Monday, May 13, 2013

The Relationship Between Amplifier Damping Factor, Impedance & Cable

What it really means within the context of a system

Ever have one of your friendly amplifier reps walk in your office to present their new mondo-gazillion-watt beast and point out the damping factor spec of greater than a bazillion? Why, gee-whiz! That’s like 10 times more than the other guy! It must be awesome! Right?

Well, as we have seen before, it depends on how you are going to use it. Let’s start with defining damping factor and see what it means to us.

Amplifier damping factor is defined as “the ratio of the load impedance (loudspeaker plus wire resistance) to the amplifier internal output impedance.” 

This basically indicates the amplifier’s ability to control overshoot of the loudspeaker, i.e., to stop the cone from moving. It is most evident at frequencies below 150 Hz or so where the size and weight of the cones become significant.

A system where the damping factor of the entire loudspeaker/wire/amplifier circuit is very low will exhibit poor definition in the low frequency range. Low frequency transients such as kick drum hits will sound “muddy” instead of that crisp “punch” we would ideally want from the system.

The formula for calculating damping factor:


ZL = The impedance of the loudspeaker(s)
ZAMP = The output impedance of the amplifier
RW = The resistance of the wire times 2 for the total loop resistance

Very few amplifier spec sheets state the output impedance, but you can generally call the manufacturer for this spec or you can calculate it by dividing the minimum rated load impedance by the damping factor rating.

For example, if we are using amplifier with a damping factor rating of 400 and it requires a minimum load of 2 ohms, then its output impedance would be calculated as being 0.005 ohms.

So let’s look at several examples and figure out what we can control in the design of our system to achieve the best results. Say we have two 8 ohm subwoofers connected to an amplifier with a damping factor of 400 with 100 feet of 12 ga. wire with a resistance of 0.00159 Ohms/foot times 100 feet gives us a total resistance of 0.159 Ohms.

Plugging the numbers into our formula, we get:


In this case, our system damping factor is just 12. Most experts agree that a reasonable minimum target damping factor (DF) for a live sound reinforcement system would be 20, so we need to consider changing something to get this up.

The critical element in this definition is the “loudspeaker plus wire resistance” part. In this case, the resistance in 100 feet of 12 ga. wire with a 4 ohm load results in around 0.7 dB of loss, much greater than the maximum target of 0.4 dB of loss, so let’s try bigger wire. 10 ga. wire has a resistance of .000999 ohms/foot times 100 feet equals .0999 ohms and will get us to the 0.4 dB target.

What will it do for DF?


OK, now we’re pretty close to the 20 we were looking for. Notice that the loudspeaker impedance can also give us a big change.

The higher the circuit impedance, the less loss we have due to wire resistance.

What if we change our wiring so we have one 8 ohm loudspeaker connected instead of two? Going back to our 12 ga. wire, we calculate:


Even better! In fact, if you run the numbers a few times, you will see that in a system with some significant length of wire, we will find that damping factor will generally be 20 or higher as long as our total wire loss is 0.4 dB or less.

What if we have a self-powered subwoofer? In this case, our loudspeaker wire is probably around 14 ga. and since the amplifier is in the loudspeaker enclosure, it is probably less than a couple feet long.

Assuming the manufacturer is connecting two 8 ohm loudspeakers to the amplifier, and 14 ga. wire has a resistance of .00256 ohms/foot times 2 feet equals 0.00506 Ohms of resistance, and our amplifier has a damping factor spec of 400, what do we get?


Wow! Now that’s a significant difference! Kind of supports the idea of using self-powered subwoofers, or at least putting the subwoofer amps as close as possible to the subs.

So we’ve looked at the differences in the size and length of our wire and the differences in hanging one loudspeaker on the line vs. two to change the impedance of the line.

What if we choose an amplifier with a higher damping factor spec., say 3,000? That’s a big difference, so we should see a much higher damping factor in our circuit, right?

Assuming this amplifier can drive a minimum 2 ohm load, we find the output impedance would be 0.001 ohms. Plugging the numbers into our single loudspeaker with 12 ga. wire system, we get:


Hmm, not such a big deal.

That higher amplifier damping factor only improved our system damping factor by 0.31 over the amplifier with a DF spec of only 400.

What if we use the amplifier with the 3,000 DF spec in our self-powered sub with 2 feet of 14 ga. wire?


Remember our calculation using the 400 DF amplifier was 264.55, so now we start to see when the amplifier spec becomes significant.

Essentially, in sound reinforcement systems where we have some significant length of wire between the amplifier and the loudspeaker, the amplifier DF spec has little affect on the performance of the system.

So what have we learned? In live sound reinforcement systems, damping factor is really driven by the length and size of our wire and the impedance of the loudspeakers we connect at the other end.

Since damping factor is mostly affects low frequency, we should endeavor to keep our subwoofer loudspeaker lines as short as possible and/or use larger gauge wire. We should keep the impedance of the connected load as high as possible by connecting only one transducer per wire instead of two.

So is more amplifier damping factor better? As one of my colleagues recently said, “Sure! If the loudspeaker terminals are welded to the amplifier output terminals!” Well, maybe he overstated it a little bit, but yes, as long as the loudspeaker wire is really short, then by all means!

Jerrold Stevens has more than 25 years of experience in the audio industry, including contracting, independent sales representative, live sound and studio engineering, and audio system consulting and design. He now works with Eastern Acoustic Works (EAW).

Posted by Keith Clark on 05/13 at 02:21 PM
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In The Studio: Seven Mixing Techniques That Can Really Pay Off… Or Get You In Trouble

It's so crazy it might just work!
This article is provided by the Pro Audio Files.


This article is about some pretty crazy techniques that can really take your sound up a notch or totally screw up your mix! These aren’t techniques I use all the time, but enough to warrant a mention.

If you’ve got some experience under your belt, here are a few things you can do when the situation warrants it.

1. Multiple Outputs Tor Group Sends
This is useful if you know you’re going to be doing some parallel processing. Most commonly I’ll use this approach on the drum bus.

Why? Well, if you use a single output into two send channels you get the same levels going to both. If you have independent control you can set them evenly to begin with, then apply your compression to your parallel track. From there you have finer control of how much of each element is going into the compressor.

Once you blend your parallel compression in, you might find that while your cymbals may start to fluff up nicely, your snare and kick are starting to “pancake” out. By individually controlling the output levels you can fine tune the compressor’s reaction.

Why not do this from auxiliary sends? Because DAW aux sends tend to be glitchy enough that there can sometimes be a delay on the return. It doesn’t help you out if you’re comb filtering your return when the goal is to get a fuller sound.

2. Leaving The Midrange Bump In Vocals
Four out of five times I find myself cutting some kind of bulky tone between 350-600 Hz in vocal tracks. This usually leads to a cleaner sound.

However, in the mix you may find the solid midrange to be very helpful, especially once you’ve added some treble. One way to control this range but not remove it is to use a compressor with a customizable side-chain circuit.

