Audio

Tuesday, July 17, 2012

Armagh Theatre Installs New Allen & Heath GLD Console

The GLD mixer is comprised of a GLD-80 mixer plus an AR84 I/O rack in the control room, and an AR2412 rack down on the stage floor. The installation is part of a wider overhaul of the venue’s analogue audio infrastructure, which includes installing Cat5 around the building to allow connection of an expander I/O rack to the GLD-80 mixer.

Market Place Theatre & Arts Centre in Armagh, Northern Ireland, is one of the first venues in the world to install Allen & Heath’s new GLD digital mixer.

The GLD mixer is comprised of a GLD-80 mixer plus an AR84 I/O rack in the control room, and an AR2412 rack down on the stage floor.

Replacing an existing analogue desk, the installation is part of a wider overhaul of the venue’s analogue audio infrastructure, which includes installing Cat5 around the building to allow connection of an expander I/O rack to the GLD-80 mixer.

“We have a team of four technicians, and we all put forward names of desks we thought may be suitable, as well as consulting industry colleagues,” explains technician, Gary Bawden. “We demoed a few desks before making our choice but the GLD came out on top for price, ease of use and input capacity.

“Moreover, it sounded the best in our comparison listening tests.”

The venue hosts many different types of events, including theatre, music, dance, comedy and conferences, which the new GLD system will be managing.

“We’ve been really impressed with how easy it is to configure and move things around on the desk. Being able to colour code things is a neat feature. Also, the way this desk accesses auxiliaries is really good, almost a reverse logic of most other desks. The RTA is amazing and makes it so easy to squeak your monitors,” concludes Gary.

Allen & Heath

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Posted by Keith Clark on 07/17 at 06:43 AM
RecordingNewsPollConsolesDigitalInstallationStudioAudioPermalink

Vintage King Audio Opens New Showroom And Demo Studios In LA

Vintage King Los Angeles (VKLA) is displaying a variety of classic and current gear in the state-of-art facility, including eight analog and digital consoles, twenty pairs of monitors, fifty microphones, and dozens of preamplifiers and outboard processors.

Vintage King Audio has opened VKLA, a private Los Angeles showroom and demo studio on Sunset Blvd., to better serve pro audio professionals in the heart of the world’s biggest recording market.

“We’re extremely proud of VKLA and eager to give our clients access to this extraordinary facility,” remarked industry vet Tom Menrath, recently named head of VK’s Strategic Development. “VKLA provides an unprecedented opportunity to compare the finest gear in the world in a proper recording studio environment.

“Visits to the facility will be arranged by appointment, so each customer will get a personalized listening experience.”

Vintage King Los Angeles (VKLA) is displaying a variety of classic and current gear in the state-of-art facility, including eight analog and digital consoles, twenty pairs of monitors, fifty microphones, and dozens of preamplifiers and outboard processors.

Clients can test out nearly a dozen newer consoles such as the API 1608 and the SSL 948, as well as boutique gear from Barefoot Sound, Shadow Hills Industries, Neve, Dangerous Music, Burl, Telefunken, and other top industry brands.

“Our VKLA headquarters is really a new way of giving the customer the most enjoyable buying experience,” explained Shevy Shovlin, Director of Partner Marketing/PR. “We wanted VKLA to be musical, a place that working artists could appreciate. Our long-term goal is to create a real cultural center for the recording scene here in LA.”

VKLA goes beyond the familiar functions of a sales dealership by connecting with the professional audio community through hosting events, panel discussions, and product launches. Plans are in place for a series of educational workshops where seasoned pros will share their knowledge with a new generation of music makers.

Vintage King Audio

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Posted by Keith Clark on 07/17 at 06:28 AM
RecordingNewsPollConsolesEngineerStudioAudioPermalink

Ana Popovic Records In Ardent’s New SSL Duality Studio C

Blues sensation Ana Popovic, a "Best New Artist" W.C. Handy Award-nominee, is recording her next release at Ardent Studios on a Solid State Logic console in Studio C (Memphis, Tennessee).

Blues sensation Ana Popovic, a “Best New Artist” W.C. Handy Award-nominee, is tracking her next release at Ardent Studios in Memphis.  Born in Belgrade, Serbia, Popovic formed her first band when she was 19, gaining a strong following throughout Europe and with her band, Hush, opening for such Blues stalwarts as Junior Wells.

Recorded in Ardent Studios’ new Solid State Logic Duality-equipped Studio C, musicians included Frank Ray Jr., Hammond B3; John Williams, bass; and Tony Coleman, drummer for BB King and producer of the new Popovic album. Longtime Ardent engineer Pete Matthews recorded the new material.

“This was one of my favorite sessions of all times,” commented Popovic. “We cut seven great tracks and the amazing sounding studio C at Ardent made this session stand out from all my previous recordings. Thanks to engineer Pete Matthews, the control room sounded just like the record at all times!  And working in the same room where Van Morrison and Bob Dylan recorded simply adds to the magic of the moment. “

Popovic’s album “Still Making History” (2007) was in the Billboard charts for 19 weeks and her “Blind for Love’ (2009) made it to the #1 radio-played Blues album on US radio.  In 2011, Popovic was winner of the Best Blues DVD at the Blues Matters Awards (UK), and her “Unconditional” was a triple nominee for “Contemporary Blues Album,” “Contemporary Blues Female Artist” and “Best DVD” (“An Evening at Trasimeno Lake”) at the 33rd Blues Music Awards in Memphis.

“We cut some blues, funk and soul, the way it used to be done back in the day, but it’s not a retro record,” remarked Popovic.  “We had the spirit of Albert King and Albert Collins in the room, and we’re coming back in August to finish this project.”

Solid State Logic

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Posted by Keith Clark on 07/17 at 05:33 AM
RecordingNewsProductConsolesEngineerStudioAudioPermalink

Monday, July 16, 2012

A Practical Guide To Key Audio Calculations

Thanks to modern technology, you can do dB calculations without knowing a thing about the mathematics of logs, anti-logs, ratios, exponents, or even much about math

This is a practical guide to doing audio calculations, particularly dB (decibel) calculations, covering most common situations.

You see dB numbers all the time in audio, and probably understand that 3 dB is considered a just noticeable change in volume level.

You may also be aware that dB calculations involve “logs” (logarithms). But perhaps you’re not quite so clear on how to figure out what 24 dB from your mixing console means to your amplifier rated for 1.4V input sensitivity.

Thanks to modern technology, you can do dB calculations without knowing a thing about the mathematics of logs, anti-logs, ratios, exponents, or even much about math. This article is necessarily long in order to cover many situations where you might need to use dB.

But, if you understand the calculations, you will find that most of them repeat the same things in different ways. If you begin to see this, it means you are beginning to understand how to calculate dB.

INTRODUCTION
First, if you don’t have one, you need to buy yourself a cheap scientific calculator. It MUST have several specific functions on it. One is a “Log” function. There are two common Log functions. One is “Log base e” or “natural Log” which you DON’T want. The other is “Log base 10” which is what you need. You can check if it is “Log base 10” by simply entering 10 and hitting the “Log” key. The display should read 1. If it doesn’t, don’t buy it.

Another necessity is a 10x function (also called 10 to the x function or the anti-log function). You can check if this is the proper function by entering 2 then hitting the 10x key. The display should read 100. You also need a +/- (change sign) key. It must also have plus, minus, divide, multiply, and equal (=) keys. You can basically ignore the other functions on it to do dB calculations.

Once armed with this tool, you can now learn to do dB calculations. Actually, you don’t really have to learn much except to push the right buttons on the calculator.

In each example, you will be told exactly what calculator key to hit and what your answer should be. Once you get the right answers as shown, you can substitute your own numbers in the various examples to figure out your own things.

