Friday, May 23, 2014
Clear Path: Keyboards In The Electronic Realm
Electronic keyboards, the start of it all. Right from the beginning of modern concert sound, DI boxes have played an essential role in getting the sound from the stage to the PA system.
Probably the most iconic “direct” instrument of all was the Fender Rhodes. Harold Rhodes started developing the idea as far back as the 1950s, but it was in 1970 that the Rhodes Stage piano took the concert stage bringing the first “portable” keyboard to market.
The original Rhodes piano tone was created by a piano-like hammer striking a “tine” that would vibrate up and down in front of a magnet to create the tone—very much the same way an electric guitar string vibrates atop a magnetic pickup. One would adjust the tone by changing the “tine-to-magnet” relationship.
And like an electric guitar, the output from the suitcase was not amplified (or buffered) in any way. So the output from the piano was generally sent to a guitar amp where it was mic’d.
Some years later, the first active DI boxes came round. They didn’t load the Rhodes pickups, which made it practical to send the “direct” sound to the PA system and monitors.
But something happened. That something was Keith Emerson and Rick Wakeman, and the Moog synthesizer, which found its way out of the electronic music department to the stage. These guys no longer had one or two keyboards—they had racks of them!
An early Fender Rhodes, the one that started it all.
The Arp 2600 and String Ensemble, Oberheim, Korg, the venerable Sequential Circuits Prophet 5 - it was an analog explosion. Everyone had a Rhodes (or Wurlitzer) and a bunch of synths.
Fast forward to 1981, and Yamaha introduced the DX7, which would go on to become one of the most successful keyboards ever. It brought along something totally new: frequency modulated digital technology. Now you could get a bell-like Rhodes sound without the weight.
The world then changed again with the E-MU and the Akai S900 digital sampler. All of a sudden, we had complete orchestration, real sounding piano samples, and digitally sampled drum tracks were everywhere. There was no going back.
Today, pretty much all keyboards and drum machines are digital, and can basically be thought of as keyboard controlled CD players. And like a CD player, the output from a digital synthesizer is relatively powerful when compared to an electric guitar or an old Rhodes piano.
Because they’re so “loud,” they needed headroom to operate, meaning that the old active DI box that may have been a boon to the low-output Rhodes piano can no longer keep up.
The headroom is limited by the internal battery or limited by the low current afforded by phantom power. To make matters worse, unlike a CD that is processed and compressed before it is mass produced, digital samplers are raw. They can generate huge transients that will overload most active DI boxes, and end up distorting horribly.
The problem is further exacerbated with digital pianos. These full-range devices are not only very dynamic, they have a frequency range that starts way down low and goes up forever.
To handle modern keyboards, there are two choices:
1) Send the keyboards into a mixing console where the internal rail voltage is sufficiently ample that it is able to handle the range.
2) Send the signal to a passive direct box where the headroom is not limited by the current afforded to them. Passive DI boxes are different- they use transformers.
Phantom Of The Power
Replace the diesel engine inside a dump truck with a 4-cylinder car engine and fill the truck with gravel. What will happen? Nothing. The engine will be unable to handle the load.
The same applies to phantom power. Folks tend to “believe” that if it’s active, it must be good. But the truth is, phantom power was never designed to power direct boxes.
As noted here, phantom was invented by Mr. Neumann as a means to charge the capsules on his microphones. He needed a lot of voltage (48 volts) and very little current (5 milliamps).
A quality preamp requires +/- 16 volts (32-volt swing) and about 50 milliamps of current. With 1/10th the current, it’s like trying to run a dump truck with a motorcycle engine.
Passive direct boxes are not power limited. They’re old-fashioned devices that basically combine a couple rolls of wire (coils) with a chunk of metal (the core).
A DI has the task of converting a high-impedance signal to a low-impedance balanced signal where it can be managed by the mic splitter and mixing console’s preamp.
With keyboards, current enters the transformer where the conversion occurs. But instead of overloading. like an active circuit, transformers distort gradually. More precisely, they don’t so much distort as saturate.
We often say that transformers sound “vintage” or have a limiting quality about them. This is because good quality transformers generate warm sounding even order harmonics, or what is commonly known as a warm Bessel Curve.
Thus the reason highly dynamic buffered signals like digital keyboards sound great when they are used with a passive direct box.
Mono, stereo or multichannel - which DI is best? It depends upon your point of view.
If audience members are sitting right in front of the left PA loudspeaker, they will be unable to hear what is coming out of the right PA loudspeaker. Will they benefit from stereo? Probably not.
Stereo may sound great in the practice room or be invigorating on stage, but in most live venues, it is rarely enjoyed by all. The advantage that a stereo DI brings is capturing the stereo sample without having to reprogram the synth.
And if you do decide to have a stereo rig on stage, a stereo DI allows the house engineer to mix both channels and pan them stereo (if beneficial).
A stereo DI is often used at the output of a keyboard mixer. Here’s why: on a stage, all of the microphones go to a mic splitter before the signal is sent to the house mix position. Mic splitters are designed to handle mic levels, typically around -50 dB. A keyboard produces a -10 dB signal while a mixer can produce +4 dB or more.
A look at one reason why a passive DI can be a better choice for electronic keyboards. Courtesy of Radial Engineering.
This excessive level will cause the mic splitter to overload. The pad on a direct box lowers the output so that it matches that of a microphone and protects the mic splitter from being overloaded by the mixer. Multichannel direct boxes bring forth the added advantage of independent control over each instrument.
Here’s the deal: when you mix the sound so that it is comfortable on stage, it may in fact not be ideal for front of house. In other words, you may find that you need extra jam to hear your piano on stage, and have the string synth pushed back in the mix.
But in the arena, the piano-to-string volume ratio may not sit well with the rest of the band. What happens? The keyboard mix gets pushed back.
When you have the luxury of sending independent keyboard signals to front of house, the mix engineer is allowed to orchestrate. With more control, the engineer can decide how much piano fits and if the strings are too loud, can simply back them off.
Who would have ever thought that a DI would have so many twists and turns, especially after being around for 40 years!
Peter Janis is president of Radial Engineering. In 1982, he was hired by CBS Fender as new product director for what would eventually become Fender Canada, and spent time at the ARP factory in Massachusetts learning to program the advanced Chroma polyphonic synthesizers. He met Harold Rhodes and spent several years servicing Rhodes pianos before they were eventually discontinued, and during that time, also added the Akai product range to the Fender sales portfolio and developed many of the Akai’s early samples.
Wednesday, May 21, 2014
Properly Setting Sound System Gain Structure
Realistically, audio signals at or near the noise floor of a system are not useful because the signal will not be significantly louder than the noise. Therefore, some minimum usable level must be assumed below which the electronic noise is considered objectionable.
A signal-to-noise ratio of 20 dB is considered minimally acceptable for good intelligibility. For a high-quality system, 30 dB would be a better figure to use. Using this value, the range from this minimum signal level (30 dB above the noise floor) to the clipping level is the usable signal range window for the system (also called the dynamic range in my way of thinking).
However, for purposes of this article, the maximum output to noise floor is used as the dynamic range.
Every audio system with more than one electronic component has a “system gain structure”. Gain structuring for a system occurs in the signal processing chain between the mixer or another signal source and the power amplifiers.
One usual scenario is to set all the signal processors to unity gain and turn the amplifier inputs to maximum. Unfortunately as you will see, given the different maximum outputs and noise levels of typical signal processors, this method will may not come close to the best gain structure.
We will be dealing with the signal voltage levels on the interconnecting cables from the output of the mixer (or signal source if there is no mixer) up to the input of the amplifier. For the convenience of using simple numbers, this analysis uses relative dB, as a voltage ratio where dB = 20 x log (V1/V2), and dBu, where 0 dBu = 0.775V. V1 and V2 are simply two voltages.
To set proper gain structure, the interconnections between devices must be constant voltage interfaces. This means an output device’s voltage at any point in time is unaffected by whether or not it is connected to the device(s) it is driving.
This type of interface is characterized by the output impedance of a device being 1/10 or less of its load. For example, if the output impedance is 100 Ohms, the total load it drives must be 1000 Ohms or greater. Virtually all professional audio equipment meets this criterion when a single device drives only one other device.
However when one device drives multiple devices, such as a mixer feeding a number of power amplifiers, this may not be true. In this case a distribution amplifier may be needed to divide the load between its multiple outputs.
The last thing to consider is the power handling of the loudspeaker(s). As long as the amplifier does not exceed the loudspeaker’s power handling capability and the system is operated without clipping, you should never blow a properly manufactured loudspeaker.
The safest criteria to use in selecting an amplifier is the RMS rating of the loudspeaker. In reality, most loudspeakers can handle peak signals in excess of this rating.
A reasonable choice is an amplifier whose rating that is 2 times (+3 dB) the RMS rating of the loudspeaker. The RMS sine wave used to rate amplifiers has an inherent peak power component of 3 dB.