Triggering the compression without the lows, upper-mids and treble of the vocal opposed to regular compression will open up the presence of the vocals like an EQ would, but without losing that dead center midrange. If you have a knee control you can make the compression very transparent even without the upper and lower ranges in the detector circuit. Conveniently, the stock digi compressor in Pro Tools will let you do all of this.

3. Overdrive For Compression
I discovered this while tracking vocals through my 1176. The transformer will actually apply its own compression when driven into.

If I set the attack on bypass and really fine tune the input, I can get a very subtle compression that brings up sustain without compromising any attack. The “drawback” here is that this process generates distortion. If I can find that sweet spot, the distortion will excite the sound coming in. I’ve done this on vocals, guitars and bass guitar very successfully – compressing without compression engaged!

The way this works will vary from each piece of gear but you might just find something new about one of your favorite (or least favorite) items.

4. Clipping Instead Of Limiting
Taking that idea a step further: one of the riskiest but potentially coolest approaches is to square off a signal instead of limiting.

Clipping is essentially limiting with a zero attack and release time. Limiting is far more transparent in terms of frequency control, but often comes at the price of punch. Likewise, clipping is not transparent in terms of tone, but leaves the dynamics outside of the peak signal completely intact.

Now, this usually sounds terrible. But over very short spans of time, particularly on sources that have broad frequency content, it can actually sound fairly transparent or even good (snare drums anyone?).

Now a lot of audio guys will jump down my throat for this one, but remember, we soft-clip things all the time. Hard clipping your converters is really not so different than overdriving the outputs of an MPC, which hip-hop producers have been doing for years.

5. Adding A “Note” To Kicks
You’ve probably triggered a sine wave from a kick drum to add weight to it. It’s one of many helpful ways of getting a beefier kick sound. This one will actually help glue the kick into the track.

Use a square wave and a low pass filter with the corner frequency set to the fundamental tone. Because slight overtones will still remain, you will get a “note” rather than just weight. This can really help the kick “belong” in the record. The pitfall is that you have to be careful what note you are choosing.

There’s only so much range in that sub area, so you have to be pretty careful about your tuning. Stay with the bass and/or the chord, whichever modulates the least, or stay a fifth down and away from those notes. Once your square is at about 65 Hz, you’re probably going to get too much tone.

6. Parallel Distortion
A cool way to excite something is to create a mult, filter it, and add a touch of distortion. Then blend that mult under the original signal. Hi-passing everything under 1 kHz can be a great exciter for vocals. Hi-passing 100 Hz and low passing 1 kHz can be cool on rhythm guitars or bass to add body. It’s like EQ, but creates a harmonic signature that helps give things a lot of depth.

Now, normally I prefer to do this with linear phase EQs because it gives me exactly what I’m going for without any kind of phase cancellation between the dry and parallel signal.

However, sometimes that cancellation can be useful. If you set your corner frequency and slope just right you can gently ease off any frequency ranges you might not want too much of — so with a discerning ear, minimum phase can be the way to go.

7. Delay The Room

Often times I get tracks to mix that were recorded with nice equipment but not in the most brilliant space. The space is either too tight or two loose. Loose can be OK, a little dynamics processing can help it out. Too tight it almost doesn’t sound like a room capture — and what do you do with it then? Turn it into the coolest delay ever!

If you have a tight room sound, instead of simply blending it up -20 dB below the close sound, strap a delay on there with some feedback and turn it to 100 percent wet return. Now you’ve got an echo that sounds exactly like the room it was tracked in — perfectly realistic and immediately blends with everything else.

If the room is loose you might be able to sneak a little more room in there by adding a slight delay (20-50 ms) with no feedback — essentially acting as a pre-delay. This way you can get a little more of it in there without losing the forwardness of your close capture.

The last thing you can try is this: if you want something to feel far way you can nudge the room capture forward in time. Instant “back wall” if you need it!

Remember the key here is discretion! These techniques can be great, but they can also destroy what’s there.

If 66 percent of the song is the performance, 20 percent is the tracking, 10 percent is the basic mixing, then the last 4 percent is this kind of stuff. Level, Pan, and basic signal processing (in that order) are far more useful and important in getting right. But every now and then you need a little extra “somethin’ somethin’.”

I hope I’ve given you a new technique to try out. I’d love for you to pay it forward by leaving a comment (here) with a cool technique you’ve used and what kind of results it has achieved!

Matthew Weiss engineers from his private facility in Philadelphia, PA. A list of clients and credits are available at

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

Posted by Keith Clark on 05/13 at 01:51 PM
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Friday, May 10, 2013

Riedel Earns Sports Emmy Award For Production Of Red Bull Stratos Project

Supplied key communications gear and support for landmark event

Riedel Communications has been awarded a Sports Emmy Award in the category of “Outstanding New Approaches Sports Event Coverage” for the production of the Red Bull Stratos project.

As a contributor to the project, Riedel received the award from the National Academy of Television Arts and Science at the 34th Annual Sports Emmy Awards, held May 7 in New York City.

In a first-of-a-kind sports production from the edge of space, the Red Bull Stratos project captured aviation pioneer Felix Baumgartner’s ascent to 128,100 feet in a stratospheric capsule and balloon and his subsequent freefall jump, rushing toward earth at supersonic speeds, and parachute landing. (Read more about it here and here.)

For the production, Riedel furnished the fiber-based video and signal distribution systems, as well as the wireless video links from the capsule’s onboard cameras, enabling the stunning pictures delivered from the Red Bull Stratos capsule.

And in fact, Riedel provided the entire communications solution for this record-breaking project, integrating both wireless and wired digital intercom systems.

Riedel team members (left to right): Karin Bock-Leitert, Scott Gillies, Matthias Leister, Jay Nemeth, Scott Bradfield, Jacqueline Voss, Phil Olsman, Thomas Riedel, Werner Eksler, and Charlie Rosene.

Riedel Communications

Posted by Keith Clark on 05/10 at 12:52 PM
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Church Sound: How To Get Audio From An iPad

Approaches in routing signal to your mixer/console
This article is provided by Behind The Mixer.


iPads are turning up in the audio booth as a sound source. Much like my article on using different types of smartphones as audio sources, iPads are something you need to consider. Let’s look a how to get the audio out.

The iPad uses a common 3.5 mm stereo headphone plug, so connecting via the 3.5 mm plug is the easiest route to go.

The back of your mixer/console is filled with 1/4 -inch TRS plugs and RCA plugs. No 3.5 mm inputs so we are going to have to do a little conversion work.

A 3.5 mm stereo plug and a TRS plus look the same in that they have a tip, a ring, and a sleeve. You must understand, however, that these cables can be used in two different ways. One is to carry a stereo signal and the other is to carry a balanced signal.

In the case of the iPad, it’s got a headphone (stereo-out) plug and the audio mixer channel inputs are for balanced signals. Therefore, while it might be easy to think the cables work the same way because the plugs look the same, it’s like having two plastic pipes where one is used to carry drinking water and the other is to carry waste.  Both might be PVC pipe but with different contents.