Remember, a calculator is a DUMB device. It will only do what you tell it to do. So you MUST use your brain a bit to see if your answers make sense. This means if you come up with a number like 23841 dB or 20,000,000 watts, it is wrong. Nothing in audio has 23841 dB of anything and 20,000,000 watts is out of the question, unless you are providing sound reinforcement for Space Shuttle launches.

All the answers given in this article only show the first two digits (numbers) to the right of the decimal point. Because fractions of a dB or a watt rarely have little practical significance in audio, these two digits are shown ONLY so you know you got the right answer.

The answer on your calculator may be 15.84893192 but the answer given here is 15.84. As you will see in one of the examples, chopping off digits like this can lead to slight, but not significant, errors. Note again the last digit to the right in the examples is NOT a rounded off value in the examples, but chopped off. If it was rounded off, the answer would be shown as 15.85.

Most calculators can be set to display only 2 digits after the decimal point. It rounds off numbers to do this so that 15.84893192 becomes 15.85. So, if you set the calculator to do this for figuring out the examples, your last digit to the right may be 1 larger or smaller than in the examples.

For actual audio work, you should only be concerned with dB numbers to the left of the decimal point. Thus an answer of either 15.84 or 15.85 dB should be rounded off to 16 dB when stating the result.

For each example calculation, the actual formula being used for the calculation is also shown in red. The actual calculation procedure is in blue.

The most common “Log” calculations you need are: dB to voltage, voltage to dB, voltage gain to dB, dB to voltage gain, calculating SPL for distances, and converting amplifier watts to SPL changes or SPL changes to amplifier watts. Therefore, these are the only examples given. With a little brainwork, you may be able to apply the example calculations to figure out other things.

DEFINITIONS
Some definitions to know:

Any dB value is a RATIO, meaning it represents one number divided by another. If you simply state something in dB then you are only stating the ratio in between one thing and another. So you might say the difference in two voltages is 6 dB but that only means one voltage is twice the other (6 dB = 2 times voltage). It doesn’t tell you anything about the actual voltages.

If you want to state the actual value of something in dB, most common audio calculations have a 0 dB reference value that is indicated by suffix (a letter or letters following dB). The 0 dB reference value is always used as one of the numbers for the ratio. You should always use a suffix when stating the dB of something as an actual value so that anyone else will know what 0 dB reference is.

So when you say your mixer is putting out + 6 dB, you really need to say + 6 dBu or whatever the 0 dB reference is. Thus +6 dB means you have twice as much voltage while 0 dBu means you have 0.775 volts or 1.55 volts.

For electronic calculations (voltage and wattage) the 0 dB references are: 0 dBu (or dBv) = 0.775 volts
0 dBV = 1 volt
0 dBm = 1 milliwatt (0.001 watts). The standard reference value for 0dBm is 0.775 volts into a load of 600 Ohms. For most audio calculations, simply assume you are NOT using a 600 Ohm load and thus dBm can equal dBu.

For example, a +24 dBm output specification on a mixing console can be used as +24 dBu to calculate voltages. The reason for this is that modern audio equipment will put out the same voltage whether there is a “load” on it or not. Thus knowing the power in a line level audio circuit is of little value and simply complicates what you need to know.

Another example: 6 dBu is the ratio of some voltage divided by 0.775 volts. That voltage is 1.55 volts.
For SPL (Sound Pressure Level) calculations the 0 dB reference is:
0 dB SPL = 0.00002 pascals (a pascal is a measure of pressure, in this case air pressure, just like a meter is for distance).

Another example: 100 dB SPL is the ratio of some sound pressure to 0.00002 Pascals. That pressure is 2 Pascals
These are the symbols used in the formulas:
”x” means multiply
”/” means divide
”^” This symbol indicates what follows is an exponent of the number preceding it. An exponent means “raised to the power of”, as in 10^2 is 10 raised to the power of 2 or more simply stated as “10 squared”. With dB calculations you get “funny” powers like 10^(34/20). Spelled out this is “ten to the power of thirty four divided by twenty”. Don’t be afraid of this. The instructions and your trusty calculator will get you through it without having to fully understand it.

”( )” in the formulas means that everything inside the parenthesis is calculated first to come up with a single number. In the first example below, (24/20) is calculated first to come up with 1.2. Then 10 is raised to the power of 1.2.

dBu TO VOLTAGE
How many volts does a mixing console put out with a maximum rating of +24 dBm?

Although rated in dBm you can “change” this, as stated above, to dBu for this calculation. Using you new calculator you need to find out how many volts +24 dBu is above 0 dBu.
Formula: Volts = 10^(dB/20) x volts @ 0 dBu or 10^(24/20) x 0.775
Enter 24 (dBu)
Hit the divide key
Enter 20
Hit the = key
Your answer should be 1.2
Hit the 10x key.
Your answer should be 15.84
Multiply this by 0.775
Your final answer should be 12.28 volts

The reason you multiply by 0.775 is that any dB number is always a RATIO (one number divided by another). So 15.84 is the numerical ratio of +24 dBu to 0 dBu. Put another way the voltage at +24 dBu is 15.84 times bigger than 0.775 volts so you must multiply 0.775 by 15.84.

The 15.84 is used as a multiplier and will be called this from now on. However, to be mathematically correct, this multiplier is actually a ratio representing one number divided by another.

dBV TO VOLTAGE
Suppose the output was rated as +24 dBV? What would its voltage be?
Formula: Volts = 10^(dB/20) x volts @ 0 dBv or 10^(24/20) x 1
Enter 24 (dBV)
Hit the divide key
Enter 20
Hit the = key
Your answer should be 1.2
Hit the 10x key.
Your answer should be 15.84
Multiply this by 1
Your final answer is, of course, still 15.84 volts
In this case +24 dBV is 15.84 times larger than 1 volt.

GAIN
Your amplifier puts out 70V with a 1.4V input. How much gain does it have?
Formula: dB = 20 x Log (volts1/volts2) or 20 x Log (70/1.4)
Enter 70 (volts1)
Hit the divide key
Enter 1.4 (volts2)
Hit the = key
Your answer should be 50.00
Hit the Log key
Your answer should be 1.69
Hit the multiply key
Enter 20
Hit the = key
Your answer should be 33.97 dB gain

Suppose you only knew your amplifier had 33.97 dB voltage gain (we’ll round this up to 34 dB). What would its maximum output voltage be?
Formula: Multiplier = 10^(dB/20) or 10^(34/20)
Enter 34 (dB)
Hit the divide key
Enter 20
Hit the = key
Your answer should be 1.7
Hit the 10x key
Your answer should be 50.11

The voltage at the output will be 50.11 times bigger than the voltage at the input. (Note: this number appeared during the first calculation for this amplifier as 50.00 but we rounded up the gain from 33.97 dB to 34 dB.)

Take the next step. The input sensitivity on the amplifier is 1.4 volts.
Formula: Volts Out = Multiplier x Volts In or 50.11 x 1.4
With the 50.11 (multiplier) still displayed, hit the multiply key
Enter 1.4 (volts in)
Hit the = key
Your answer should be 70.16 volts

Thus, your amplifier will put out 70.16 volts with a 1.4 volt input with the input control at maximum. If you wanted to put 8 volts in your amplifier it will clip unless you turn the input control down. But, your control is marked in dB, so how far do you turn it down? Not a problem to figure out. Read on.

VOLTAGE TO dB
The question is what is the difference in dB between 1.4 volts and 8 volts?
Formula: dB = 20 x Log (volts1/volts2) or 20 Log (1.4/8)
Enter 1.4 (volts1)
Hit the divide key
Enter 8 (volts2)
Hit the = key
Your answer should be 0.17
Hit the Log key
Your answer should be -0.75
Hit the multiply key
Enter 20
Your answer should be -15.13 dB

So guess what? Turn your input control down to the -15 dB point and now when you put 8 volts in, the amplifier will put out its full 70.16 volts. Why? Back to the calculator.