So this all works out to a 6 dB allowance for power peaks over the loudspeaker’s RMS rating. This is a pretty safe figure for the way most professional loudspeakers are rated (pink noise with a 6 dB peak factor) and given the peak to RMS content of most audio signals.
However, sustained sine wave signals from the likes of a synthesizer could exceed the loudspeakers RMS capability by 3 dB without clipping the system. If you expect these kinds of signals and you expect to drive the system to maximum output levels with them, use the loudspeaker’s RMS rating as the power rating for the amplifier.
With these basics in mind, we’re ready to examine how to achieve proper gain structure in detail.
Picturing Gain Structure
Before you get out your equipment and start setting gain structure you have to learn just what it is you are trying to accomplish. Go through the following “on paper” analysis of a typical system.
Only after you understand this can you appreciate where to actually set the controls on equipment to achieve optimum gain structure.
Figure 1 shows a simple system consisting of six pieces of equipment. The device clip level (maximum output) is listed for each device as published by the manufacturer. For this example, all devices between the mixer output and the amplifier input are set for unity gain and the amplifier input is set for maximum sensitivity.
Each device is represented by what looks like a bar. Rather than a bar, picture it as a tall, narrow window. The maximum output or clipping point from the specifications for each device defines the top of the window using the absolute dBu scale on the right.
The published noise floor (or signal-to-noise ratio) specification below maximum output determines the height of the window. The relative dB scale on the left is used to determine this height. All usable signals must pass between the top and bottom of the window.
However, remember that your low level signals won’t be near the noise floor. Realistically the minimum usable signal is one that is at least 30 dB above the noise floor.
Next, a horizontal line is drawn across the top of the lowest window (in this case the amplifier). This is the system clip level, and for the rest of the analysis this line stays in the same place. Another line is drawn across the highest bottom window sill (in this case the mixer).
The relative dB scale is used to measure the distance in dB between the 1st and 2nd lines. As you can see, it is only 72 dB for this set of devices and gain structure.
That’s equal to the performance of your average consumer cassette deck—and you thought that professional equipment automatically guaranteed a professional grade audio system. Oh well, live and learn!
Now subtract 30 dB to find the “true” dynamic range (30 dB above the noise floor to the clipping level). The result is 42 dB.
Measurements of the maximum dynamic ranges for acoustic instruments and voice yield maximum figures in excess of 40 dB. This means our system really doesn’t have enough dynamic range to reproduce them.
Most Common Approach
As seen in Figure 1 from the absolute scale on the right, the amplifier input sensitivity limits the maximum signal level in all the other devices to +3 dB.
Above +3 dB the amplifier will clip—period. It doesn’t matter how much “headroom” is in the mixer, you can’t use it without distorting the amplifier.
Well, you say, the obvious step is to put a pad (usually the amplifier input attenuator) so the amplifier will clip at about the same point as the next least capable device.
In this case it is the notch filter. Using a -12 dB pad, the notch filter and amplifier will both clip at once and the signal level will be 12 dB higher through the other devices at the amplifier’s maximum output.
The chart, as seen in Figure 2, was changed from Figure 1 by moving all the device windows (except the amplifier) down by 12 dB using the relative dB scale on the left. A +15 dB signal (the notch filter clip level) is now attenuated to +3 dB by the amplifier’s input attenuator.
The noise floor line is redrawn through the highest window sill (in this case still the mixer). Because this ends up 12 dB lower than in Figure 2 relative to the system clip level, we see that the system’s overall window height is now 84 dB. This is a 12 dB improvement - much better.
Note that the absolute device clip levels no longer relate to the absolute dB scale except for the amplifier’s input after its input attenuator. Our usable signal range (from 30 dB above the noise floor) is 54 dB. This means our system is now able to squeak out enough range to reproduce the dynamic range of instrumental and vocal sources.
Unfortunately, the mixer is still the primary noise source by 3 dB over the signal delay. However, according to their published specifications, the mixer should have some 6 dB better noise performance than the signal delay.
It should also be obvious the signal delay is the weakest dynamic range link because it has the shortest window. Therefore, we must conclude that there is more that can be done to optimize the system’s gain structure.
You Can Make It Better
To optimize the system, pads or gain must be added at the input of each device so that its clipping level and the clipping level of the preceding device occur at the same point. Think of the following procedure as a graphic picture of what would happen to the signal on a volt meter as you work your way through the system.
To create the chart shown in Figure 3, the windows are shifted up and down as needed so that all the tops are lined up on the system clip level line. To do this, start with Figure 1 and work from left to right in signal flow fashion. If you move a window up you need gain between it and the next device. If you move a window down you need a pad.
First, move the mixer window down so its top is even with the graphic EQ window. This movement is measured on the relative dB scale, which in this case is -6 dB. Therefore, you need a 6 dB pad at the input of the graphic EQ.
Next, move BOTH the mixer and graphic EQ windows down together so the graphic EQ window is even with the top of the notch filter window. This also turns out to be -6 dB. Therefore, a 6 dB pad is needed between the EQ and notch filter.
Now, move the mixer, graphic EQ and notch filter windows together so the top of the notch filter window is even with the top of the signal delay window. To do this, move ALL the previous devices up 3 dB. This means you need 3 dB of gain between the notch filter and signal delay.
Repeat the process again by moving the windows of the first four devices together so the top of the signal delay window is even with the limiter window. This distance equals 3 dB. This means 3 dB of gain is needed between the signal delay and limiter.
Lastly, you must lower ALL device windows to line up with the input to the amplifier. They are moved the distance between the top of the limiter and the top of the amplifier. In this example the distance is 18 dB. (In the actual system this would usually be done with the amplifier input attenuator.)
After completing all these steps, the tops of the windows end up on the system’s clip level line as shown in Figure 3 (+3 dB on the absolute dB scale). Looking back from inside the amplifier after its input attenuator, all devices appear as though they are clipping at +3 dB. In reality, they are all clipping at their specified device clip levels. If one device is clipping—everything is clipping.
The Results Are Worth It
Measure the distance between the system’s clip level line and the bottom of the shortest window using the left-hand scale. This result is 90 dB which is 18 dB better than the raw system gain structure in Figure 1. The “true” dynamic range, considering a 30 dB above the noise floor signal as the minimum, is now 60 dB.
Also, the primary noise source is now the signal delay. This is the weak link, which agrees exactly with the published specifications for all the devices. It also should be apparent that more of the usable signal window within each device is being used. What a concept.
In some cases, pads or gain will not have any effect on the overall usable signal range. For example, in Figure 3, the 3 dB of gain at the output of the signal delay could be omitted. The 18 dB pad for the amplifier input would become only 15 dB, and the top of the limiter window would end up 3 dB above the system clip level. The key here is that the bottom of its window is still well below the noisiest device (in this case the signal delay).
You can use this reasoning to save yourself the hassle of making up small pads or small amounts of gain. If you omit one of these along the chain you MUST move all the devices preceding it up or down in your chart by the dB of gain or loss that you are omitting. Otherwise, you will not see the effects of the omission on the noise floor.
Now that you have set up proper system gain structure on paper, it is time to hook-up the system and do the same thing for real. Once completed, audibly evaluate the noise floor heard from the speakers. If all is quiet, pack up and go home. If the noise floor is too high, there are two possibilities:
A. The maximum sound level is higher than necessary, which means you over-designed the maximum capability for the system. If this is the case, turn down the amplifier input attenuator. You will lower the noise, and the maximum output level for the system will be reduced by the amount you decide is over-kill.
B. The maximum sound level you can get out of the system IS necessary, which means your system does not have enough usable signal range. You now have three choices; the first two are compromises.
1) Accept the noise and achieve the maximum sound level you need.
2) Turn the amplifier input down to make the noise acceptable. This will, of course, reduce the maximum output level capability for the system. (Sorry, you can’t have it both ways unless you pick choice 3.)
3) Change the primary noise source in the system to something with lower noise performance.
Doing Your Own Analysis
A similar chart for setting up proper system gain can be created for any system. Using graph paper, make a vertical absolute dB scale from about +30 dB to -120 dB so you can plot increments for 3 dB or less.
The relative dB scale simply uses the same graph increments for plotting and measuring distances in dB. You could also follow this procedure by using some simple math.
If you don’t trust your addition and subtraction, or would rather work with pictures (they are more dramatic and will quickly show errors in your thinking) cut out rectangular paper bars (windows) like those shown in the figures.
The length of each should equal the distance in dB between the device clip level and its noise floor. Be sure to convert noise figures to noise below maximum output.
Write in the clip level for each device on its window. Using these numbers and the absolute dB scale, position the top of each window on the graph paper in signal flow order from left to right. Move these “paper cut-outs” up and down on the chart as outlined above, by measuring the distances using the relative dB scale. You can very quickly determine the necessary pads and gains—probably faster than with a calculator.
A way to check your work is take the maximum output for the first device and subtract the dB for the all the pads and the gain to that number, including the pad before the amplifier. The result should equal the maximum input sensitivity for the amplifier. This calculation should give math mavens an interesting insight into the gain structure process.
