allows powerful customized workflow within Studio One. With this feature, users can create macros that string together multiple commands to form a single action.

Groups and buttons can be freely added to the Macro Toolbar for existing commands, making it possible to bring what is important to your personal workflow to the surface.

Users also can map buttons on a MIDI controller to trigger macros, even when using control surfaces that are not natively supported.

In addition, the Studio One manual is now available in German, Japanese, Spanish, and French.

The free Studio One 2.0.5 update includes more than a dozen additional enhancements and fixes a variety of issues—for a complete list go here.

PreSonus