You amplifier has 34 dB gain. You turn your input control down 15 dB so now it effectively has 34 dB - 15 dB = 19 dB gain between the input jack and the output.
Formula: Multiplier = 10^(dB/20) or 10 ^ (19/20)
Enter 19 (dB)
Hit the divide key
Enter 20
Hit the = key
Your answer should be 0.95
Hit the 10x key
Your answer should be 8.91

Your output will now be 8.91 times bigger than the input. Your input is 8 volts so multiplying 8 times 8.91 gives you 71.30. Well, that’s not exactly the 70.16 volts we got before. Why? Because the numbers were chopped off in these calculations. If you used the numbers with all the digits, the answer would have come out as 70.16621271 volts which is the precise answer arrived at previously. Is this difference significant? No. You can find out why it isn’t by using the next example calculation to find the dB difference between 71.30V and 70.16V.

Suppose you have one device that has a maximum 15.5 volt output and the device you wish to drive with it accepts a maximum input of only 7.75 volts. What is the dB difference?
Formula: dB = 20 x Log (volts1/volts2) or 20 x Log (15.5/7.75)
Enter 15.5 (volts1)
Hit the divide key
Enter 7.75 (volts2)
Hit the = key
Your answer should be 2
Hit the Log key
Your answer should be 0.30
Hit the multiply key
Enter 20
Hit the = key
Your answer should be 6.02 dB
The output of the first device is 6 dB more than what the second device can accept.

You’ll notice that in the first voltage to dB calculation you ended up with a minus dB number and in this one a plus or positive dB number.
+dB AND -dB

In the last example, if you used the formula 20 x Log (7.75/15.5), your answer would be -6.02 dB. It simply depends whether a larger number is divided by a smaller one (answer is always +dB) or a smaller one is divided by a larger one (answer is always -dB). The basic number will be the same no matter which way you divide. Try reversing the 7.75 and 15.5 in the last example. You should get -6.02 as the answer.

Also, if you calculated the dB difference between the 71.30V and the 70.16V for the amplifier outputs, you would have gotten 0.139 dB or -0.139 dB, depending on which number you divided by which. This is why the difference was not significant: you can not hear a 0.139 dB difference.

The main reason for dividing a smaller by a larger number when calculating dB is to figure out the LOSS in dB. You will see this in the SPL and amplifier calculations below.

VOLTS TO dBu or dBv
You have a device that puts out 8.8 volts. What is that in dBu?
Formula: dBu = 20 x Log (volts1/volts2) or 20 x Log (8.8/0.775)
Enter 8.8 (volts1)
Hit the divide key
Enter 0.775 (volts2)
Hit the = key
Your answer should be 11.35
Hit the Log key
Your answer should be 1.05
Hit the multiply key
Enter 20
Hit the = key
Your answer should be 21.10 dBu

To find dBv simply substitute “1” for “0.775” in the calculation. Your answer should be 18.88 dBv.

SPL CALCULATIONS
You do SPL calculations using exactly the same formulas as for voltages. Both SPL and voltages are “pressures” and the factor “20” is used for both. Your loudspeaker puts out a maximum of 120 dB SPL at 3.3 feet = 1 meter. What is the SPL at 60 feet?

First, you must understand that SPL drops 6 dB for each doubling of distance. Why is this? Calculate the multiplier for a pressure change of 6 dB.
Formula: Multiplier = 10^(SPL/20) or 10^(6/20)
Enter 6 (dB SPL)
Hit the divide key
Enter 20
Hit the = key
Your answer should be 0.3
Hit the 10x key
Your answer should be 1.99 = about 2

So the multiplier for a 6 dB SPL change is 2. This means if you move twice as close to the loudspeaker, it will be 6 dB louder.

Now recalculate this using another function on your calculator called the change sign key. This changes a number in the display from a plus to a minus number.
Formula: Multiplier = 10^(-SPL/20) or 10^(-6/20)
Enter 6 (dB SPL)
Hit the “+/-“
Your display should change to -6 (dB SPL)
Hit the divide key
Enter 20
Hit the = key
Your answer should be 0.3
Hit the 10x key
Your answer should be 0.50 = 1/2

So the multiplier for a -6 dB SPL change is 1/2. This means if you move so you are only 1/2 as close (meaning twice as far) to the loudspeaker, the SPL is 6 dB less.

Back to the problem. Your are 60 feet away and you know what the SPL is at 3.3 feet = 1 meter. You already know this should be a -dB number so you are going to divide the small number by the big one.

Distances are equivalent to voltages so you simply divide the two distances. You can do this for any two distances and substitute the SPL at one of the distances for the Max SPL in this example.

If this SPL is for the further distance, you must divide the further by the nearer distance. Always divide the distance you are finding the SPL for into the distance for which you know the SPL.

Otherwise, you will find your answer will not make sense, such as having more SPL at a further distance. Sound just doesn’t seem to work this way.
Formula: dB SPL = Max SPL + (20 x Log (distance1/distance2)) or 120 + (20 x Log (3.3/60)
Enter 3.3 (distance1)
Hit the divide key
Enter 60 (distance2)
Hit the = key
Your answer should be 0.055
Hit the Log key
Hit the multiply key
Enter 20
Hit the = key
Your answer should be -25.19
Hit the + key
Enter 120
Hit the = key
Your answer should be 94.80 dB SPL

Note: the SPL formula is only valid outdoors and for the direct sound from the loudspeaker indoors. Indoors the reverberation of the room will take over at some distance from the loudspeaker and the sound level will remain more or less constant the further you move away. This is a subject beyond the scope of this article, but it will at least give you a rough idea of what to expect indoors.

AMPLIFIERS
There are really only two dB calculations you generally need for amplifier outputs.

One is how many dB SPL is there between two wattage levels.

The other is what is the difference in watts you need to achieve a given change in dB SPL.

Because amplifier watts are power, that number 20 you’ve been using thus far in the calculations changes to 10 for doing the “log” calculations.

Otherwise, the formulas are the same. Through a quirk in mathematics, you can use the dB answers you get when calculating differences in wattages as representing the change in SPL from the loudspeaker.

While this is not mathematically correct, it is a perfectly practical thing to do. Substitute some other numbers in the following example, such as 2 watt and 4 watts. Then use 2000 watts and 4000 watts. You will find the answer is exactly the same (but the price tags are definitely not!!).

WATTS TO dB (or Watts to SPL)
You have a 100W amplifier and want to change to a 350W amplifier. How many more dB will you get out of your loudspeaker?
Formula: dB = 10 x Log (watts1/watts2) or 10 x Log (350/100).
Note the larger number is divided by smaller will give you +dB. You DO expect higher SPL, don’t you?
Enter 350 (watts1)
Hit the divide key
Enter 100 (watts2)
Hit the = key
Your answer should be 3
Hit the Log key
Your answer should be 0.54
Hit the multiply key
Enter 10
Hit the = key
Your answer should be 5.4 dB

This is the power difference in dB between these amplifiers. If you were going from a 350W to a 100W, the answer would be -5.4 dB as the smaller number would be divided by the larger. Try it.

You could go through correct mathematical conversions, but because we are being practical, the SPL from your loudspeaker will be 5.4 dB greater with the 350W amplifier than the 100W amplifier. Or, it will be -5.4 dB less using a 100W amplifier instead of a 350W amplifier.

dB TO WATTS (or SPL to Watts)
Suppose you want to increase your SPL by 8 dB and you have 100 watts to start with. How many more watts would this take?
Formula: Multiplier = 10^(dB/10) or 10^(8/10)
Enter 8 (dB)
Hit the divide key
Enter 10
Hit the = key
Your answer should be 0.8
Hit the 10x key
Your answer should be 6.30
So you need 6.30 times as many watts to get an 8 dB SPL increase. How many watts is that?
Formula: New Watts = Multiplier x watts or 6.3 x 100
Enter 6.3 (multiplier)
Hit the multiply key
Enter 100 (watts)
Hit the = key
Your answer should be 630W
Thus, if you start with 100W, you need 630W to increase your SPL by 8 dB.