Doing It For Real
To actually adjust a system you need to do exactly what you did on paper except you are now doing it for real. You start from the console output and find out what you need (gain or loss) to adjust its maximum output signal so that it just drives the next device into clipping. And so on.
You don’t need to know the specifications of the equipment. When you go through the system you’ll find out what those specifications are in terms of maximum output levels. As you should have understood by going through the exercise on paper, the noise floors of the equipment will take care of themselves.
Some device (like the signal delay in the above example) will be the weak link. There is nothing you can do to make this better except to replace it with a device with a better maximum output to noise floor window (better signal to noise ratio specification.
Because of production variations and possibly conservative specifications, you may be able to pick up a few more dB of dynamic range by adjusting the pads or gain values you determined on paper. If things are not reasonably close to your on-paper calculations, you have a problem such as bad wiring or a misadjusted or defective device.
What To Adjust
When you set gain in the system, the attenuation or gain needed between devices can be added externally or by using a device’s input level control, if it has one. DO NOT ADJUST THE OUTPUT LEVEL CONTROL ON ANY DEVICE—this should be left at maximum. This is because it is rarely the last thing in the internal circuitry before the output connector. Unlike some input level controls, it usually does NOT adjust actual gain.
Therefore using it will squash the dynamic range in that device’s output stage and you may end up making things worse, even though the signal level is matched up to the next device. Use an output control only if you KNOW ABSOLUTELY that it is a simple attenuator feeding its output connector. The reason it usually is not is that this topology would cause changes in the output impedance when the control is set for anything other than maximum.
Among other things this could would wreak havoc with - guess what - the gain structure. If the device has a noise floor below other devices when you have set the overall gain structure, you can use output gain.
But reduce it only by 3 dB less than the amount between the device’s noise floor and the device that determines the noise floor of the system. This is because if you bring its gain, and hence its noise floor, up to the worst case device its noise will add to the worst case device and give you 3 dB less dynamic range.
The Tools You Need
To find the clip points in a system, you need to use an oscilloscope and a pink noise test signal. There is really no good substitute for this equipment to set gain structure. Sine wave signals are not recommended as they only show one frequency at a time and you can easily miss something.
The pink noise should be full-range (20 Hz - 20 kHz) and have at least a 6 dB peak to average ratio. If you can find one with a 10 dB peak to average ratio, you will more closely simulate real audio signals.
If you must use sine wave signals, you will have to check each and every EQ boost frequency or range of frequencies very carefully. When measuring electronic crossovers or other frequency response limiting devices, only a full-range pink noise signal will allow you to see full-range signal energy losses easily. (See sections on crossovers and band limited devices.)
If using sine waves you must set the frequency to the center point of each frequency band of the crossover or the center of the band pass for a band limited device.
For simple systems (e.g., no electronic crossover), there is “poor man’s” method where you use a Piezoelectric tweeter and a 400 Hz sine wave to find clip levels.
Basically, you connect the tweeter directly to the output of each device. When the device hits clipping, the tweeter will emit a very noticeable buzzing sound due to the harmonics in the clipped signal.
For high-powered amplifiers, a resistive pad should be used to avoid burning out the tweeter. This method is detailed more rigorously by Pat Brown of Syn-Aud-Con. You can find this information here.
You start the whole procedure by inputting the pink noise test signal into to mixing console. Set it so that it’s output just clips as seen on the oscilloscope.
Make sure it is the output of the mixing console that is clipping. Determine this by reducing the master fader. The clipping should stop. If it doesn’t, you are clipping something before the output fader.
While you’re at this point, note the reading on the output meter. This is a good indication of what the meter will read when you have reached the system’s maximum output after you set its gain structure. If you are using sine waves, this will NOT be a reliable indication.
Once completed, if the system noise levels are low enough, you may want to increase the setting of the amplifier(s) input level control. This will make the mixer more sensitive for operation.
If you reduce the amplifier input level control, something in the front end of the system will clip first. This means the amplifier will not reach full output. But it WILL reproduce that clipped signal and possibly damage the loudspeakers. Either way—if you choose to increase or reduce the amplifier’s input sensitivity from the optimum gain structure setting—you really don’t gain (pun intended) anything.
There is possible exception to this: by reducing the amplifier’s input level control, the output meters on the console will indicate you have reached the system’s maximum output before the amplifier’s clip.
This is useful so that a less than capable mixing engineer will THINK he’s pushing things to the limit but there will still be something left in the amplifiers. This may help protect the loudspeakers but, bear in mind, it will limit the maximum output of the system to something less than it could be.
Note that to reach a system’s maximum output analog Vu meters on mixing consoles may “peg” before the system clips. If you can afford the reduction in dynamic range, operating the system so the meters don’t peg means you’ll never clip the system. Generally, this means you won’t ever blow the loudspeakers assuming the amplifiers are chosen not to exceed the loudspeaker’s maximum ratings.
More Complex Situations
Up to now we’ve looked at a simple systems. Here is where gain structure gets more complicated. However the ideas are exactly the same. You just have to think about what specific pieces of equipment do and/or about more signal paths.
Devices with Gain/Loss and EQs: Parts 1 - 3 assumed devices in the signal chain have no gain (unity gain devices). However, a device may have gain or loss, or you may want to allow for boosts in an EQ, which may be needed to tune the system.
EQ boosts are like adding overall gain to the device. In such cases, as illustrated in Figure 4, input of the device’s window is shifted down below the output of the device’s window. The distance will be the gain in dB for this device or the desirable dB boost you choose for the EQ.
In this case, it is assumed the boosts will be limited to a maximum of 6 dB. Match the top of the window of the preceding device to this line. On the output side you still use the top of the window to match it to the next device.
When adjusting gain in an actual system, first set up the system with the EQ set to flat. Then make any EQ adjustments.
If all of your EQ is cut only, you can usually leave everything as is. However, if you add ANY EQ boosts, you will then have to redo the gain structure starting from the input to the EQ by finding the new maximum level it can accept without clipping.
This will of course require attenuation at the input to the EQ input. In some instances a device might introduce a loss in signal level.
The procedure is similar except that the output side of the device’s window is shifted to below the input of the device’s window a distance equal to the loss in dB.
Use this new output point to match the device to the top of the window of the following device. In the example it is assumed the limiter threshold is set so the maximum signal through the limiter is 6 dB below its maximum output.
However, on the input side you still use the top of the device’s window for matching to the preceding device’s window. The relative signal level prior to the limiter is 6 dB higher. As shown in Figure 4, everything, including the noise floor is raised 6 dB. The system dynamic range is still determined by the signal delay, because that is still the smallest “window” in the overall picture.
Because the maximum input of the amplifier (after its input attenuator) is still + 3 dB it has remained in the same position throughout
Multiple Signal Paths, Arrays and Delays
Another variation in this procedure is when a system has several branches, such as a mixer feeding multiple sub-systems. You have to separately analyze each branch and include in each analysis the source common to all branches (the mixer in the example Figure 5).
This will automatically optimize the system so that the common source and all the branches clip at once. To do this, the mixer In Figure 5 must feed each branch through a separate pad. Note that the dynamic range is different in each branch.
To balance the multiple branch systems acoustically in the actual system, you will probably need different operating levels in the branches than what the optimized electronic gain structure provides.
An example would be a central cluster with delayed balcony speakers. To balance operating levels in these instances, use the branch that is lowest in acoustic level as your reference branch (i.e. the one you are itching to turn up because it isn’t loud enough - but don’t touch that dial). Use the input attenuation on the amplifiers for each of the OTHER branches.
This will reduce their output levels and achieve proper acoustic balance with the reference branch. This will also have the effect of lowering the noise levels and reducing the maximum capability of the other branches. In this case, less capability is acceptable because you have determined that the maximum capability can’t be used in these branches unless you drive the reference branch into clipping.
However, if you find, for example, that you have to significantly reduce the maximum output capability of the central cluster so you won’t clip the balcony system, then your balcony system is under-powered. Instead of attenuating the central cluster, you could add gain prior to the balcony system amplifiers (or “unattenuate” the amplifier input).
While this will balance the system, the balcony amplifiers will be driven into clipping before the central cluster amplifiers.
In this situation, the only way you can have your cake and eat it too, is to increase the size of the balcony amplifier, which translates to more voltage (power) capability for the balcony speakers.
You will not spot this problem by analyzing the electronic gain structure.
This could only have been spotted on paper with proper analysis of the acoustic output for each branch based on loudspeaker sensitivities and listening distances.
Electronic crossovers require special attention. Consider a full-range signal with equal energy per octave (e.g. pink noise). A crossover will divide the total energy of such a signal among two or more frequency bands. This causes an inherent signal loss at each band-limited output, compared to the full-range crossover input signal.
In effect, crossovers are NOT unity gain devices when fed a full-range signal. You can approximate these losses by calculating how much of the total energy is in each frequency band by using the following procedure:
Example: A 3-way crossover with frequency bands of 50 Hz - 125 Hz, 125 Hz - 500 Hz, 500 Hz -10 kHz.