Time To Convert
There are a few ways you can convert that 3.5 mm stereo plug into something usable. You can go the route of an RCA cable or you can use two dual TS unbalanced plugs. Plug the 3.5 mm end into the iPad and the other into a stereo channel on the mixer.

For the 1/4-inch plug, go the route of a 3.5 mm stereo jack to dual 1/4-inch mono plugs. For the RCA plug, you can use the standard RCA adapter.


The Better (But More Expensive) Route
Using adapters, such as I’ve listed, is an effective way to get the signal into the mixer. However, there is a direct box that will help you with your signal levels as well.

That’s where the Radial ProAV2 DI box comes into play. Use a 3.5mm male-to-male end cable and plug one end into the iPad and the other end into the Radial DI, and it will convert the signal to the right type as well as convert the signal to the proper line level as pro and consumer-grade electronics usually output at different levels. The DI plugs into your mixer/console using a right and a left XLR cable.

Converting audio from an input device like an iPad to your board can be easy when you have the right tools at hand…and you know the type of signal that’s coming out of your equipment and the type of signal that’s expected to come in.

Ready to learn and laugh? Chris Huff writes about the world of church audio at Behind The Mixer. He covers everything from audio fundamentals to dealing with musicians, and can even tell you the signs the sound guy is having a mental breakdown. To view the original article and to make comments, go here._blank_blank

Posted by Keith Clark on 05/10 at 12:47 PM
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Thursday, May 09, 2013

AVnu Alliance Welcomes New Member Companies To Advance AVB Certification

New Alliance additions include Bose and LG

AVnu Alliance, the industry consortium that certifies Audio Video Bridging (AVB) products for interoperability, has introduced four new member companies: Bose, LG, Arrive Systems, and Revolabs.

Standing at 55 global members, AVnu Alliance is a consortium of professional A/V, automotive, and consumer electronics companies working together to drive certification and interoperability of open AVB standards and to enable interoperable platforms.

The alliance has recently announced new AVB educational programs and new partnerships, as well as created the AVnu Alliance Broadcast Advisory Council to advance the broadcast industry’s AVB requirements.

“Bose is passionate about high-quality digital audio and supports customers who are looking to make audio networking easy and reliable-after all, it should be,” says Akira Mochimaru, general manager of the Bose Professional Systems Division. “Through the AVnu Alliance, Bose is proud to be active in an industry initiative that combines connectivity and performance with seamless interoperability-a valuable benefit to our customers.”

“We are excited about these new members joining the Alliance. Not only do they each add a valuable mix of experience and innovation, but a diversity of products in automotive, audio/video, imaging, wireless and network infrastructure,” adds Lee Minich, chair, AVnu Alliance Marketing Work Group, and president, Lab X Technologies. “These new members are representative of the variety of markets that are poised for the greatest value from the Alliance’s universal desire for robust networked AV transport using a single set of non-proprietary standards.”

“AVnu Alliance is dedicated to creating a new ecosystem of interoperable AVB devices through certification.  These new members will contribute to accelerating this mission across our core markets of professional A/V, automotive, and consumer.”

The AVnu Alliance recently opened its certification testing for AVB-enabled networking bridges and professional audio endpoints at its appointed testing house, the University of New Hampshire InterOperability Laboratory (UNH-IOL). AVnu Certification is available to alliance member companies.

AVnu Alliance

Posted by Keith Clark on 05/09 at 03:11 PM
AVLive SoundNewsAVBusinessDigitalEthernetInterconnectNetworkingPermalink

Soundcraft Vi6 Consoles Chosen For Largest Choir Event For Guinness Book Of Records

Innovative 4-card MADI configuration boosts channel count and flexibility

Soundcraft helped The London Community Gospel Choir (LCGC) into the Guinness Book of Records last weekend when house engineer Raphael Williams and monitor engineer Nikoma Bell specified 96- and 64-channel Vi6 consoles respectively to mix the largest gospel choir ever assembled on a stage.

The event was to celebrate LCGC’s 30th anniversary—with a 3-day series of performance and workshops at London’s South Bank over the Bank Holiday weekend. The highlight was the concert at the Royal Festival Hall on the Sunday, and since this was being multi-tracked for a live album and DVD, recording engineer Simon Changer also had a third, remote 96-channel Vi6, isolated in the back control room.

In order to beat the previous record of 1,138 voices singing together in concert conditions, LCGC invited 30 other gospel choirs from across Europe to participate, so that the front tier stalls of the hall also became a performance area for the extended choir (leaving the audience to occupy the balcony/under-balcony seats and upper tiers). The show included X Factor runner-up Jahmene Douglas and Magic FM DJ Angie Greaves — and the existing record was broken thanks to an ensemble of 1169 vocalists performing simultaneously.

All the sound equipment was provided by Richard Nowell Sound Systems (RNSS), who have also installed a series of Soundcraft Vi series digital desks into the South Bank venues. The choice by Williams and Bell to use the Vi6s was a no-brainer as both are staunch devotees of the platform, and have previously used this winning combination on tour with TinieTempah.

“In fact it was Nikoma who first introduced me to the Vi6,” states Williams. “Since then I’ve become a Soundcraft man through and through.”

With a strong background in church music, Williams had been contacted by the choir “because they knew I had an understanding of their requirements.” He started out with a minor role back in 2010 before choir principle Reverend Bazil Meade earmarked him for the lead role for this project.

While the choir, directed by Becky Thomas, features 50 singers, the prospect of micing up nearly 1,200 vocalists created a number of logistical pressures as he set about providing a large inventory of industry standard close and ambient/overhead mics from the RNSS inventory.

Williams, who also acted as overall technical adviser, explains the challenge he faced in compacting his line inputs. “First of all I put together a complete list of what could happen in a perfect world; we started at 150 channels, consisting of band, acoustic section and vocals, with 40 direct mics, and a further 48 dotted around the hall.

“But closer to the time I reined it in and made it more realistic in keeping with the 96-channel desk count, with nine direct mics for the main members of the choir and 12 large diaphragm mics for the remainder. These are dotted around the stage for the choir to look more artistic.”

He ended up filling his Vi6 to capacity with 30 band and 21 LCGC mics, plus four radio and 24 mics for the extended choir, eight ambient mics within the room and some playback tracks from the hard disc recorder.

“What I like about the Vi series is ultimately the sound, which is clear, crisp and clean,” he adds. “The gain structure allows you enough headroom, in a sense the more you put in the more it gives out.”

“I also like the usability — it’s very easy to move around on and because of the Vistonics [interface] each screen is independent and so you can do two things at once — it makes things flexible.”

With the channel count that was required to handle the recording workflow of 96 channels to a MADI device, he had to think carefully about his card configuration. “To get all 96 ins and 96 outs I had to swop some of the cards to get more MADI outs,” Williams notes.

Williams’ solution was to design a 4-card MADI card configuration based on 64 channels input, 32 channels output (card 1); 32 channels input, 0 channels output (card 2); 64 channels input, 64 channels output (card 3); and 24 channels input, 32 channels output (card 4). This provided 184 input channels in workflow (but only 96 channels active at any one time), and 128 channels output workflow (but only 96 channels active at any one time).