SUMMARY
The above examples should give you valuable tools you need to calculate dB in the most of the ways you need for audio signals. Use them wisely.

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Posted by Keith Clark on 07/16 at 01:34 PM
AVFeatureBlogStudy HallAmplifierAVEducationLoudspeakerProcessorSound ReinforcementAudioPermalink

Recording Electric Bass - Going Direct Or Use A Microphone?

Either way can be effective - here are some approaches and anecdotes
This article is provided by BAMaudioschool.com.

 

My first day as a real engineer rather than an assistant was all about bass. The engineer (who was the studio manager as well) took a break after we recorded basic tracks on a Salsa song.

Before leaving the room he told me to punch where the bassist wanted. I started, and was easily able to hear and punch individual notes rather than whole phrases.

A few times I disagreed about which note was pulling the groove off but punched where I was told anyway.

After doing the punches, the bassist and producer agreed that we should have punched what I indicated instead, and we had to punch BOTH notes…the one that was originally out and the one we “fixed”.

After a while, I realized that the engineer should have returned. I turned around to see him sitting in the back of the room watching. When I jumped up and said, “Oh sorry, I didn’t see you” he told me, “I’ve been watching…sit back down, ‘cause it’s now YOUR gig.”

In order to be able to punch the bass, you have to capture it first. There are two aspects to recording electric bass - direct or putting a mic in front of the bass amp’s speaker.

DIRECT
Recording electric bass using a direct box is rather simple. The problem many people encounter is too much compression. I tend to use slight compression to smooth out the transients (usually caused by popping techniques and uneven notes) rather than try to force every note to be the exact same volume.

Although I know many people that automatically crank up as much bottom as possible on every direct Bass they record, I usually add a little 100 Hz (WHEN NEEDED) and also a little bump at around 3-5k (AGAIN, WHEN NEEDED) so the “note” comes through more clearly. Sometimes I will add a little higher frequency to hear more of the “finger attack” or “pick.”

My favorite signal path for recording direct electric bass is a Neve 1073 or 1081 mic pre going into an LA2A, with just enough compression that the needle stays at zero but drops down no more than 2-3 dB at times.

The trouble with trying to compress and squeeze every note to be the same volume when recording is that you may end up losing some of the tone and dynamics of the performance.

YOU CAN ALWAYS COMPRESS MORE OR DIFFERENTLY DURING THE MIX. ALTHOUGH YOU MAY BE ABLE TO MAKE A “FLAT” SOUND MORE FULL, YOU CAN NEVER UNDO COMPRESSION.

I remember, in the analog days, trying to experiment with how to record bass so that the bottom did not saturate the tape. At one point I even tried dropping the bottom and boosting the top when recording, then reversing that process on playback (sort of like Dolby).

For that experiment I used an API EQ, dropping 100 Hz and boosting 10 kHz going into the tape machine, and another API with opposite settings coming out. Of course I recorded the bass on another track straight, without the EQ changes. The track with the EQ had a “rounder” bottom end, but the track recorded straight had a thicker lower midrange that worked better in the mix.

Now that the whole world is digital, tape compression is not an issue (so I recommend just going straight without playing the EQ-in/EQ-out game).

Oh, if you hear a strange occasional buzz on the bass that you can’t track down, see if anyone is using a copy machine in the studio. I once spent an hour tracking down a buzz before noticing it only happened when someone in the lounge was making a copy.

MIC ON THE AMP
The trouble with putting a mic directly in front of a bass amp speaker is that although you do get some bottom due to the “proximity effect”, the REAL bottom of the bass needs much more room within which to develop.

If you mic far enough for the low sound waves, you may introduce a slight delay. What I prefer to do is stick a mic close to the cabinet as well as one a few yards away for the real bottom. Sometimes I just use the far mic.

In either case, it is very important to make sure the far mic track is moved earlier, either by sliding the track back (if you are digital) or flipping the tape and bouncing in Repro (if you are analog).

I was recording a famous jazz fusion band, and we just finished all of the basic tracks (so all of the drum and other mics were still out). We took care of some bass punches and I flipped the tape and bounced in Repro so I had an “early” bass track to send into the bass amp.

I mic’ed the bass cabinet with just a far Neumann U 47 microphone, and sent the early bass track through a delay on the way to the amp so I was able to tweak the time and really lock the amp sound with the direct one (I intended to combine them when mixing).

The bassist of the fusion band was scheduled to be interviewed and photographed, so as a goof we put every single mic around the bass cabinet. The photographer was amazed and snapped away at the bassist posing by his rig…never realizing that the only mic that was really being used was the lone U 47 in the distance.

Finally some reflections on some great bassists I’ve recorded:

Marcus Miller: What can I say? Marcus is amazing, and we developed such a close relationship that he was able to stop playing, look at me, and then play a single note. Because I was always paying close attention I would usually know the note he was talking about and be able to rewind then punch into record at that note (of course sometimes he would play me the phrase before the note he wanted punched).

One day he was recording a bass solo through a Marshall amp. The amp blew (complete with light show) in the middle of a phrase. A half hour or so later when the amp was repaired (after we took a break), I rolled back to the beginning of the phrase and then punched in RIGHT at the note the amp blew on. Marcus played through seamlessly, as if the punch was seconds after the original performance rather than over 30 minutes.

By the way, people often ask me what chorus I used on certain phrases of “Mr. Pastorius” on the Miles Davis “Amandla” album. That was no chorus, that was Marcus DOUBLING his parts so closely people thought it was an effect on a single track.

Bootsie Collins: When I recorded Bootsie, he was playing a bass with three outputs. Each output went into a different effects chain, and I substituted my Mutron III envelope filter for the box he had (his Mutron had long since died). Although I was told that people usually combined the three signals into one recorded bass track (and the producer suggested I do so as well) I had enough tracks to record each output on a separate track.

When I mixed, I started by getting general sounds, then automating the balances between all three outputs on a part-by-part basis. The sound was great, and I was able to emphasize different aspects of each output as well as each sound combination. (Bootsie played a very funky guitar as well, with his foot stomping the beat as he played).

Bruce A. Miller is an acclaimed recording engineer who operates an independent recording studio and the BAM Audio School website.

 

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Posted by Keith Clark on 07/16 at 01:16 PM
RecordingFeatureTrainingAnalogDigitalInterconnectMicrophoneProcessorSignalStudioAudioPermalink

Extron Announces New High Resolution VGA Line Driver with EDID Minder

Extron Electronics has introduced the new Extender Plus VGA and Audio Line Driver with EDID Minder.

This one input, one buffered output line driver extends VGA-QXGA and HDTV component video along with audio signals up to 250 feet (75 m).

Unbalanced computer stereo audio is converted to balanced, line level stereo audio to eliminate noise usually associated with unbalanced audio when distributed over long cable runs.

It includes EDID Minder, which automatically manages EDID communication between connected devices to ensure that the source powers up properly and reliably outputs content for display.

The Extender Plus is available in Decora-style and AAP form factors, providing convenient AV access and signal extension for a wide variety of environments.

“VGA line drivers are standard in nearly every professional AV system design, but now they need to do more than just boost AV signals,” says Casey Hall, vice president of aales and marketing for Extron. “The new Extender Plus products expand the capabilities of the popular Extron Extender Series by including EDID Minder, a valuable technology that ensures reliable EDID management for computers and displays.”

To maintain signal integrity over long distances, the Extender Plus provides video amplification and peaking control to compensate for attenuation that can occur in long cable runs. Proper signal compensation supports a more detailed image with greater contrast.