1) Multiply the lowest frequency in each band by 2 until you get to the highest frequency for that band. The number of times you multiplied = the number of octaves. Round off the results for each band to the nearest whole octave [= 1, 2, 4].
2) Add up the total octaves from all bands [= 7].
3) Divide the octaves in each band by the total octaves [= 0.14, 0.29, 0.57]
4) Push the LOG key for each result [= -0.9, -0.6, -0.2].
5) Multiply each result by 10 to find the approximate losses [= -9 dB, -6 dB, -2 dB].
Note the low frequency output is down almost 10 dB. That is why many systems have problems achieving enough drive levels for the subwoofers.
Now you must draw horizontal lines on the output side of the crossover’s window. Draw these lines at a distance below the top of the window equal to the loss in dB for each output as found above. This line for each crossover output is used to match up the crossover window to the top of the window of the device it feeds (usually an amplifier).
In the example, a different pad would be needed for each output (assuming the amplifiers have equal input sensitivities). The top of the window of the device feeding the crossover is still matched to the top of the crossover’s window.
In the actual system, the amplifier input levels are adjusted to acoustically balance the system similarly to a multiple branch system. Use the frequency band that you want to turn up the most - typically the subwoofer (but of course you won’t turn it up - right?) as the reference output. Balance the other bands to it by turning DOWN their amplifier input level controls.
Once you have the system balanced to your acoustical liking, you may find that amplifier input level controls, in particular for horn amplifies, may be set too low for them to reach full output - even with a single frequency sine wave in their pass band. You can increase all the amplifier input level controls by the same amount to get some or all of this unusable capability back for limited frequency range signals.
Keep in mind, however, that this will have two consequences: It will raise the acoustic noise floor of the system and the capability for full-range signals will remain the same. However, some amplifiers will clip before the signal processing in the system.
This is another situation where you must accept a compromise or change amplifier sizes to get a better match in gain and capability between the different frequency bands.
Other Band-Limited Devices
There is a more general case, similar to the crossover scenario. If you have full-range signals at the input of a device that limits the frequency response—such as with high or low pass filters—there will be an energy loss from its input to output.
Calculate this loss using the same procedure outlined in the previous section on electronic crossovers. The significant energy of full range music signals effectively spans about 9 octaves (approximately 30 Hz to 15 kHz).
Example: An under balcony system band limited from 150 Hz to 5 kHz.
1) Multiply the lowest frequency limit of the device by 2 until you get to the highest frequency limit for the device. The number of times you multiplied = the number of octaves. Round off the results to the nearest whole octave [= 5].
2) Divide the number of octaves by the 9 full-range octaves [= 0.56].
3) Push the LOG key for this result [= -0.3].
4) Multiply this result by 10 to find the approximate loss [= -3 dB].
Now you must draw a horizontal line on the output side of the device’s window. The line is drawn at a distance below the top of the window equal to the loss in dB as found in #4 above. This line is used to match up the device’s window to the top of the window of the following device.
The purpose of a system limiter in a properly gain structured system is to prevent any signals from exceeding the system’s maximum level. As such, it is used as an “emergency” device meaning it is intended to provide a hard, never-to-exceed maximum output level.
Limiter/compressors with soft-knee thresholds are not as ideal for protection. You really want something that doesn’t do anything up to a certain point then stops any further increase cold in its tracks. Because you need some margin between the devic dynamic range (30 dB above the noise floor to the clipping level). The result is 42 dB.
Measurements of the maximum dynamic ranges for acoustic instruments and voice yield maximum figures in excess of 40 dB.
This means our system really doesne’s maximum input and its limiting threshold they are a bit tricky to implement properly without compromising the system’s dynamic range.
Just as with any other device you must introduce the limiter using its input, output, noise floor specifications, and gain setting just as with any other device in the system. Because of the way they work, the threshold setting is used as the maximum input.
For proper functioning the threshold should be set to at least 3 dB lower than the maximum output signal from the device preceding it. The limiter’s output gain should be used to adjust its maximum output at threshold to about 2 dB below the input level of the device it feeds. This allows a little “margin for error” in the protection.
If you think about it, the only point you can put a limiter in a properly gain structured system that will truly work perfectly is at the output of the signal source. It would be set so that the input level to system would never allow the first device it feeds to clip. This is technically practical if only one signal source is used at a time (i.e. not mixed with others).
Therefore, if you are simply switching between multiple input sources, put your limiting device at the output of the switcher. The system sees one input source and really doesn’t care which one it is and no input signal can drive the system into clipping.
With multiple mixed sources and a properly set system gain structure, the next best place to put the device is at the output of the mixer. This is because any mixer output voltage at any frequency exceeding the system’s maximum output will clip the system somewhere.
With multiple branch systems you might think to use a limiter on each branch. But with proper gain structure they would either all work at once or compress some part(s) of the system and not others, thus upsetting the acoustic balance. Thus a single limiter for each main output that is controlled by the operator makes the most sense.
“Controlled by the operator” means outputs such as a separate sub-woofer output where the acoustic balance is actively “mixed” by the operator based on the input signal content. You should make sure the operator can see when a threshold is exceeded to avoid clipping the mixer.
If the noise floor of the mixer is low enough compared to the other devices in the system you can allow more than the 3 dB margin the mixer has above the limiter’s threshold. You can do this by reducing a pad between the mixer and the limiter. You can also lower the limiter’s threshold level (and increase the output of the limiter by the same amount) if the noise floor of the limiter allows this.
Because it is used as a “hard-line” device, you should set the compression ratio to maximum (10:1 or higher if available). As to any attack and release settings, they do not affect the gain structure. However, as the limiter is intended to function only as an emergency protection device, there is every reason to use the fastest attack and release times. You are not going for sound quality here; you are protecting the system from any overdrive.
So get into and out of protection as fast as possible. If you think sound quality IS important, then you are not thinking correctly. What you should have thought about is a more powerful system that would rarely be pushed into limiting. In other words, if the system is constantly pushed into limiting, it is under-designed.
Gain structure is only a problem because invariably we use equipment with different input/output capabilities and noise floors. There is no easy way to properly set system gain except to analyze each device in relation to the device that feeds it and in relation to the entire signal path.
Fortunately, the little information you need is readily available on equipment specification sheets. By working out proper gain structure on paper before you purchase and wire up the equipment, you can spot potential problems and make appropriate substitutions.
In any case, with the gain properly structured, you can make significant, or in some cases, spectacular improvements in the system’s dynamic range and its noise floor. In the example system, 18 dB is certainly a spectacular improvement.
Monday, May 19, 2014
Prism Sound Atlas Interfaces Assist With Gavin Harrison’s Drum Workshop Masterclass
Used in recording performances of those taking part in worship
Noted drummer Gavin Harrison, who has worked with a wide range of artists including Iggy Pop, Sam Brown and King Crimson, recently held a drum workshop at Ananda Studios in Cambridge (UK) where he shared secrets of creating a great drum take, with recording achieved via dual Prism Sound Atlas audio interface units.
After starting the session with some basic coordination exercises, Harrison explained the drum part in a specific song and described the techniques to deliver and capture the sound. Afterward, those taking part were given the opportunity to play the drum part on his kit and have it recorded in Logic by Ananda Studio engineer Neil Cowlan so that they could review their performance.
“The opportunity to play along with Gavin and then have him critique my performance was great,” says Russ Tarly, one of the 24 drummers to attend the workshop. “This is the first time I’ve come across such a hands-on event and it worked really well. I knew I wanted to do it because it was so specialized. I’d love to see more events like this because I’m sure there are plenty of other drummers who would enjoy going through this experience.”
Cowlan deployed two recently acquired Prism Sound Atlas audio interfaces, which were used to track nine microphones from the drum kit into Logic. Atlas incorporates the company’s latest CleverClox dual-hybrid clocking technology as well as eight of the microphone preamplifiers as standard. The mic preamps are based on the successful Orpheus preamps, but are upgraded with 20 dB pads for all inputs.
Atlas provides analog and digital I/O for Mac or Windows PC at sample rates up to 192 kHz. In addition to the USB host interface, it also includes the new MDIO interface expansion slot, which was first incorporated into Titan. Using this miniature expansion slot, users can, for example, directly connect to Pro Tools|HDX systems and can also run with Apple and Windows native applications over USB. Other expansion cards are slated for future expansion of computer interface options. Prism Sound also loaned Ananda Studios a Lyra audio interface for additional monitoring.
The microphones used for the project included a Shure SM91 and AKG D12VR for the kick drum, a Shure SM57 (top) and a Beta 57 (bottom) on the snare, an AKG C451 on rack toms, and a Shure SM57 on floor toms.
Cowlan says: “The Prism Sound Atlas interfaces performed flawlessly, both during the Gavin Harrison workshop and at a private seven hour session that took place the day before. They are truly amazing AD converters and mic-pres that sound fantastic. It’s no surprise that Prism Sound units are found in some of the finest recording studios in the world and the fact that we can now offer them, too, gives people another great reason for coming to record here.”