However, after considering his dilemma, Williams acknowledged with a smile, “Nikoma (monitor engineer Nikoma Bell) probably had it tougher than me, mixing monitors down at the stage.”

Bell states, “The Vi6 is always my desk of choice; there are other great desks out there but I can configure and use the Vi with my eyes closed, often getting 80 percent of the mix sorted via the offline software before even switching the desk on. It sounds amazing too.

“The ability to configure the show offline really helped on this occasion due to the channel count. I was using every input and output, seeing 64 lines from stage then 12 return lines from FOH, as well as my effect returns and local inputs. But as I could configure the console on the train to rehearsals it saved a lot of time, allowing the band to get on with what they needed to do.”

The final word came from musical director/production manager Ayo Oyerinde: “The Soundcraft Vi6 was the central hub that enabled us to break the world record for the largest choir ever. From FOH, monitors to multi-track recording, the Vi6 took care of it all with the highest quality sound.”

Harman Professional

Posted by Keith Clark on 05/09 at 10:09 AM
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Wednesday, May 08, 2013

Cable Anatomy Part 3: Everything You Wanted To Know About Instrument Cable

Construction, materials, shielding, specifications and much more

See Part 1, Microphone Cable here and Part 2, Loudspeaker Cable here.

Are instrument cables used for high-impedance or low-impedance lines?

Generally, the source impedance is the determining factor in cable selection. Instrument cables are used for a wide range of sources. Many keyboard instruments, mixers, and signal processors have very low (50 to 600 ohm) source impedances.

On the other hand, typical electric guitar or bass pickups are very inductive, very high impedance (20,000 ohms and above) sources. Typical load impedances are greater than 10,000 ohms, which limits the electrical current flow to a very small amount on the order of a few thousandths of an ampere (milliamps).

How much power does an instrument cable have to carry?

The voltages encountered range from a few millivolts, in the case of the electric guitar, to levels over ten volts delivered by line-level sources such as mixers. By Ohm’s Law this represents power levels of less than a thousandth of a watt.

What kind of frequency response does an instrument cable need? What are the lowest and highest frequencies produced by the source?

The bandwidth spans the entire audible range of frequencies, from the 41 Hz (and below) of bass guitar and synthesizer to the 20 kHz harmonics of keyboards and cymbals. Recording applications demand wide bandwidth to preserve the “sizzle” of a hot performance. Even an electric guitar has a bandwidth of about 82 Hz to above 5 kHz.

How big does an instrument cable need to be? Will a bigger cable sound better? Will a bigger cable last longer?

In order to be compatible with standard 1/4-inch phone plugs the diameter of the cable is effectively limited to a maximum diameter of about .265 of an inch. Larger cable diameters demand larger plug barrels, which sometimes won’t fit jacks that are located close together or in tight places. In terms of both sound and durability, “it’s not how big you make it, but how you make it big.”


What are the basic parts of an instrument cable and what does each one do?

The coaxial configuration is generally used for unbalanced instrument cables. At its simplest it consists of a center conductor, which carries current form the source, separated by insulation from a surrounding shield, which is also the current return conductor necessary to complete the circuit. These three components are augmented by an electrostatic shield to reduce handling noise and an outer jacket for protection and appearance.

What is a stranded center conductor? Why is it important?

A stranded conductor is composed of a number of strands of copper wire bunched together to form a larger wire. Solid conductors having only one strand are the cheapest and easiest to work with when assembling cables, because they do not require the twisting and tinning that stranded types need to prepare them for soldering.

The problem with a solid conductor is that it quickly fatigues and breaks when it is bent or flexed. This makes stranded conductors a must for cables that are frequently moved around, especially when they are attached to human beings playing music.

Finely stranded conductors increase the cost of the cable because of the increased production time and the expensive and sophisticated machinery required to assemble very small and fragile strands into a single conductor. The stranding of the center conductor is only one of a number of factors that influence the overall flexibility of a given cable, but it is generally true that finer stranding increases the flexibility and the flex life of the cable.

What is wire gauge? What gauge wire is used in instrument cables?

The diameter of copper wire is typically given in AWG (American Wire Gauge), with the larger numbers signifying smaller size. For instance, a 20 AWG (or “20 gauge”) wire is smaller than an 18 AWG wire. Generally, instrument cable center conductors are in the range of 18 to 24 AWG, with strands of 32 to 36 AWG.


What gauge should the center conductor of an instrument cable be?

Since the current involved in instrument applications is negligible, the amount of copper in the center conductor has only a very slight effect on the strength of the signal reaching the amplifier. In practice, the center conductor’s size is determined primarily by (1) the necessity of obtaining a maximum diameter of .265 of an inch or less while (2) providing sufficient tensile strength to withstand the rigors of performance without breaking.

The 20 AWG center conductor has become quite standard, normally in the form of 26 strands of 34 conductor has become quite standard, normally in the form of 26 strands of 34 AWG(26/34) or 41 strands of 36 AWG (41/36).

A 20 AWG conductor has a breaking point of approximately 31 lbs. Reducing conductor size to 22 AWG reduces breaking point to about 19 pounds (a reduction of 39 percent); increasing it to 18 AWG increases the strength to over 49 pounds (an increase of 58 percent).

The most common cause of failure for instrument cables is broken center conductors.

What are the differences between tinned copper and bare copper stranded conductors?

Sometimes the individual strands of the center conductor are run through a bath of molten tin before assembling them into a wire. Tinned copper wire is often easier to solder, especially if a lengthy (months to years) shelf life is required, because the tin coat prevents copper oxides from forming. If the cable is to be used immediately upon manufacture pre-tinned strands are not required and add unnecessary expense.

Furthermore, an electrical phenomenon known as skin effect makes the use of tinned conductors a potential threat to the high-frequency signal-carrying properties of the cable. However, the aging effects of the formation of copper oxides on untinned conductors may also cause a gradual deterioration of performance.

What is skin effect and how does it affect tinned copper?

Briefly, skin effect is caused by the magnetic field generated by the current flow in the cable causing electron flow to be concentrated more and more on the outer surface of the conductor as frequency increases. If this outer surface is coated with tin, which has higher resistance than copper, the cable will have a falling high-frequency response and act as an attenuator.

What is oxygen-free and linear-crystal copper? How do they affect sound in cables?

There is a continuing debate concerning the use of oxygen-free and linear-crystal copper wire. These types of wire contain lower levels of oxide impurities and fewer crystal boundaries than standard copper. Since these impurities form tiny semiconductors within the cable, the theory is that the cable itself introduces signal distortion, especially of low-level “detail” information. These claims have been very difficult to document with scientific test equipment, but numerous listening tests suggest there is something to them.

What materials are used for insulation of the center conductor?

The insulation that surrounds the center conductor can be made from thermoset (rubber, E.P.D.M., neoprene, Hypalon) or thermoplastic (polyethylene, polypropylene, PVC, FPE) materials. The thermoset materials are extruded over the conductor and then heat-cured to vulcanize them. This process yields a very high melting point which makes soldering very easy, but the vulcanizing stage adds to the cost and introduces unpredictable shrinkage which can make it very difficult to maintain the desired wall thickness.