Additional integrator-friendly features include an EDID capture mode, selectable resolutions and refresh rates, and real-time status LED indicators for system monitoring. Both versions also include an energy-efficient, external universal power supply for worldwide compatibility.

Extron Electronics

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Posted by Keith Clark on 07/16 at 01:08 PM
AVLive SoundNewsProductAVInstallationInterconnectNetworkingAudioPermalink

Peavey MediaMatrix NION Training Seminar Scheduled For Bay Area In July

Peavey Commercial Audio will host a two-day MediaMatrix NION Certification training course in Emeryville, Calif., during July, where AV designers, consultants, contractors and end users can learn the 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, a 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 at Meyer Hall on the campus of Ex’pression College in Emeryville on July 30-31, 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, please visit mm.peavey.com/education or http://www.peaveycommercialaudio.com/education.

Peavey has educated thousands of AV system designers, integrators and end users from around the world since MediaMatrix was introduced in 1993.

MediaMatrix courses are also offered online at mmtraining.peavey.com. Completion of the online Peavey MediaMatrix Basic or Advanced course earns two hours of credit toward InfoComm’s CTS and CTS-D Certification Renewal.

Peavey Commercial Audio

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Posted by Keith Clark on 07/16 at 12:07 PM
AVLive SoundChurch SoundNewsProductTrainingAVDigitalEducationNetworkingProcessorAudioPermalink

C3 Productions Brings L-Acoustics KARA Line Source Loudspeakers To Brighton

The south coast-based production and hire company has added 12 KARA cabinets with eight SB18 subwoofers

UK-based L-Acoustics Certified Provider SFL Group has supplied a KARA WST line source system to C3 Productions of Brighton.

SFL Group also demo’d the system for C3 prior to purchase in several London venues and provided training at its Reading premises.

The south coast-based production and hire company has added 12 KARA cabinets with eight SB18 subwoofers, three LA8 amplified controllers and flying hardware to its inventory.

This is C3’s first L-Acoustics purchase, and the company chose KARA because it needed a medium format high quality line array to use across live music, performance arts and corporate live events.

“Our mission statement is to provide our customers with the latest and highest performance technology and this system enables us to deliver that,” says C3 director Jon Crawley. “We investigated all the systems currently available on the market and felt that KARA offered the best combination of cost, performance and gave us a unique product in the local area. It is small enough to be discrete for theatre shows but powerful enough to deliver a full-on rock show.

“We have always been fans of the K1 system, and KARA has a similar clean, powerful and dynamic sound, which suits our core live music business. The achievable levels from the system are hard to believe given the size.”

C3’s KARA system made its debut at one of Brighton’s main annual events, the Great Escape festival, which showcases new talent from around the UK. It was used on the NME stage in the 2,000 capacity Corn Exchange, which featured emerging and existing artists such as Dry the River, Mystery Jets and Bookashade.

“There were some very difference music styles but the system sounded incredible throughout and we only got positive responses from all touring parties,” continues Crawley. The system has since been used at the Jubilee celebrations, for a theatre production on Hastings seafront and will be on the main stage at BLOC weekend in London in July with artists such as Snoop Dogg and Orbital.

C3 Productions
SFL Group
L-Acoustics

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Posted by Keith Clark on 07/16 at 08:25 AM
AVLive SoundNewsAVBusinessLine ArrayLoudspeakerSound ReinforcementAudioPermalink

Friday, July 13, 2012

Church Sound: Establishing Proper Gain Setting On A Mixing Console

One of the primary keys to a good sounding system

The purpose of any sound reinforcement system is to amplify a source, whether that is a single speaking voice, a group of singers, a collection of musicians, a piece of pre-recorded music, or other sound sources, and to deliver that to the listener.

Each individual piece of equipment in the signal chain between the source and the loudspeaker can affect the level of the signal.

That means that the mixing console, outboard effects processors, crossovers, and amplifiers (obviously) can all add to or subtract from the signal level.

The mixing console can almost be considered to be an audio system in itself, containing microphone preamplifiers, equalizers, and perhaps even dynamics processing.

Each includes a gain stage that can amplify the signal. Then there are the input channel faders, subgroup faders, and the master faders. Again, each can add gain to the signal.

Each component part of the mixing console, and piece of audio equipment in the following signal chain, also adds noise to the system, since all electrical circuits produce some level of noise. The amount of noise depends on the circuit design, and may be negligible in the case of a digital device or may be a great deal, especially where analog gear is involved, and particularly with older or poorly maintained equipment.

If you turn everything up in your system with no signal going through it then what you hear coming out of the loudspeakers—that hiss—is the sum of the noise being introduced by everything in the signal path (the equivalent of the tape hiss we mentioned).

The level of hiss that you hear is known as the noise floor. It may also include buzzes from grounding problems or other interference, but those are issues for a separate article.

At the other end of the scale, too much gain introduced by any device in the signal chain will “clip” the signal, which means that the signal peaks are flattened by the circuit’s inability to handle the level, resulting in a distorted sound.

With analog circuits a little distortion is tolerable. Digital clipping, however, just plain sounds nasty, and is to be avoided at all costs.

There is a window in which the system operates at its optimum, where the noise floor does not mask a signal that has been set too low and high-level signals do not distort.

The fundamental aim when setting the gain through a mixing console is to ensure that the loudest signal passes through at a level just below clipping, while also allowing for signal peaks. That extra leeway is known as headroom.

Let’s take a basic signal path, a microphone connected to the input of the mixing console, the output of which is connected to an amplifier and a loudspeaker. To set up the console, begin by turning down all the input gain (also known as trim or input level or input sensitivity) controls. Disengage the pad switches, if there are any.

Pull all the faders—inputs, subgroups, and masters—all the way down. Set the equalizer level controls to the 12 o’clock position or switch the EQ out of the signal path. Turn all of the auxiliary sends down and set the pan (also known as balance) controls to the 12 o’clock position.

On the relevant input channel, select PFL or Pre Fade Listen. This switch routes the input signal to the meter (and usually the headphone output) from before the fader, so the fader position has no effect on the signal.

The loudest level of voice or instrument that is likely to occur during the service or performance should be produced during this setup.

As the person speaks, sings, or plays their instrument, adjust the channel input gain control while watching the LED or VU meter until the loudest sounds peak at between +6 dB and +9 dB on the meter scale.

On a console outfitted with bargraph meters this zone is typically delineated by yellow LEDs. Avoid setting the level so high that the red LEDs illuminate, or the VU meter needle enters the red area, which will result in a distorted signal.

To allow yourself some wiggle room later, back the level off slightly.

By the way, note that not all models of mixing console offer the same amount of headroom.

Once you’re familiar with your console you will know just how much headroom you have before distortion and can setup the levels accordingly.

Repeat the procedure for each of the input sources. Remember that gain can be added at other stages in the console. If you add EQ, for example, you will need to re-check the level using PFL and the meter.

Those numbers around the EQ controls refer to the dB value added or subtracted in the selected frequency range. Since maximum EQ gain is typically 15 dB, that will affect the signal level and may require the input gain control to be reset lower to compensate.

A mixing console typically gives its best signal-to-noise performance—where the amplitude of the signal is proportionally at its greatest compared to the amplitude of the inherent circuit noise—with the input faders at 0. This is also known as ‘unity’ and is often marked by a heavier line or shading next to the fader. At this position, the fader is not adding to or subtracting from the signal level. 

Also, fader scales are generally logarithmic, which means that small movements at the bottom end of the fader travel will have more of an effect than higher up the scale.

You also don’t want to be at the top of the scale, as that will prevent you from being able to add any more level, if required.