Located in the Cambridgeshire village of Littlington, Ananda Studios is a purpose built facility housed in the old Neve factory. Originally designed by Andy Munro, the studio was once used for R&D purposes and to demonstrate Neve consoles. Today, it is popular with a variety of artists who appreciate tranquil surroundings, great room acoustics and an extensive equipment list.
“The live room is a fully acoustically designed space that sounds wonderful for anything you can imagine recording. Drums, string quartets, guitars, vocals, percussion – it all sounds great,” Cowlan says. “We also have a large range of microphones, pre-amps, EQ’s and other equipment to cater for almost any project, along with a huge collection of guitar and bass amplifiers including Marshall, Fender, PRS, Orange and Matamp.”
Following on from the success of the drum workshop, Cowlan plans to host similar events at the studio and will be announcing these very soon. Prism Sound sales director Graham Boswell, who attended the workshop with Bernhard Nocker from the company’s sales and marketing team, adds: “We were delighted to be able to support this event by loaning the studio equipment and look forward to working with Neil again in the future.”
Friday, May 16, 2014
Focusrite Announces Saffire PRO 26 FireWire/Thunderbolt Compatible Audio Interface (Includes Video)
Record at 24-bit/96kHz with four Focusrite preamps, 18 in/8 out, using FireWire or Thunderbolt
Focusrite announces Saffire PRO 26, the latest addition to the Saffire PRO range of FireWire/Thunderbolt compatible audio interfaces for expanding recording and live system capabilities, housed in a portable, desktop-sized chassis.
Saffire PRO 26 provides an extensive selection of professional analog and digital I/O options —a total of 18 inputs and eight outputs includes four preamps, two instrument inputs, two headphone outputs, six line outputs, and ADAT and S/PDIF connectivity.
Saffire PRO 26 connects to a Thunderbolt port via a FireWire to Thunderbolt adaptor (not included) or directly to a FireWire 800 port with the cable provided. Its dual-protocol compatibility (Firewire and Thunderbolt) means it will work seamlessly for years to come with the next generation of computers.
The four Focusrite preamps provide a great deal of recording flexibility while also ensuring low noise and distortion with plenty of headroom to capture the full dynamic range of even a loud drum kit or guitar amp. Precision 24-bit/96-kHz digital conversion and JetPLL jitter-elimination technology maintain pristine audio quality in both analog and digital domains.
Saffire Mix Control, a control software application that runs on the host computer is particularly useful for live situations. The low-latency 26 x 8 DSP mixer/router provides flexible output routing and monitoring for custom monitor mixes as well as intuitive one-click presets to help in setting up sessions as quickly as possible, whether tracking, mixing or monitoring.
Although Thunderbolt provides some advantages, it’s only available on the very latest computers. An audio interface fitted with FireWire can be used on both older computers and Thunderbolt-equipped computers via an inexpensive adaptor.
ADAT optical input allows extensive input expansion of up to eight additional preamps or line inputs. Pair Saffire PRO 26 with Focusrite’s OctoPre MkII eight-channel preamp and instantly transform it into a 12 preamp unit.
Included is a free DAW in the form of Ableton Live Lite, Focusrite’s professional Midnight and Scarlett plug-in suites, Novation BassStation virtual synthesizer and 1 GB of Loopmasters sample content.
Thursday, May 15, 2014
Core Brands Debuts Furman P-8 PRO C Power Conditioner
Offers advanced power management for audio, video and broadcast applications
Core Brands has introduced the Furman P-8 PRO C power conditioner, a 20Amp model joining the Classic Series that includes proprietary Furman technologies such as Series Multi-Stage Protection (SMP) and Linear Filtering Technology (LiFT).
Housed in a rugged 1RU chassis with a minimalistic front panel and nine outlets, the P-8 PRO C is designed for power conditioning applications where front panel metering and illumination are not required.
SMP surge protection circuit technology safely absorbs, clamps and dissipates transient voltages, eliminating service calls.
Meanwhile, the device’s over-voltage circuitry (EVS) protects against accidental connections to 208 or 240 VAC by shutting off incoming power until over-voltage is completely corrected.
LiFT provides filtration against excessive AC line noise, helping to ensure optimal performance without any leakage to ground.
“Our new P-8 PRO C classic series power conditioner for audio/video professionals builds on the success of Furman’s renowned P-8 PRO II,” says John Benz, director of power and accessories for Core Brands. “Using our power protection and filtering capabilities, the P-8 PRO C features our venerable Series Multi-Stage Protection circuit as well as Furman’s exclusive Linear Filter Technology, creating advanced and comprehensive transient voltage suppression and conditioning for any professional A/V application.”
RedNet Powers AV Operations At 2014 FIFA World Cup Final Draw
Recently, Brazilian outside broadcast (OB) company Mix2Go invested in multiple RedNet units as part of their wider digital audio/visual network for the broadcast and recording of the 2014 FIFA World Cup Final Draw event.
Focusrite’s RedNet range of network audio interfaces are playing a key role in the live broadcast and recording of some of Brazil’s biggest entertainment events.
Recently, Brazilian outside broadcast (OB) company Mix2Go invested in multiple RedNet units as part of their wider digital audio/visual network for the broadcast and recording of the 2014 FIFA World Cup Final Draw event.
Like most professionals working in live sound, remote recording and broadcast, sound engineers Daniel Reis and Beto Neves from Mix2Go have adopted Audinate’s Dante platform for all their networked audio.
Their OB truck features a Dante-enabled Allen & Heath mixing console and stage racks, which route audio from the stage to any device connected to the Dante network. This enables the team to route signals to a multitude of different recording, mixing and live-broadcast systems simultaneously, while giving them the opportunity to upgrade and expand their network, simply by connecting more Dante devices to their gigabit switches.
Daniel is no stranger to Dante technology, or to the sound quality that Focusrite gear provides. “I started using Dante back when the Dolby Lake loudspeaker management system came out, and was amazed by how it gave you access to a whole new world. I’ve also used a lot of the Focusrite ISA products, so when I heard about RedNet, it was great news: finally a company committed to sound quality had invested in the best protocol available.”
Mix2Go use a RedNet 2 16-channel A-D/D-A interface, the RedNet 3 32-channel digital I/O, a RedNet 6 MADI Bridge and three RedNet PCIe cards.
“I need to give the option of delivering the program through MADI, AES/EBU or analog,” says Daniel.
For the FIFA World Cup Final Draw, the team decided their RedNet 2 was an excellent choice to deliver the final live mixes to the video OB truck. They also used their RedNet 3 as an additional monitoring reference via the AES/EBU outputs.
Incorporating the RedNet gear into their somewhat complex setup was incredibly fast and simple, thanks to the straightforward nature of the Dante network and the rock-solid reliability of the RedNet interfaces.
“Installing RedNet was very fast – plug-and-play – and almost idiot proof,” Daniel recalls.
One of the main benefits of using RedNet, and the wider Dante network, is its ability to send large amounts of audio over long distances with minimum cabling. The setup for the Final Draw event consisted of a large Dante network that linked two OB trucks parked in different locations outside the venue to the main stage. T
wo stage boxes (one primary and one backup) were located in the main arena, connected to gigabit network switches using Cat-6 cables. Signals from this stage rig were sent via multi-mode fiber cables to another pair of switches inside the music OB truck, about 150ft (50m) away. Inside the truck, the fiber signal was converted back to Ethernet and routed to two Mac Mini computers, which handled the multi-track recording, using Pro Tools and Audinate’s Dante Virtual Soundcard (DVS) driver.
The signals from stage were also routed simultaneously to the live mix console, from where the final broadcast stems were then routed to RedNet 2, located inside the video OB truck, about 100ft (30m) away. Here, the stems were converted to analog then mixed with the rest of the event’s audio and added to the final video feed, which was then sent to a distribution truck to be broadcast to television stations all around the world.
After the event, Daniel was extremely happy with how the RedNet equipment performed. “The converters sound transparent and tight. RedNet has truly given us reliability and quality, and pride to have such amazing equipment from a brand that we truly respect.”
Posted by Julie Clark on 05/15 at 10:57 AM
Monday, May 12, 2014
Riedel Communications Expands Marketing & Communications Team With New Assignments
Christian Bockskopf promoted to head of marketing; Serkan Güner joins company as new marketing and communications manager
Riedel Communications has announced that marketing manager Christian Bockskopf has been promoted to head of marketing, with Serkan Güner joining the team to serve as the new marketing and communications manager.
“Christian has done a remarkable job handling various marketing and communications roles over the past years,” states Thomas Riedel, president at Riedel Communications. “This new role is the direct result of his unwavering ability to take on any challenge as he builds on his 10 years of experience at Riedel.”
Güner will report to Bockskopf in his new role as marketing and communications manager, where he will develop press and marcom opportunities for Riedel Communications while implementing the company’s sales plans on a global scale.
“Serkan’s remarkable breadth of experience in the field of marketing and communications is complemented by his facility with diverse languages and cultures, and this combination of skills and knowledge will be valuable as we continue to build brand awareness and expand our global footprint,” says Riedel. “We are very pleased to welcome him to the company.”