Thermoplastic insulations are cheaper to process but will return to a liquid state when overheated, requiring great care during soldering when used to insulate large conductors. The insulation of choice for instrument cable has largely shifted from rubber or E.P.D.M. to high-density polyethylene, with cost being a major factor.

How does the insulation affect flexibility?

The insulation material and its thickness can be very dominant in determining the flexibility of the cable. A finely-stranded conductor insulated with a stiff compound will behave much like a solid conductor, as will a conductor insulated with a very thick layer of a more flexible compound. The thinner the insulation is, the more flexibility it allows in the overall cable.

How thick does the insulation need to be?

The basic electrical requirement for insulation thickness is called dielectric strength and is determined by the cable’s working voltage. The voltages involved in instrument cable applications are very low and very little dielectric strength is necessary to prevent the insulation from breaking down. However, a very important consideration when the cable is to be used for instruments like electric guitars is the amount of capacitance between the center conductor and shield.

What is capacitance and what does it do?

Capacitance is the ability to store an electrical charge. In cables, capacitance between the center conductor and shield is expressed in picofarads per foot (pF/ft.), with lower values indicating less capacitance. Combined with the source impedance, cable capacitance forms a low-pass filer between the instrument and amplifier; that is, it cuts high frequencies, much as the instrument’s tone control does.


Why is low-capacitance cable an advantage? How can cable capacitance be eliminated? How long of a cable can I run before I lose high frequencies?

Lower cable capacitance allows more of the natural “brightness,” “presence,” or “bite” of an instrument to reach the amp, which in turn allows the treble controls to be run lower, reducing “hiss” and other unwanted noise. High-frequency loss from the cable becomes audible and objectionable depending on the source, the amplification and other circumstances. Raising the source impedance or increasing the length of the cable increases the loss; there is no point at which high-frequency loss suddenly appears or disappears.

Guitars typically have much higher source impedances at higher frequencies because of the inductive nature of their pickups, which aggravates the effect of cable capacitance. A guitar will often sound noticeably “muddier” when run through a 40-foot cable, whereas keyboard instruments, samplers, mixers and other line-level devices with low source impedances can usually drive cable runs of hundreds of feet without problems.

How is low-capacitance cable made?

Given that the overall outside diameter of the cable is limited by the plugs that must be used, cable capacitance is largely the result of trade-offs between conductor size (and hence strength), insulation material (cost) and insulation thickness (size and flexibility).

The term dielectric constant is used to rank the insulation quality of a material.

Some materials are great insulators but impractical for use as wire insulation—glass, for instance! As far as practical materials are concerned, the thermoplastics are generally far superior to the thermoset family.

For instance, polyethylene has a dielectric constant of 2.3, while that of rubber is 6.5. This allows a cable with polyethylene insulation to have perhaps one-third of the capacitance of a cable insulated with the same thickness of rubber. It can make an audible increase in the clarity of the sound.


What is the best all-around insulation material for instrument cables?

Polyethylene is very economical and dielectrically hard to improve upon (teflon is slightly better, but its cost is far greater, and its flexibility is far from ideal). Its only drawback is a low melting point which requires a skilled touch with the soldering iron to avoid problems in production.

What does the electrostatic shield do?

As the cable is flexed and bent, the copper shield rubs against the insulation, generating static electricity. The electrostatic shield acts as a semi-conducting barrier between the copper shield and the center insulation which discharges these static electrical charges. Without it any movement of the cable would result in obnoxious “crackling” noises being generated.

What are electrostatic shields made of?
Electrostatic shields first appeared in cable as a layer of rayon braid. Nowadays carbon-impregnated dacron “noise-reducing tape” is a common element in any good high impedance cable. Conductive-plastic (carbon-loaded PVC) electrostatic shields have also become common. Conductive PVC is extrudable just like an insulation, which guarantees 100 percent coverage of the insulation with a very consistent thickness and a very low coefficient of friction.

The superior conductivity of C-PVC makes it much more effective than the semiconductive tape in bleeding off the small electrical charges that cause “the crackles.” Extruded C-PVC is also thinner and more flexible than dacron tape, which is applied longitudinally and restricts the “bendability” of the cable. Although conductive plastic (with a copper drain wire) has been used to completely replace copper braid or serve shields, its effectiveness falls off above 10 kHz.

Why are some cables microphonic?

As noted previously, the center conductor, insulation and shield of a coaxial cable form a capacitor; and, as many a microphone manufacturer will tell you, when the plates of a capacitor are deflected, a voltage is generated. (This is the basis of the condenser microphone!) Similarly, when the plates (conductor and shield) of our “cable capacitor” are deflected (for instance, by stepping on it or allowing it to strike a hard floor), a voltage is also generated.

Unfortunately, this voltage generally pops out of the amplifier as a distinct “whap,” and can be very hard on ears and loudspeakers alike. Effects of this type are called triboelectric noise.

How can cable noise be reduced?

The electrostatic shield’s charge-draining properties help greatly to diminish triboelectric effects. Triboelectric impact noise is also reduced by decreasing the capacitance of the cable with thicker and softer insulation because the deflection of the conductor is proportionally reduced. This is the main reason that the single-conductor coaxial configuration remains superior to the “twisted pair” for high-impedance uses—it allows thicker insulation for a given overall diameter.

Triboelectric effects are accentuated by high source impedances, and are at their worst when the source is an open circuit—for instance, a cable plugged into an amplifier with no instrument at the sending end. Testing for this type of noise requires termination of the cable with a shielded resistance to simulate the source impedance of a real instrument.

What does the shield do?

The copper shield of a coaxial cable acts as the return conductor for the signal current and as a barrier to prevent interference from reaching the “hot” center conductor.

Unwanted types of interference encountered and blocked with varying degrees of success by cable shielding include radio frequency (RFI) (CB and AM radio), electromagnetic (EMI) (power transformers) and electrostatic (ESI) (SCR dimmers, relays, fluorescent lights).

What makes one shield better than another?

To be most effective the cable shield is tied to a ground—usually a metal amplifier or mixer chassis that is in turn grounded to the AC power line. Cable shielding effectiveness against high-frequency interference fields is accomplished by minimizing the transfer impedance of the shield.

At frequencies below 100 kHz, the transfer impedance is equal to the DC resistance—hence, more copper equals better shielding. Above 100 kHz the skin effect previously referred to comes into play and increases the transfer impedance, reducing the shielding effectiveness.

Another important parameter to consider is the optical coverage of the shield, which is simply a percentage expressing how complete the coverage of the center conductor by the shield is.

What are the characteristics of the three basic types of cable shields? Which is best?

A braided shield is applied by braiding bunches of copper strands called picks around the insulated, electrostatically shielded center conductor. The braided shield offers a number of advantages.