Audio subgroups may be used to control a group of instruments or voices on one or two faders. If your console has subgroups, these faders, which receive their inputs from the group bus assignment switches on the input channels, should also be set at 0. If you find while you are mixing that they are more than a little above or below the unity position then you should adjust the input gains slightly.

An audio bus is a means of getting the audio from an input to an output of your choice. Think of an input channel as being similar to a narrow side road that dead ends. Traffic on that road, and other roads, may all need to go to the same destination. To do that, they must all join a larger road—a bus.

Each bus links each input channel with a master channel or section that typically features an output. So, for example, subgroup assign switch #1 on each input channel routes that input signal to subgroup #1, through the subgroup fader, and to an output connector.

In turn, that subgroup may also be assigned to the stereo or LCR (left/center/right) master output via another set of buses, through the master fader or faders, and out of the master output connectors.

Consoles may also include any number of auxiliary send and matrix buses, all fed by switches or level controls on each input channel or the subgroup channels.

As with the input levels, you do not want to overload the console buses and cause distortion. Maintaining a good gain structure that essentially passes the signals through at unity, as described above, will help you to achieve this.

With all of the console gain stages operating properly together, and at the optimum signal-to-noise ratios, you have one of the primary keys to a good sounding system.

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Posted by Keith Clark on 07/13 at 02:07 PM
Church SoundFeatureStudy HallConsolesDigitalSignalSound ReinforcementAudioPermalink

Power Viruses: The Source of Electronic Influenza & The Cures

Prevention of power viruses is possible only when prevention is systematic

Software viruses are common occurrences in the computing world.

In fact, almost everyone is familiar with their potential for damage, and the news media routinely reports the details of system disruption related to their appearance.

It’s a reasonable term to apply to these rogue programs. They enter a system unseen, often incubate in silence, and eventually come to life with results that range from merely annoying to disastrous.

Electrical disturbances are quite similar.

In fact they could reasonably be called “power viruses” since they, too, are unseen and can cause serious and expensive electronic system failure.

POWER VIRUS ORIGINS
Power viruses are contracted the same way as other viruses. They’re passed along - often by your system’s electrical neighbors. Plug your system into the wall, turn it on, and look out. You’ve just been exposed to an epidemic, and there are a lot of very sick electrons looking to cause problems.

Some of them may take time to cause noticeable damage. Others are immediately catastrophic (like a lightning strike).

How do power viruses affect an electronic system? What can you do to prevent power viruses in the first place?

Start by understanding them, and then you can tackle the job of immunizing your system against their harmful effects.

There are six main power viruses that can invade a system. The symptoms they cause can vary as can the proper course of treatment.

Voltage Spikes & Impulses
This virus is mostly the result of electrical equipment inside your facility. Electrical loads like elevators, motors, relays, induction furnaces, copy machines, and similar devices can cause sudden large increases in voltage inside the electrical system.

Conditions outside your facility can be to blame, too. Switching activities by the electric utility and lightning strikes can cause transient impulses so intense they literally “blow up” sensitive micro-circuitry.

This virus is deadly to electronic systems - but not always immediately. Sometimes voltage spikes and impulses are relatively small in amplitude. In these cases, the virus weakens the system components over time leading to deteriorating health and eventual failure.

Other times the impulses may be so large that they cause immediate system failure.

Electrical Noise
Like voltage spikes and impulses, electrical noise is generally created inside the facility by the system’s electrical neighbors. Almost every electricity- consuming device contributes its share of electrical contamination.

Things like appliances, photocopiers, laser printers, and electronic lighting ballasts are all noise sources that can cause computers to lock-up, lose data, or behave unreliably.

Even computers themselves generate electrical noise. It’s truly a paradox that our computers often infect other computers with power viruses.

Common Mode Voltage Problems
Traditionally, this power virus hasn’t received much attention.

But detection of common mode voltage problems is easier and, as a result, more system problems are being traced to its existence.

The condition is characterized by unwanted voltage measured between neutral and ground (the white wire and the green wire or conduit) in the electrical system.

In fact, the common mode voltage virus is probably the most serious power virus infecting electronic systems today.

It occurs as a result of high impedance safety grounds, neutral conductors shared with other circuits, and branch circuit lengths that are excessive.

When the electrical noise virus (already mentioned) appears between the neutral and ground conductors it becomes a common mode virus with the ability to cause lost files, system lockups or re-boots, communication errors, and “no problem found” service calls.

Voltage Regulation

This virus is characterized by abnormal variations in the electrical circuit’s nominal operating voltage (120 volts, for example). These variations are generally greater than ±10 percent of nominal voltage and may last for several line cycles or more.

Traditionally this virus has been referred to as the “sag” or “swell,” and it is typically caused by large loads turning on and off, and overloaded branch circuits or distribution transformers. In some cases, voltage regulation viruses can be the responsibility of the power utility.

If an electronic system requires tightly regulated voltage (most of today’s systems don’t) the voltage regulation virus is likely to cause system lock-ups and unreliable operation in addition to damaged or destroyed components.

Blackouts
Blackouts are the most visible and easily identifiable of all the power viruses. And they have the most obvious cause and effect relationship.

One moment power is present—the next moment it’s not and your system is dead in its tracks as a result. The effects of unanticipated power loss are obvious. This is especially true if the system is a network or some other “fault intolerant” architecture.

Fortunately, in spite of what most UPS manufacturers advertise, blackouts account for comparatively few occurrences of all the power viruses.

Backdoor Disturbance
This virus (as its name implies) infects your system via a secondary path.

Even though they’re not an AC power connection, things like serial ports, telephone lines, network cabling, and I/O connections can all permit power viruses to invisibly enter a system.

This virus causes driver chip failure and communication errors.

The back door disturbance virus is often unrecognized.

Without treatment, serious damage can occur, and lost productivity can result in substantial financial losses as well.

AN OUNCE OF PREVENTION

There’s an old adage that “An ounce of prevention is worth a pound of cure.” Nothing could be closer to the truth when it comes to power viruses.

We’re familiar with the damage that results from software viruses and we’ve all experienced the debilitating and sometimes deadly results of real life viruses like influenza. We go to great lengths to avoid both.

In our personal lives, we get vaccinations, eat healthy diets and exercise (most of us), shun contact with infected people, and generally avoid living the type of high risk life style that leads to illness.

Where our electronic systems are concerned, we’ve learned to practice “safe computing.” We back up our data regularly; avoid logging onto’ questionable bulletin boards and networks, or sharing data sources (USB, CD-ROM, etc) of unknown origin. We also run anti-virus programs on a routine basis.

It doesn’t require a huge leap of logic to ask, “Why don’t we practice safe computing where power viruses are concerned?”

They have the same potential effect where our systems are concerned. They enter unseen. They can cause damage ranging from annoying to catastrophic. And like most other viruses, prevention is possible if you understand the basics.

There are five simple devices you can use to prevent the equivalent of electronic influenza. All five are required for complete immunity.

THE MAGIC PILL

If there’s a magic pill to prevent power viruses, it’s understanding that prevention must be practiced as a “system.” What that means is that certain prevention techniques must be used together.

Voltage spikes are addressed with a surge diverter and electrical noise with a noise filter. Each of these by themselves, however, is capable only of weakening or slowing down a virus not eliminating it.

Isolation transformers eliminate common mode voltage problems. When surge diverters and noise filters are added to the isolation transformer, the resulting “system” kills all three viruses.

Uninterruptible power supplies eliminate blackouts, but in spite of many manufacturer claims, most aren’t capable of preventing other viruses. Once again, the UPS must be used with the other parts of the system to achieve total virus immunity.

The backdoor disturbance can be addressed several ways. Fiber optic connections are one means of electrically closing the back door, but if ordinary copper wiring is used for communication lines, it may be necessary to employ special surge diversion techniques for these connections.