Prior to joining Riedel Communications, Güner served as international marcom manager for a German manufacturer of pro A/V system products. In this and in preceding roles in marketing and communications, Güner has developed expertise in coordinating and implementing all aspects of marketing and press campaigns, from traditional releases to engagement across social media platforms. He is fluent in German, Turkish, English, Italian, and Spanish.
Friday, May 09, 2014
Monitor Engineer Brian Montgomery Utilizing SSL Live Console For Santana
Montgomery was introduced to SSL Live by Rob Mailman, FOH for Santana and GM of touring for Sound Image, which supplied the console
An SSL Live console is on the road with iconic Latin rocker Carlos Santana, whose perennial monitor engineer, Brian Montgomery, is using it to mix both stereo wedges for Carlos and IEMs for his band.
Following last winter’s residency at Mandalay Bay’s House of Blues (HoB), Santana has been globetrotting, making stops at the Dubai Jazz Festival, Johannesburg’s FNB Stadium, Cape Town’s Grand Arena and the New Orleans Jazz Festival as part of its Corazon tour. Santana and his crew will then join Rod Stewart for The Voice, The Guitar, The Songs tour this spring and summer.
Santana is the winner of nine Grammy Awards – the most for a single project – for Supernatural, which revitalized his career, leading to the first Las Vegas Hard Rock residency and his ongoing HoB appearances each winter.
An industry legend in his own right, Montgomery has mixed Santana’s monitors for 16 years. “I was an analog guy for a long time, but, it got to where I’d actually outgrown most of the analog consoles on the market,” he explains. “I was using a pair of competitor consoles to get 26 mixes, and I needed more space. When we went to Japan, the model I’d been using wasn’t available, so we switched to another competitor, which I’ve used as my primary console for Santana ever since.
“That is,” he adds, “until the SSL Live came out. I’m an old school guy and I like that SSL has an additional focus channel screen, which gives me all of the control that I need, alongside the main screen. This setup lets me have multiple modes of accessibility to my work surface, which is key.”
Montgomery was introduced to SSL Live by Rob Mailman, FOH for Santana and GM of touring for Sound Image, which supplied the console. “I’d been looking to switch consoles, so when Rob started telling me about the way SSL Live’s control surface was laid out, I was extremely interested,” explains Montgomery. “Once I finally got to see the console, it drew me right in. I actually never heard the console before I decided to use it.
“I know what an SSL sounds like in the studio; they’re sonically incredible. For me, it’s mostly about design, use, accessibility and how I could lay the desk out for what I need to do. I don’t have any static mixes with the band, every song is different, so there’s never quite the same setup each night. I have to have a console that I can get around on very rapidly.”
With eight stereo wedge mixes across the front of the stage, Montgomery has them set as post-fade mixes that are tapered, so when Carlos is near somebody, the sound is turned down a little to give him an even mix across the stage. With this setup, Carlos can go anywhere and still hear what he wants or move away from sounds that he’s not interested in hearing.
“I’ve always liked to have Carlos’ setup accessible through post-fade on my surface so that I can do anything for him at any point in time, even if I’m messing around with the mix for somebody else in the band,” continues Montgomery. “Most of the band members are on in-ear monitors, so I need the versatility to mix Carlos while I’m changing the IEM feed for another band member. With many other digital consoles, that’s difficult to do because you have to stop what you’re doing for one performer to do something for another. With the SSL Live, I don’t have to do that. I tap the screen for the mix and I can change something for someone else rapidly without popping through four different layers.”
Montgomery first used SSL Live at HoB Las Vegas during a sound check. “I did a line check with the backline members and I started to notice how sonically I didn’t need to EQ as much,” he says. “The preamps sound very nice, very natural, no additional high-end, no tin that I often hear out of other digital consoles, very warm and very forgiving. With our current set, which is very loud, I might get plenty of input gain as well as a lot of output. All in all, with the SSL Live, I don’t have to EQ vocal mics to avoid feedback. It gives me a good natural sound and, regardless of the mic placements, I don’t have drastic EQ changes on the instruments.”
After Montgomery started using the SSL Live, every band member was impressed with its quality. “They all came to me and said it sounds brilliant,” he concludes. “It was something that kind of threw them because it was very natural sounding. Over the years, a lot of them have gotten used to something that was a little brighter or a little thin sounding. No matter what you did to the midrange, you had that digital imperfection. Now that I’ve switched to SSL Live, all of the musicians have come to me and said that sonically they’re very pleased with how it sounds. With a lot of my ear mixes, I don’t have to drive the units as hard. The musicians actually have to turn their packs down and I don’t have to send as much signal.”
Solid State Logic
Simplicity Rules: A Well-Considered Sonic Approach For Broken Bells
Technically, Broken Bells is a duo, a Los Angeles-based indie-rock band comprised of Brian Burton (also known as Danger Mouse) and The Shins lead singer/guitarist James Mercer who first conceived the idea of doing a project together after meeting at the Roskilde Festival in 2004.
On stage, however, the band is rounded out by multi-instrumentalists Jon Sortland (drums, keys, bass) and Dan Elkan (guitars, keys, bass), and all of players routinely take over other instruments or work one of four keyboard rigs.
So there’s a lot going on, but the group’s mix engineers have endeavored to keep the audio approach as straightforward as possible. I caught up with London-based Dave McDonald (front of house) and Chicago-based Steven Versaw (monitors) prior to a show on the recent tour at Toronto’s Danforth Music Hall. McDonald told me that the old school theatre, now converted to a live venue, reminds him of the Ritzy Picturehouse in Brixton, UK.
Broken Bells are currently on the road in support of their sophomore full-length album, After the Disco, with the tour slated to stretch well into October. Right now they’re playing a variety of theatres and clubs and aren’t carrying loudspeakers with the exception of wedges – and fewer of those as time goes on.
Dave McDonald at the Allen & Heath iLive-112 mix surface at front of house. (Photo credit: Jordan McLachlan)
“I’m trying to phase the wedges out,” McDonald notes, “because a clean stage is a wonderful thing for an engineer, so we’re in the last stage of the clear-out. Brian’s the only one using wedges at the moment, L-Acoustics 115XT HiQs, which are lovely.”
It’s the first time out with this band for both engineers. Burton had tried to get McDonald onboard for the previous tour but he was already out with Adele, while Versaw signed on after five years as monitor engineer and production manager with Wilco. Both desire simplicity as a general practice, and in particular given the nature of the current assignment.
McDonald chooses to mix on an Allen and Heath iLive-112 digital mixing surface – which he praises for its ease of use, compact footprint and accuracy – combined with an iDR10 MixRack.
Broken Bells in action on the latest tour. (Photo credit: Jordan McLachlan)
Right now, he’s working solely with the mix surface rather than employing a tablet or laptop, with the addition of a Dante card delivering up to 64 channels to a recording rig. He’s also giving the console’s effects a considerable workout in emulating the band’s recorded sound in the live realm.
“There’s so much going on with the vocals and so on. It’s very ‘Beatles-esque’ and psychedelic, but the iLive has all that I need onboard,” he told me as he fired up the console for that night’s show. “Sometimes when you add effects you’re struggling to hear them, and it’s not quite right. With the iLive, they’re very accurate,” adding that the Symphonic Chorus, ADT Doubler and EkoChorus plug-ins have proven particularly useful.
“When I went to rehearsal,” McDonald continues, “they brought a box of outboard gear and I opened it up and said ‘That’s nice.’ And then I closed the lid and put it back in the truck,” he laughs, “and that was that.”
The iLive iDR10 MixRack on stage playing a pivotal role in signal routing.
One of the potential complexities of the gig is the number of keyboard stations on stage and the fact that the players move from instrument to instrument, but Versaw takes that in stride. And, rather than isolate or turn the band’s amps away from them, he prefers to preserve the feel on stage by taking “an additive approach to the IEM mix.”
All keyboard sounds are triggered from an Ableton rig hosted by two Mac pros located at monitor world and patched directly into a splitter that goes to monitors and FOH.
“For the IEMs, I’ve created a general drum mix with kick and snare at 100 percent and rack and floor at about 40 percent,” Versaw adds. “That’s being sent to a group that I’m squashing a bit to provide a gritty, even sound. Then I pepper in my overheads by themselves to make it more live.
“When Brian plays the drums, it’s lighter than Jon, so I add in some non-compressed snare in the ears,” he says. “The group mix is sent to everyone as the drum mix, and then I can add in individual drums if necessary.”
Currently Versaw is mixing on an Avid VENUE Profile, chosen primarily because of his familiarity with its interface and workflow. “Both Dave’s iDR10 MixRack and my Avid Stage Rack are on stage. I have a splitter that feeds all inputs from stage to the Profile and sends to Dave’s iLive rack via a Cat-6 line. So he has a control surface out at FOH, and that’s it.”
Like McDonald, Versaw uses only onboard plug-ins. “Nothing external. When it comes down to it, I need EQ, compressors and a few gates. The rest of it is aesthetics,” he says. “I’m using reverb to add ambience to the backing vocals, and some delay – typically for James’ vocal – to recreate a slap back effect from the record, but that’s only for me and James.