It’s coverage can be varied from less than 50 percent to nearly 97 percent by changing the angle, the number of picks and the rate at which they are applied. It is very consistent in its coverage, and remains so as the cable is flexed and bent. This can be crucial in shielding the signal from interference caused by radio-frequency sources, which have very short wavelengths that can enter very small “holes” in the shield.

This RF-shielding superiority is further enhanced by very low inductance, causing the braid to present a very low transfer impedance to high frequencies. This is very important when the shield is supposed to be conducting interference harmlessly to ground. Drawbacks of the braid shield include restricted flexibility, high manufacturing costs because of the relatively slow speed at which the shield-braiding machinery works, and the laborious “picking and pigtailing” operations required during termination.

A serve shield, also known as a spiral-wrapped shield, is applied by wrapping a flat layer of copper strands around the center in a single direction (either clockwise or counter-clockwise). The serve shield is very flexible, providing very little restriction to the “bendability” of the cable. Although its tensile strength is much less than that of braid, the serve’s superior flexibility often makes it more reliable in “real-world” instrument applications.

Tightly braided shields can be literally shredded by being kinked and pulled, as often happens in performance situations, while a spiral-wrapped serve shield will simply stretch without breaking down. Of course, such treatment opens up gaps in the shield which can allow interference to enter. The inductance of the serve shield is also a liability when RFI is a problem; because it literally is a coil of wire, it has a transfer impendance that rises with frequency and is not as effective in shunting interference to ground as a braid.

The serve shield is most effective at frequencies below 100 kHz. From a cost viewpoint, the serve requires less copper, is much faster and hence cheaper to manufacture, and is quicker and easier to terminate than a braided shield. It also allows a smaller overall cable diameter, as it is only composed of a single layer of very small (typically 36 AWG) strands. these characteristics make copper serve a very common choice for audio cables.

The foil shield is composed of a thin layer of mylar-backed aluminum foil in contact with a copper drain wire used to terminate it. The foil shield/drain wire combination is very cheap, but it severely limits flexibility and indeed breaks down under repeated flexing. The advantage of the 100% coverage offered by foil is largely compromised by its high transfer impedance (aluminum being a poorer conductor of electricity than copper), especially at low frequencies.

What type of shield works best against 60-cycle hum from power transformers and AC cables?

The sad truth is that the most offensive “hum-producing” frequencies (60 and 120 Hz) generally emitted by transformers and heavy power cables are too low in frequency to be stopped by anything but a solid tube of ferrous (magnetic) metal—iron, steel, nickel, etc.—none of which contribute to the flexibility of a cable! For magnetically coupled interference, the only solution is to present as small a loop area as possible. This is one of fhe reasons that the twisted-pair configuration generally used in balanced-line applications became popular.

Fortunately the high input impedances generally found in unbalanced circuits minimize the effects of such interference. Don’t run instrument cables parallel to extension cords. Don’t coil up the excess length of a “too long” cable and stuff it through the carrying handle of a amp—this makes a great inductive pickup loop for 60 Hz hum!

What does the outer jacket do? What is it made of?

The jacket is both armor and advertisement; it protects the cable from damage and enhances the marketability of the assembly. As armor, the jacket must resist abrasion, impact, moisture and sometimes hostile chemicals (Bud Light, for instance).

As advertisement, it may be distinctively colored or printed with the name of the manufacturer or dealer for product identification. The materials used for jacketing are the same type as those used for the inner insulation (thermoset or thermoplastic), but the choice is dictated less by electrical criteria and more by physical durability and cosmetic acceptability.

What is the best cable jacketing material?

For years rubber or neoprene were preferred for their superior abrasion resistance and flexibility, but modern thermoplastic technology has produced a number of PVC compounds that are soft and flexible but also very tough. As previously noted, thermoplastic processing is cheaper, faster and more predictable than that for thermoset materials. Only very specialized situations requiring oil or ozone resistance or extremes of temperature and climate demand neoprene or Hypalon jacketing.

The use of PVC has two other major advantages. PVC is not as elastic as rubber or neoprene, and this lack of “stretch” lends additional tensile strength to the resulting assembly by taking some of the strain that would otherwise be borne solely by the center conductor. This has made a dramatic improvement in the reliability of currently manufactured instrument cables.

The other important property of PVC is its almost limitless colorability. Once found only in gray or “chrome vinyl,” PVC-jacketed cable now ranges from basic black through brilliant primary colors to outrageous “neon” shades of pink and green.


• Ballou, Greg, ed., Handbook for Sound Engineers: The New Audio Cyclopedia, Howard W. Sams and Co., Indianapolis, 1987.
• Cable Shield Performance and Selection Guide, Belden Electronic Wire and Cable, 1983.
• Colloms, Martin, “Crystals: Linear and Large,” Hi-Fi News and Record Review, November 1984.
• Cooke, Nelson M. and Herbert F. R. Adams, Basic Mathematics for Electronics, McGraw-Hill, Inc., New York, 1970.
• Davis, Gary and Ralph Jones, Sound Reinforcement Handbook, Hal Leonard Publishing Corp., Milwaukee, 1970.
• Electronic Wire and Cable Catalog E-100, American Insulated Wire Corp., 1984.
• Fause, Ken, “Shielding, Grounding and Safety,” Recording Engineer/Producer, circa 1980.
• Ford, Hugh, “Audio Cables,” Studio Sound, Novemer 1980.
• Guide to Wire and Cable Construction, American Insulated Wire Corp., 1981.
• Grundy, Albert, “Grounding and Shielding Revisited,” dB, October 1980.
• Jung, Walt and Dick Marsh, “Pooge-2: A Mod Symphony for Your Hafler DH200 or Other Power Amplifiers,” The Audio Amateur, 4/1981.
• Maynard, Harry, “Speaker Cables,” Radio-Electronics, December 1978,
• Miller, Paul, “Audio Cable: The Neglected Component,” dB, December 1978.
• Morgen, Bruce, “Shield The Cable!,” Electronic Procucts, August 15, 1983.
• Morrison, Ralph, Grounding and Shielding Techniques in Instrumentation, John Wiley and Sons, New York, 1977.
• Ott, Henry W., Noise Reduciton in Electronic Systems, John Wiley and Sons, New York, 1976.
• Ruck, Bill, “Current Thoughts on Wire,” The Audio Amateur, 4/82.

Thanks to Pro Co Sound for this article.

Posted by Keith Clark on 05/08 at 04:41 PM
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Monday, May 06, 2013

Church Sound: The Importance Of Audio Team Leadership & Organization

Working to be a good delegator, administrator and teacher

Over the years that I’ve worked with churches, the problems I find often have foundation in basic communication, organization and administrative skills - or lack thereof.

Quite often I will visit with a church that is complaining of a lack of consistency in the technical aream and the explanation goes something like: “When Jim is here everything works, but when he’s not, it’s a disaster.”

I know at that point that while Jim may be a great tech operator and may understand the system very well, he is most likely not a good delegator, administrator or teacher.