Luckily the voltage regulation virus is no longer a serious hazard. Once upon a time, this virus was responsible for many system failures. However, most of today’s systems use switch mode power supplies. This technology was designed as a way of reducing both power supply size and cost while simultaneously increasing electrical efficiency.

To achieve these goals, switch mode supplies are designed to consume electrical power differently than their predecessors. These operational differences have created a beneficial by-product where voltage regulation is concerned.

As a result, most systems enjoy substantial immunity to the voltage regulation virus. Additional preventative measures (voltage regulators, etc.) are unnecessary.

CONCLUSION

Power viruses are an appropriate description of the power quality problems that can plague electronic systems. Like other viruses, they are invisible - often announcing their presence only after some initial damage has already been done. Their effects can be a minor annoyance like a lockup or system error or they can be catastrophic like a blown up integrated circuit or power supply failure.

Our dependence on sophisticated technology has created an increased awareness regarding the need to safeguard system integrity. Software viruses have led to the introduction of “antivirus” programs and system data is routinely backed up to prevent loss.

Part of this “safe computing” lifestyle should be the prevention of power viruses, too. This is possible only when prevention is systematic.

Voltage spikes, electrical noise, and common mode voltage is eliminated by a package that contains an isolation transformer, surge diverter, and noise filter. UPS and data line protection can be added to the system as applications demand.

Also see Protection Or Quality? With AC Power For Systems, It’s Vital To Know The Difference.

Dennis Ver Mulm works with POWERVAR, based in Lake Forest, Illinois.

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Posted by Keith Clark on 07/13 at 12:51 PM
AVFeatureBlogStudy HallAVPowerSystemAudioPermalink

Joe Tessone Joins Sensaphonics As Audio Engineering Consultant, a.k.a., “Sound Guy”

Joe Tessone has joined Sensaphonics as the company’s new audio engineering consultant, a.k.a., the new “Sensaphonics Sound Guy,” where he will advise customers on the implementation of in-ear monitoring, handling questions ranging from the basic to the highly technical, helping ensure a positive IEM experience.

Joe Tessone, 27, is a 2007 graduate of the Audio Arts and Acoustics program at Columbia College Chicago, with specialties in Audio Design and Production.

In addition to his duties at Sensaphonics, he owns a commercial studio, Mystery Street Recording Company, and is the Audio Archivist at the Old Town School of Folk Music in Chicago.

He is also a performing musician, most recently as singer/songwriter for the eclectic punk band Behold!, and as banjo/ukulele player for an Americana music group, the Rust Belt Ramblers.

Tessone is also passionate about occupational health risks and workers’ rights, which is a perfect fit with Sensaphonics’ focus on hearing conservation as an industry-wide issue among both engineers and musicians.

“I’m a big believer in healthy hearing. Our ears are the most important tools we have. Just as it would be crazy for a welder to work without eye protection, it is just as dangerous for us in the music industry to work without protecting our ears,” he says. “I own two pair of custom Musicians Earplugs, the ER-15, and always have one with me.”

Sensaphonics president Michael Santucci states, “Having a sound engineer on staff is a big help to our customers, and Joe Tessone really stood out as the obvious choice. Being a performing musician with strong engineering skills and customer focus means that Joe has the breadth of knowledge and perspective required for the role of Sound Guy.”

Tessone views his new job as a natural next step in his personal development within the music industry. “Working with the industry leader in both in-ear technology and hearing conservation is a great opportunity,” he says. “One thing I learned at Columbia College is that all of us in the audio industry really need to support each other. Being the Sound Guy at Sensaphonics will allow me to share knowledge and solve problems for other musicians and engineers. I’m very happy to be here.”

Tessone is available to answer IEM questions via phone and email, /To contact him, go to www.sensaphonics.com and click on “Ask the Sound Guy.”

Sensaphonics

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Posted by Keith Clark on 07/13 at 09:35 AM
AVLive SoundRecordingChurch SoundNewsTrainingAVBusinessEducationManufacturerMonitoringAudioPermalink

UK-Based Theater Sound Designer Richard Brooker Relies On Waves Audio

For the new musicals I Dreamed a Dream (based on the life of Susan Boyle) and an adaptation of The Bodyguard, Waves tools play an important role in Brooker’s sound design

Richard Brooker has a long history as an acclaimed sound designer on some of the industry’s biggest theatrical performances, including such credits as Jesus Christ Superstar, Never Forget, The Sound of Music, Cabaret, Chess, Daddy Cool and many more.

He has witnessed his job evolve as audience expectations and the theaters themselves have changed, and recently he has come to rely on tools from Waves Audio for both live situations and his personal studio.

His use of Waves features prominently in the sound design for the new stage musical I Dreamed a Dream (based on the real-life story of opera crossover superstar Susan Boyle) and the upcoming The Bodyguard (an adaptation of the award-winning Whitney Houston/Kevin Costner film).

For I Dreamed a Dream, Brooker’s console of choice is a DiGiCo SD8 powered by the fully-integrated Waves SoundGrid platform for real-time processing; for The Bodyguard, he is using a DiGiCo SD7 with SoundGrid.

“I began working with Waves tools at home in the studio on my Pro Tools rig,” notes Brooker. “I love them, and they are always a part of building my mix. They are unbelievably well-thought-through products, with something available for whatever you want to do. And of course they are great sounding plugins too, which is probably the most important issue.”

Brooker reflects on the difference between using Waves for theater versus a rock performance, as well as the shifting dynamics of the audience’s relationship to a show:

“In theater, the sound design should support the storyline, the emotional journey of the characters and the score itself. We must draw the audience in during the dialogue, in order that they can commit to the show emotionally.

“And we have to push them back into their seats and overwhelm them in the big numbers. Traditionally, sound designers have taken a ‘less-is-more’ approach in musical theater. That is still partially the case, but we have seen the role of the sound designer evolve moving into the current era.

“Audiences today tend to be a bit more demanding of things like a fully-produced sound, a surround-sound layout and special effects. It is a challenge, but Waves tools have helped me make those transitions. As intense as it can be, I love my job and it’s tremendous fun. The addition of Waves to live consoles is like opening up an Aladdin’s cave for me.”

Brooker’s favorite Waves plugins include the Renaissance Compressor, Renaissance Reverb, Bass Rider, SSL G-Equalizer and SSL G-Master Buss Compressor.

He also uses the CLA (Chris Lord-Alge) Artist Signature Collection in his recording studio. He is also fond of the IR1 Parametric Convolution Reverb, as well as the One Knob Phatter and One Knob Wetter.

His go-to “problem-solver” is the Renaissance Compressor. “It is transparent even when working really hard,” he notes.

The new Susan Boyle musical has presented some unique challenges, as Brooker points out.

“This is a tricky show. There is a range of intensity. All of the show is amplified using lavalier microphones and wireless transmitters, with miniature DPA mics hidden in the actors’ hair or wigs. After the show is done, we present them with a full-on concert-style, epic set of numbers.

“Susan will personally make an appearance at some, singing into a handheld unit. That portion is really a great big sound, so we need to be set up for a range of dynamics. The Renaissance Compressor is perfect to place Susan just where she needs to be in the mix.

“To the audience, it is imperceptible, but they would certainly notice if it weren’t there. Then, the band sound is shaped by the Renaissance Reverb and SSL Buss Compressor, with a little bit of the SSL G-Equalizer for good measure.

“I am a DiGiCo man through and through, currently using the SD8 and SD7, and the Waves plugins enhance what is already an amazing sounding console. DiGiCo and Waves are the perfect combination.”

Waves Audio

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Posted by Keith Clark on 07/13 at 06:37 AM
Live SoundNewsPollConcertConsolesDigitalSoftwareSound ReinforcementAudioPermalink

Soundcraft And Studer Consoles Support RAI’s Landmark Recording Of Rossini’s La Cenerent

Harman’s Soundcraft and Studer’s Italian distributor, Leading Technologies Srl, along with sister company Sound Light & Technologies (SL&T) recently participated in an exciting recording project with RAI Radio, using a combination of Soundcraft and Studer digital mixing platforms in order to optimize the sound.