“It’s a sweet and simple setup and it’s only going to get simpler,” he continues. “I actually plan to move over to an iLive because of its simplicity, and also so I can work more on my mixes and less on setting up. Simplicity rules.” He adds that he’s happy to switch platforms and technology when necessary to keep the system streamlined and his workflow fluid. “You use your ears and your intuition, and if something’s hard to use or you can’t figure it out really quick, then try something else.”
Blend Of Components
A primary goal is to have as little sound coming off stage as possible, with all band members on Ultimate Ears IEMs, primarily UE11s. The gear the tour is carrying is a blend of components supplied by Rat Sound Systems (Camarillo, CA) along with equipment drawn from Burton’s studio and The Shins touring rig, including Sennheiser EW 300 IEM G2 wireless monitoring systems and some of the microphones.
Monitor engineer Steven Versaw at his Avid VENUE Profile.
“When it comes to mics it’s very straight up,” McDonald says. “In rehearsals, when a guy starts playing his guitar I’ll stand in front of it for a few minutes and see what the guitar’s saying and what he’s saying, and then put a mic in front of it. If it doesn’t sound like that at FOH, we move the mic or change it. It’s simple – there’s no magic.”
Two Sennheiser e 902 dynamic cardioids are applied for kick out and floor tom, with a Sennheiser e 901 dynamic cardioid for kick in. Neumann KM 184s cardioids are deployed for hi-hat and overheads. “The e 902 on the floor tom provides depth,” Versaw says, “but apart from the Neumann microphones on overheads and hi-hat. it’s really traditional. We’ve got a Shure SM57 on snare top from The Shins’ locker and clip-on Sennheiser e 604s for snare under and rack toms.”
The stage also hosts several Radial passive DIs as well as SM57s on guitars – a Fender and Marshall combo located center stage for Mercer and stage left for Elkan, respectively. “With those mics, you can just light the fuse and run away,” Versaw adds. “And we went with SM58s for the vocals because James (Mercer) is accustomed to them.”
A partial view of the drum miking approach.
Less Is More
“I’m hauling a board, a brain and an engineer on this tour,” McDonald says as we were coming up on sound check time in Toronto. “I come from a world of where you would have the biggest board in the world and have as much outboard gear as possible, but what really matters at the end of the day is what’s coming out of the speakers.”
Speaking of which, the loudspeaker count began shrinking even before the tour kicked off. Stacks of side fills with subwoofers were originally specified but only made it through pre-production, although they actually served as the FOH system in a makeshift rehearsal studio in Portland.
“We had rehearsals in LA and then just before the tour kicked off, we brought in all the elements – Dave at FOH, as well as lighting and video systems – into a raw loft workspace and built our show,” Versaw adds. “Initially, I specified components to suit all occasions, but we’ve never brought the full rig in for a show.”
Based in Toronto, Kevin Young is a freelance music and tech writer, professional musician and composer.
Thursday, May 08, 2014
Soundcraft Releases Si Expression/Performer/Compact Software Update With BSS Soundweb London Control
Also provides external clocking via option cards, selectable input solo modes, select follow solo, pre-dynamics global bus send and more
Harman’s Soundcraft website announced that new Si Expression/Si Performer V1.6 software and Si Compact V3.1 software is available for download.
The software updates offer several new features and improvements to Soundcraft Si Series consoles, including BSS Soundweb London preamp control, external clocking via option cards, selectable input solo modes, select follow solo, pre-dynamics global bus send, knob bubble pin, and momentary on.
With the V1.6/V3.1 software update, Si Expression, Si Performer, and Si Compact have new features specifically geared to improve performance for networked audio and monitoring applications. All console software updates can be accessed from the appropriate product page on the Soundcraft website.
In addition, updated user guides for the Si Dante Card, V1.6/V3.1 software update, and Mini Stagebox 32 are also now available from the Soundcraft website.
Additionally, supporting how-to videos on the V1.6/V3.1 features are now available on the Soundcraft YouTube channel.
Wednesday, May 07, 2014
Belfast (Ireland) PA Company Wackiki Deploys Allen & Heath GLD For Arenacross UK Tour
GLD-80 control surface and AR2412 remote I/O rack for motocross tour that visited seven arenas from Belfast to Wembley
Belfast (Ireland) PA company Wackiki recently employed its Allen & Heath GLD digital mixing system, comprising the GLD-80 control surface and AR2412 remote I/O rack, for the Arenacross UK tour, which visited seven arenas from Belfast to Wembley in showcasing an international line-up of pro racers and Freestyle motocross.
“We chose GLD because of its small footprint and I/O processing abilities,” says Steve Woods, hire manager for Wackiki. “The use of compressors, high-pass filters and 4-band EQ on each channel was a must. With some venues posing difficult cable runs, we liked that the GLD only required a single Cat-5 to connect the stage box, instead of multiple cables like most desks.
Woods adds, “The truck pack was also very tight with the amount of truss and PA involved, and the only way the tour budget would work was if all the sound and lighting production fit into one truck. The GLD 80 offers a small case for the control surface, a stage box that fit into the RF rack and a single Cat-5.”
Wackiki also utilized the GLD’s I/O in the back of the control surface for handheld radio mics and IEM systems utilized by race officials and announcers. “Having the radio mics in view of the operator was necessary to monitor the RF signal strength because of the long distances involved with announcers on the opposite side of the arena,” Woods notes. “The iPad software was also key because of the mix position. The operator was often behind the PA where it was difficult to tell what the PA sounded like to the audience.”
Allen & Heath
Tuesday, May 06, 2014
The Great Beyond: Expansion Options & Accessories For Digital Consoles
Digital consoles are also referred to as “mixing systems,” and it’s an appropriate moniker because in addition to supplying control surfaces with a lot of onboard functionality as well as I/O, they can be further expanded in a variety of ways with the addition of “outboard” capabilities such as offline editing stations, remote apps, interface cards and modules, and of course, digital snakes, networking, and personal monitoring.
Offline editing provides the ability to set up patching, scenes and other parameters on a computer and then upload these settings to the console. Many programs also allow remote control of the console, and can also be used as an additional display screen when the console is in use.
Remote apps vary in scope and focus but offer the ability to remotely access most—if not all—key functions. And when at the console, the app on the tablet can also be used as an additional display device.
Many consoles are equipped with slots for interface cards providing extra I/O with a variety of analog and digital connectivity, along with extra processing like auto-mixing and program leveling as well as a conduit to integrate personal mixing devices and recording units. Some have external modules that foster interconnection of various digital protocols and processing.
Consoles all have a least a few “local” inputs and can be used as stand-alone, as well as digital snake links to accommodate one or more stage boxes offering an impressive amount of I/O. The boxes can be located at the mix position, on stage, or anywhere else that is best for the particular application.
In addition to digital snakes, digital networks can be used to provide source inputs to more than one console (i.e., FOH and monitor), broadcast feeds, and send/return to recording devices.
We can all appreciate the fact that digital consoles are usually smaller and lighter than their analog predecessors, with the “modular” nature of these packages furthering that cause while also delivering greater capability and flexibility. For a solid roundup of what’s available, take our Gallery Tour of what’s available with recent models.
Senior contributing editor Craig Leerman is the owner of Tech Works, a production company based in Las Vegas.
Extron Shipping 2-Channel Balanced/Unbalanced Audio Converter With Input Signal Matching
Provides two independent channels of signal matching between unbalanced and balanced audio equipment
Extron Electronics has announced the immediate availability of the BUC 202, a 2-channel balanced and unbalanced audio converter and audio line driver.
The BUC 202 includes two balanced or unbalanced inputs and outputs, both on captive screw connectors, and provides two independent channels of signal matching between unbalanced and balanced audio equipment. Two rotary switches provide precise trim adjustment from -21 to +21 dB in clearly-labeled 3 dB steps.
In addition to serving as an interface from consumer products to professional AV systems, the BUC 202 is a high-quality line driver for sending and receiving audio signals up to 1,000 feet (300 m). The unique 1U, quarter rack width enclosure, together with the included patented ZipClip 200 mounting bracket, offers the versatility to install the BUC 202 in a discreet location, such as on a rack rail, beneath a table, or within a lectern.
“The BUC 202 demonstrates Extron’s continued dedication to offering best-in-class audio ‘problem-solver’ products,” says Casey Hall, vice president of sales and marketing for Extron. “It features several new design enhancements with unique installation features and refinements in audio performance, enabling high quality integration of consumer audio products with professional systems while being mounted in a hidden location.”
The BUC 202 is designed for typical applications where a laptop is connected at the lectern and the audio needs to reach a remote equipment rack. It can be easily hidden by mounting the ZipClip 200 to any surface inside the lectern, while the unique form factor of the audio converter allows installers to snap it securely onto the mount.
For added convenience, detented level controls on the BUC 202 enable precise source input matching and gain structure. It also includes an Extron-engineered, energy efficient internal power supply that reduces operating costs and provides a long operational lifespan.