When I’m at a church that is suffering from the “Jim’s the man” syndrome I can almost guarantee that the mixing board/patching is either not labeled, labeled incorrectly or just poorly labeled. The poor guys who are working tech on the weeks Jim is not there end up scrambling just to get things properly connected and working.

Also, because they are volunteers and “Jim the man” is the golden child in the eyes of the worship leader, people are afraid to step in and to try to organize and logically lay out the board. Other things that end up happening usually relate back to clear organization, such as:

• Batteries failing in the middle of the service because everybody thought someone else had changed them.

• Trying four microphone cables until you find one that works, because nobody throws out or labels the bad cables.

• The last minute scramble to find a mic (or stand, or direct box) that is missing because somebody used it during the week in another room at the church.

• Nobody shows up to mix on a Sunday morning. Bob traded with Steve who traded with George and now nobody really knows who on for the next month.

• The sound operator who is “on” for a given week shows up late because “Jim the man” never told him the worship leader was bringing in a mini-orchestra of 10 players and utilizing six vocalists. The poor guy was actually on-time for a typical Sunday not knowing he had an hour of setup to do.

I’m sure you can add your own list of frustrationsm but rather than moan over them, let’s look at how to prevent them.

1) Get together as a group and agree to a consistent layout of the mixing board and create a channel/patch list that sits next to the board. Also, commit to each other that if for some reason you need to deviate from the standard layout, immediately following the service you will reset the boars to the standard layout.

2) Make a rule that first thing every Sunday new batteries go in the wireless mics. This takes the guess work out of the equation and also lets you use the mics during the week without wondering when the batteries will die.

Wireless mics usually last 6—10 hours on a fresh set of batteries. To be precise, check the specs of your system. So, you can simply do the math: rehearsal/first service/second service = 4 hours plus evening service = 1 hour, and then decide if you need to put in fresh batteries for a midweek event.

3) Throw away bad cables. I know that this is not eco-friendly and everyone likes to occasionally get out the soldering iron. However, in my experience, the repair never happens, or the cable accidentally gets placed back with the good ones, or a repair ends up being rather poorly done.

4) Organize mics, cables and all accessories and put a sign out sheet that details who took the item and to what room they took it to. This way everyone knows where that missing equipment should be located.

5) Hand out or post online a schedule for 6 months of who is on for a given Sunday. In the sound booth (or online) keep a master schedule with the rule being if your name is on for that day you better be there. This doesn’t mean that you can’t trade dates, what it means is that if you do trade it has to be immediately updated on the master schedule.

6) Put the burden on the worship leader to communicate with the actual person that is on for that Sunday ahead of time.

A simple email with a stage layout and instrument list will give the soundman for that week the information he needs to plan on what time he should arrive to setup.

These may sound like simple suggestions, but may churches are simply negligent in these fairly basic tasks.

If there is a no leader of the crew, volunteer to be the coordinator or facilitator that will facilitate the items above. If there is a clear leader, offer to help with organization of the ministry. And if you’re a leader and don’t think you need the help that is being offered, at the minimum it’s time to step aside for awhile, as you’re likely doing your ministry more harm than good.

Gary Zandstra is a professional AV systems integrator with Parkway Electric and has been involved with church tech for more than 30 years.

Posted by Keith Clark on 05/06 at 12:50 PM
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Friday, May 03, 2013

Xilica Audio Design Introduces New Dante-Enabled Digital Processors

Both Neutrino and Uno Series offer models with Dante networking capability

Xilica Audio Design and Audinate have announced the launch of Xilica’s new Dante-enabled Neutrino and Uno Series digital processor models.

Neutrino is an open architecture-drag and drop based DSP while Uno is a hybrid architecture/apps based DSP. 

Neutrino and Uno digital processors are available in 8x8, 8x16, 16x8 and 16x16 I/O models.  Dante enabled models provide an additional 32 channels (16x16) of I/O via Audinate’s Dante network solution.

Note that Neutrino and Uno Series Dante-enabled models not available in North America.

“Licensees such as Xilica are making Dante’s interoperability an easy choice for many end users.  We look forward to seeing more products by Xilica in the future,” says Dave Anderson, Audinate director of sales.

“Product design and engineering for all our DSP products is located here in Canada,” notes Barry Steinburg, Xilica sales and marketing manager for Canada and the U.S. “Providing Dante digital networking capability for our new Uno and Neutrino digital processors aligned perfectly with our established quality standards. Research, testing and customer feedback resulted in our decision to offer the Dante solution.”

Xilica Audio Design

Posted by Keith Clark on 05/03 at 09:37 AM
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Thursday, May 02, 2013

Slick & Seamless: Developing A Live Digital Recording System

Innovation in forming a marriage between live audio and live recording

Choosing an audio console can be likened to a guitarist choosing an ax – it’s the pallet with which one creates. Some engineers have built their mixing techniques around plug-ins while others rely on complex channel grouping or outboard gear.

It’s difficult to classify one platform as “better” or “worse” than another, simply because it boils down to one question: does it help produce the best-sounding show possible in the shortest amount of time?

In addition, due to the enormous influx of digital consoles in recent years, the added criteria of multitracking capability has quickly become a deciding factor in many console choices – whether on tours, in theaters, or houses of worship. With many reliable platforms on the market such as Dante, MADI, CobraNet, AES50, EtherSound and so on, the ability to multitrack one’s show has almost become an expectation.

In my previous article (here), the goal was to offer an inside look at the selection and design process for a road-ready audio system for The Austin Stone Community Church in Austin, TX. Here I’m going to expand specifically on the console and live recording portions of the system, and share some of the results that have come out of the learning process.

Upgraded Capture
In 2011, The Austin Stone released its debut live album, which spent its first week at the top of the iTunes Christian & Gospel charts, and subsequently formed a marriage between live audio and live recording. For that album, I captured the tracks by taking direct outputs from an Avid VENUE Profile console at front of house, via MADI, over fiber optical cable into an RME MADIface.

The data was then piped over FireWire into a MacBook Pro through a PCI Express card. The solution was solid, reliable, and error-free during the six weeks of live tracking.

Upon completion of the album, our producer mentioned that some re-amping was required in post-production to “warm up” the tracks, and asked if we would consider using external microphone preamps for future recording dates for some of the more essential inputs. This began my quest to find a better way to capture the highest quality tracks possible, without lugging around racks of external mic preamps (and spending a fortune in the process).

At that same time, the church was in the process of moving away from its long-time rental system provider. This fortuitous timing afforded me the opportunity to construct a control package from the ground up that would not only handle week-to-week live mixing operations, but seamlessly carry the recording load as well.

I spent the next few months doing as much research as possible on the industry’s leading digital consoles – specifically focusing on the following criteria. First, the headamps and A/D converters had to sound exceptional. In essence, it would be the first and most critical component in capturing the raw sounds, and simply could not be an area of compromise.

Second, the navigation needed to be fast. Since I’m also mixing the band’s monitors from front of house (five stereo in-ear mixes, two mono), the ability to quickly send and pan an input without flipping through a bunch of layers was a must.