The vast production of Rossini’s famous La Cenerentola (Cinderella), which involved around 250 members of cast and technical crew, was mixed on a combination of three Studer Vista 9 and Vista 5 consoles, with three further Soundcraft V1 desks providing backup.

These were used in four different scene locations: while the actors, singers and choir were variously located in Parco della Mandria in Venaria, Palazzina di Caccia and Palazzo Reale in Torino, the orchestra was recorded remotely at the Auditorium RAI in Torino.

All locations were linked by fibre or radio bridge to give the actors the sense of performing in real time with the orchestra—and a Studer stagebox was positioned in each venue.

On duty were two Vista 9 consoles (configured with 52 faders), a further Vista 9 with 42 faders and a 42-fader Vista 5 located in the OB truck, which were used for mixing the Orchestra and actors’ microphones, as well as for 5.1 post-production, while the three Soundcraft Vi1 consoles were used for pre-mix of secondary inputs and backup.

A multitrack recorder received the audio via a MADI port on the Vista 9. The orchestra required 68 microphone channels on the desks with a further 12 channels for the choir (who were based in the same location as the actors).

RAI Radio provided technical staff while SL&T provided personnel support and other ancillary equipment from the HARMAN Professional portfolio, such as JBL EON loudspeakers and AKG microphones and headphones. AKG C414 XLII and various K141 and K702 microphones were used for mixing and monitoring by the musicians and engineers.

The La Cenerentola recording marked the latest success in a long relationship between Italy’s national broadcaster and SL&T/Leading Technologies.

The event itself was transmitted live around the world over a 2-day period in early June and the DVD, produced by Rada Film, will be available soon for marketing and promotion.

RAI Radio has the technical production rights for La Cenerentola, with Antonio Ciano taking overall audio responsibility, Fiervisaggio Giorgetti, technical project and installation supervisor, Marco Diodato and Domenico Narducci as Vista 9 engineers and Dario Chiapino as Vista 5 engineer.

Rada Film has the production rights for Cinderella under the authorisation of Andrea Andermann with the direction of Carlo Verdone. Providing the technical interface between Rada Film and RAI was Alessandro Bernardi.

Studer
Soundcraft
Harman

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Posted by Keith Clark on 07/13 at 06:30 AM
RecordingNewsPollConsolesDigitalNetworkingStudioAudioPermalink

Stevie Wonder Uses Three SD7s At Rock In Rio Lisboa

Headlining the fourth night of this year’s Rock In Rio (Lisboa) festival, Stevie Wonder used three DiGiCo SD7 consoles, ensuring that his sound didn’t rely on any Superstition.

Playing a mixture of hits and cover versions to an audience of over 75,000, Stevie Wonder himself requested the three SD7s - one for Front of House, one for vocal an in-ear monitors and the third for band monitors.

Supplied (along with the rest of the festival’s audio system) by Brazilian rental company Gabisom Audio Equipment, Stevie Wonder’s set included over 80 inputs with each SD7 having two dedicated SD-Racks.

The FoH console was on a redundant Optocore loop, with the monitors on redundant MADI. All SD7s were dual redundant mirrored engines (audio and control), which assured total permanent redundancy.

“When Stevie made the jump from analogue to digital, the SD7 was the ideal choice because it’s the digital console that sounds closest to the analogue console he was using before,” says Wonder’s FoH engineer. “All other digital consoles sounded and felt very much like computers.”

“Having Stevie Wonder personally ask for three SD7s for his Rock In Rio performance was a real endorsement of the console’s standing within both the artist and technical production communities,” says Ian Staddon, DiGiCo vice-president of sales. “Gabisom bought the third SD7 specifically for this show, which shows a fantastic commitment to DiGiCo.”

DiGiCo

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Posted by Keith Clark on 07/13 at 06:22 AM
Live SoundNewsPollConcertConsolesDigitalSound ReinforcementAudioPermalink

Veteran Producer/Mix Engineer Loads Up Sessions With Metric Halo

Veteran producter/mix engineer David Kalmusky loads up his sessions with Metric Halo Production Bundle Plug-ins

David Kalmusky is an effective producer, a natural mix engineer, and a world-renowned guitarist. His list of credits includes The Wilkinsons, The Road Hammers, Emerson Drive, and many others. His work is Grammy-nominated, Juno-awarded, and chart-topping.

About ten years ago, Kalmusky approximated his console workflow on his in-the-box sessions by relying on the sound and functionality of Metric Halo’s ChannelStrip.

Recently, he acquired Metric Halo’s new Production Bundle of plug-ins, which adhere to the same high standards of ChannelStrip and have thus replaced 90% of the plug-ins from other manufacturers.

Kalmusky’s in-the-box sessions are now populated almost entirely by Metric Halo plug-ins.

“I initially hesitated to shift to in-the-box mixing because my console-mixing workflow was so well established and efficient,” explained Kalmusky. “I have great sounding outboard gear, and I can pull up a decent mix in just fifteen minutes. That’s a fast workflow.

“It was about a decade ago that I found Metric Halo’s ChannelStrip. I hadn’t imagined it was possible to have that same fast workflow in the box, but ChannelStrip made it possible. To this day, I always start a session by inserting ChannelStrip on every channel across the session.”

Currently, about half of Kalmusky’s work is executed entirely in-the-box.

In many respects, Kalmusky likes to remain at the technological cutting edge.

“I’m an early adopter of the Pro Tools|HDX system,” he said. “But I’ve been pretty underwhelmed by the third-party support so far offered. That’s why I was pleased that Metric Halo jumped ahead of most of the rest of the industry in embracing this platform for all of its plug-ins.”

But like his hesitance to shift to in-the-box mixing, Kalmusky can also be conservative.

“It usually takes me about a year to fold a new piece of gear or software into my workflow,” he said. “But the integration of Metric Halo’s Production Bundle was fast, which is a real testament to its quality and usability. It has replaced almost all of my other plug-ins.”

Kalmusky has used the new Metric Halo Production Bundle on sessions with Journey, Vince Gill, Carolyn Dawn Johnson, The Wilkinsons, Neal Schon, and Robin Zander (of Cheap Trick) already this year, and that list is growing with every new session he takes on.

The Production Bundle includes all of Metric Halo’s cutting edge plug-ins: ChannelStrip 3, Character, HaloVerb, Multiband Dynamics, Precision DeEsser, TransientControl, and Multiband Expander.

Kalmusky uses Transient Control on all of his drum tracks, and he also likes to use it to tame the transients of small stringed instruments, such as mandolin. He uses Multiband Dynamics on bass guitar and vocals.

“The Multiband plug-ins are smooth and transparent,” Kalmusky said. “The whole Production Bundle seems to be as well thought-out as ChannelStrip. HaloVerb has quite a different sound from other plug-in reverbs. It has a useful character that almost sounds like a vintage echo chamber to me.

“The upgrades on ChannelStrip 3 are also awesome,” he continued. “The integrated spectrum analyzer is useful, and the new compression algorithms expand the range of instruments and situations in which ChannelStrip is the right choice.

“It seems like everyone has tried to copy ChannelStrip, but none of them feel or sound as good as the Metric Halo original. I usually hate terms like ‘musical EQ,’ but it’s really apt in this case. The whole thing sounds natural and analog.”

And despite it’s excellent sonic characteristics, it is so conservative of DSP resources that Kalmusky can simply apply it to every single channel without giving it second thought.

Metric Halo Labs

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Posted by Keith Clark on 07/13 at 06:15 AM
RecordingNewsPollProcessorSoftwareStudioAudioPermalink
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