Riedel Artist Intercom Delivers Flexibility For The House of Dancing Water
One Riedel Artist 64 frame and two Artist 32 frames support 62 panels distributed across the Dancing Water Theatre
A Riedel Communications Artist digital matrix intercom system is being used for communications throughout The House of Dancing Water, a breathtaking in-the-round water show created by the Franco Dragone Entertainment Group for the Macau (China) City of Dreams entertainment complex.
“The Riedel solution was originally specified by the sound designer, Francois Bergeron of the Thinkwell Group in the U.S.,” says Jason Graham, head of sound at Dragone Macau Limited. “We continue to work with Riedel’s Artist primarily for its reliability, stability, and seemingly limitless flexibility. All of these factors are important due to the integral role the system plays in uniting everyone who is working on the show.”
Designed by Pei Partnership Architects, the state-of-the-art Dancing Water Theatre created for The House of Dancing Water includes a stage pool that holds a record-breaking 3.7 million gallons of water, equivalent to five Olympic-sized swimming pools, within an arena boasting a 40-meter-high steel-trussed space that provides the generous height required for the show’s diving and acrobatics elements. The show itself centers on an epic love story and spectacular journey through time, showcasing dazzling costumes, special effects, and performances.
One Riedel Artist 64 frame and two Artist 32 frames support 62 panels distributed across the Dancing Water Theatre. The motocross stunt riders and acrobatic performers, coupled with the technical teams, use the Artist intercom system for continual communication during the show, which features visuals, water, and atmospheric effects. Even in the aquatics area, the Artist system supports communications with performers and underwater performer handlers in the pool via underwater speakers, in-mask communications systems, and buddy phones. Lighting, stage management, rigging, automation, sfx, and audio technicians use the remaining bulk of the panels.
Riedel Director software enables intuitive management and configuration of the system while also facilitating real-time system monitoring from multiple computers. Monitored by the sound department, the Director software gives the sound team the ability to respond to configuration requests and to perform troubleshooting at a moment’s notice.
“Our Artist intercom system delivers the exceptional versatility and performance required in the most demanding and complex live production environments,” says Hans Chia, rental operations manager at Riedel Communications. “The House of Dancing Water is no exception. The Artist provides the clear, reliable communications that are essential in orchestrating this ambitious live show.”
Monday, May 05, 2014
Road Test: 10EaZy From SG Audio
Evaluating an innovative new SPL measurement platform, distributed in the U.S. by the "Smaart" folks at Rational Acoustics...
If you need a class compliant sound pressure level (SPL) monitoring and logging system or just want a great way to keep track of SPL at gigs—keep reading. SG Audio of Denmark has developed an SPL measurement system called 10EaZy, and the folks at their exclusive U.S. distributor, Rational Acoustics, were “Smaart” enough (see what I did there) to send me a system for review.
SG Audio offers measurement systems designed to meet the needs of those who require IEC/ANSI Class 1 or Class 2 compliance, as well as a basic system for those that do not require compliance but still want a full-featured logging SPL rig.
This is a particularly timely series of new products because, increasingly, venues and municipalities are establishing SPL limits for concerts, events, and businesses (think manufacturing). Handheld portable meters, laptops with measurement software, and certain smart phone apps can do basic SPL measurements, but they may not be entirely accurate.
Also, most of these options don’t offer a means to log the data over time or offer the user an easy way to archive any data. To get reliable results, especially if you need to be compliant with local codes or laws, higher quality equipment is required, and further, the entire measurement chain should be properly calibrated.
A 10EaZy Class 2 system.
10EaZy offers a turnkey solution by providing tamperproof hardware and a measurement microphone that are calibrated as IEC/ANSI compliant, combined with easy-to-use software that offers a host of features. Systems are available in four versions: Class 1 compliant, Class 2 compliant, RT (Class 2 compliant with a reduced feature set), and SW (a software- and dongle-only system that requires users to provide their own quality measurement microphone, I-O, and calibrator).
The differences are as follows. Class 1 and Class 2 systems offer all the same software features but are tailored to the different classification of measurement specifications. RT and SW, the reduced feature-set versions, do not offer a running order, an event log, or a minute-by-minute resolution logfile for post processing of measurement results. However, they do provide a file, listing a compilation of key measurement results. And given the variability of the hardware that can be used with the SW dongle, measurements made using the SW version cannot be guaranteed IEC/ANSI compliant by the manufacturer.
Specifically, Rational Acoustics supplied me with a 10EaZy Class 2 system. It consists of a small (approximately 4.2 x .5 inches) measurement mic that comes with a nice, compact aluminum storage case, a 15-foot BNC-to-BNC mic cable, a tamper-proof plastic interface box (compact at approximately 5 x 3 x 1 inches), a 6-foot USB cable, and the software.
I noticed that the measurement mic wouldn’t fit any mic stand I owned, but I took another look in the box and discovered an Audix McMicro clip with a 3/8 - 5/8 threaded adaptor. Rational Acoustics also offers an upgrade kit for the Class 2 & RT systems which includes a sturdier mic clip with a 3/8 - 5/8 thread adaptor, a special 1/2-inch bushing to securely hold the 10EaZy mic, and a windscreen.
The next thing that caught my eye was that the mic sported a BNC connector instead of XLR connectors that I usually deal with. The cable that ships with the unit is 75 ohms, high-resolution/low-loss, and of very high quality. At 15 feet long it may be a little short for some uses but in my shop and at the gigs where I used 10EaZy, I was within feet of my laptop, so it wasn’t an issue. Per the manufacturer, a cable length up to about 250 feet can be used without a problem, if properly isolated.
Right out of the box, installation is straightforward. A CD is provided that will work with Windows systems XP and above, and with just a few clicks, the software is installed. The software then prompted me to plug in the hardware, and the system was all set.
To begin measuring, I simply set a destination for the log file and gave it a name. If you have a known target Leq limit and time period for the session (for example, 103 dBA for 3 hours), you can enter this upon start up. The software is very intuitive and easy to use.
Within a few minutes I was confident that the system was working correctly and started to do some testing, comparing 10EaZy’s readings to my usual handheld SPL meters (a mid-priced professional measurement unit) as well as a few SPL apps on my iPhone.
Using a steady 1 kHz tone, I compared my trusty meter to the readings of 10EaZy, and was pleased to find that it was within .5 dB of the laptop display. My iPad apps didn’t fare as well. One was off by 1 dB, while another was off by about 2.5 dB (To be fair, that’s still pretty good for a free app using a built-in mic on a mobile phone).
Next, I used 10EaZy to help check out some new powered loudspeakers that were just shipped to our shop. I ran a variety of tones through my signal generator into a Mackie 1604VLZ4 mixer and then into the loudspeakers, checking to see if any unit varied widely from the others. Happily, all were in proper working order, so it was time to crank up the music and see what the loudspeakers, as well as 10EaZy, could do.
10EaZy dedicated USB interface.
The display is very easy to read, large green letters against a black background. There are four buttons on the screen: Event Log allows you to add a time stamp with a simple click to the log; Running Order lets you add band names, playing times and duration to the log; History shows what’s been happening since the measurement session started, and it also allows you to change the plot and look to a variety of styles and colors; and finally, Full Screen toggles between normal and full screen views.
The MaM (Maximum Average Manager) is particularly interesting. This display shows you how much above or below you are from your target SPL over time, in 1 dB increments. What this means for a festival, for example, is that if one band engineer runs their entire set 4 dB above the target level, they effectively use up available loudness for the duration of the Leq. Other acts would have to run lower in level to even out the Leq and get back to your target level.
A closer look at a 10EaZy screen.
I took the system out to several gigs and basically used it the same way every time, setting up the mic and laptop at FOH and then using 10EaZy as a reference for the overall show volume. I found I could also run the program in the background and it would still log what was happening, and I could also run a music playback program at the same time. Nice.
The 10EaZy feature set is plentiful. Right now, with much of my present work focused on corporate shows/events, I’m rarely encountering (at least as of yet) the need for sophisticated SPL metering or logging, but for those working in the touring and festival sector (where they’ll be encountering increasing SPL restrictions), this is a great system to have in the toolbox. It’s also perfect for venues needing to comply with local noise ordinances.
The display is easy to read, and the logging will come in handy when a neighbor complains about the noise level. And because it’s a calibrated system, the data will stand up to governmental and organizational scrutiny.
U.S. MSRP: 10 EaZy Class 1 system—$2,793; Class 2 system—$2,360; RT system—$1,742; SW software dongle—$299; Class 1/RT system clamp & windscreen upgrade kit—$50.
For more information about 10EaZy, including in-depth descriptions of class compliance and the various 10EaZy systems, check out the U.S. 10EaZy website at 10eazy.us. And to purchase 10EaZy, go here.
To comment on this review and/or ask Craig questions, go to the PSW Road Test Forum.
Senior contributing editor Craig Leerman is the owner of Tech Works, a production company based in Las Vegas, and he’s also the moderator of the Road Test Forum here on PSW.