I had long enjoyed the 24 mix send encoders on the Yamaha PM5D that we used, and had developed a quick and efficient workflow for monitors-from-FOH applications. I wanted to find something equally quick without the worry of possibly forgetting which layer I was on.

Finally, because of the monitors from FOH needs, it also meant that the console must have a minimum of 24 auxiliary mix buses in addition to the Left-Right/Mono buses.

Finding Direction
My attention turned to the Midas PRO Series. After performing several blind listening tests with several competitors, I was blown away by the sonic quality of the Midas. In my view, the XL8 headamp has the girth and richness of an XL4 pre (a long-standing personal favorite), but with a clarity and “openness” in the high-end that I’ve not heard on any other digital desk.

My only concern was the lack of affordable recording solutions offered by Midas and its sister brand Klark Teknik. There wasn’t a way to record more than 32 channels at 96 kHz for under $14,000.

The RPM Dynamics RPM-TB48 I/O and the Lynx AES50 card it contains. (click to enlarge)

The widely used K-T DN9650 network bridge offers the ability to convert the Midas AES50 format to just about any third-party platform up to 32 channels, at which case a down-conversion to 48 kHz is needed. However, my hope was to keep all of the tracking at 96 kHz to preserve as much of the natural “Midas sound” as possible.

Around this time, I met Jim Roese of RPM Dynamics while mixing the mtvU Woodie Awards last year. Over the week, we talked at length about a 48-channel, 96-kHz, 24-bit recording/playback solution for Midas that he was working on, utilizing a pair of Lynx Studio Technology AES50 to PCI cards, all mounted in a Sonnet Thunderbolt chassis.

The RPM-TB48 I/O is a stand-alone, elegant solution with no external interfaces required, as the Lynx cards perform one single AES50 to Core Audio conversion (at approximately 1 millisecond of added latency). Because the processor load of the conversion is being handled by the interface, the load on the CPU of the recording computer is shockingly low.

A single Thunderbolt cable connects the RPM-TB48 I/O to a computer (a rack mounted Mac Mini running Reaper, in our case), providing a quick, easy, turnkey solution.

At roughly $2,600, the solution is about the same price as purchasing all of the components yourself, but with added benefits such as heavy-duty Ethercon jacks mounted to the modified chassis, a full warranty, and most importantly, it works as soon as you plug it in.

The resulting “recording rack” consists of the RPM-TB48 shock-mounted in a foam-lined rack shelf, a Sonnet Rack-Mac Mini enclosure, and a hard drive running off a UPS. A pair of 6-foot Neutrik Ethercon cables quickly connects the recording interface to the PRO2 via two of the AES50 ports on the console surface.

An added benefit is the ability to quickly and easily take RTA traces of individual channels, simply by picking an input or output on the AES50 network. It accomplishes the same thing as routing a solo through an A/D to a PC running Smaart tuning and analysis software, but only uses one conversion and is much cleaner.

The added functionality allows me to look at the frequency response of each channel, which can be an invaluable tool for both FOH and monitor applications.

Quick Workflow
Once the recording platform was established, the only task remaining was to select the components for the control package.

Front and back views of the recording rack built around the RPM-TB48 I/O. (click to enlarge)

We chose the PRO2 console, specifically, because of the advanced surface navigation and high input fader count. It offers 56 primary inputs, 8 auxiliary returns, 27 phase-coherent mix buses, and 27 faders on the surface. With the “advanced navigation” mode engaged, the ability to access a mix, effects engine, or graphic EQ is merely one button-push away, making the workflow exceptionally quick.

Another benefit is the ability to group and spill inputs via the 8 VCAs and 6 Population groups, which, in my experience, has proven to be much easier than remembering what layer a particular input is on. I use the VCAs to group the band, vocals, and effects returns, while the Population Groups are used to quickly access anything that needs to be cross-fades, such as line-in playback sources or a pastor’s wireless lavalier microphone.

The stage rack houses two DL251 I/O units, each with 48 analog inputs and 16 analog outputs. The primary DL251 handles all of the inputs and IEM outputs, while the secondary unit serves as the system outputs, as well as an extra 48 inputs with which to easily patch a supporting act. Finally, AES50 audio is sent to the PRO2 surface via a 4-channel Cat-5e snake.

The author with The Austin Stone’s Midas PRO2 console. (click to enlarge)

After six-plus months, this solution has proven to meet our expectations, and then some. The band was sold on day one as soon as they heard their in-ear mixes – they couldn’t believe the separation and clarity they were experiencing.

The results in the FOH mix have proven to be almost identical. Unprecedented separation in the mix, paired with a thick, tight low-end and rich, warm midrange has made Midas digital the new standard across The Austin Stone’s campuses.

As we near completion of tracking our third live album, the recordings have been nothing short of stellar, and our studio engineers have fallen in love with the sound quality of the headamps. There is no doubt in my mind that we will continue to see a growing trend with solutions of this type becoming a forerunner in the live recording world in the years to come.

Todd Hartmann is the audio engineering coordinator for The Austin Stone Community Church as well as a free-lance audio engineer.

Posted by Keith Clark on 05/02 at 02:51 PM
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Wednesday, May 01, 2013

Defying Gravity…Safely: Approaches And Best Practices In Flying Loudspeakers

Proper equipment and procedures in putting loudspeakers in the air

Some in audio think that the term “rigging” only applies when loudspeakers are flown, but it also pertains to lesser endeavors such as placing a single loudspeaker on a tripod stand. The bottom line is that for any piece of production gear not sitting directly on the ground, steps must be in place to insure that it does not fall and injure someone (or worse).

The Occupational Safety and Health Administration (OSHA), the U.S. agency that sets and enforces work safety standards, states a company must have “competent” and “qualified” persons in charge of rigging. A competent person is described as one who is capable of identifying existing and predictable hazards in the surroundings, which are dangerous to employees and has authority to take prompt corrective measures to eliminate them.

Meanwhile, a qualified person is defined as one who by possession of a recognized degree, certificate or by extensive knowledge, training and experience, has successfully demonstrated the ability to solve problems relating to the subject matter and the project. Therefore, a qualified person designs a rigging system, and a competent person installs and monitors the rig, and inspects its components.

While OSHA rules and standards are mainly focused toward rigging in the construction industry and are geared toward safety of employees, we also need to look after the safety of performers and members of the public who attend events. Not everyone at a gig may be at the level of a qualified or competent person, but all should focused on safety. Anyone on a production crew who sees a problem with rigging (or any other safety issue, for that matter) can call “stop” and point out the issue so it can be addressed and corrected to avoid an accident or injury.

Good places to look for training and information are manufacturers who make rigging equipment or loudspeakers that fly. They provide specific safety and operating instructions for their own gear. Two organizations focused on entertainment rigging training are the ESTA Foundation ( and PLASA, which offers the ETCP rigging certification program ( for entertainment riggers who work in theaters or arenas.

There are also several independent rigging schools and manufacturers that offer training and certification programs as well. A good reference book is Entertainment Rigging by Harry Donovan. While reading about proper rigging practices is highly recommended, it’s not the same as getting hands-on training. Experience along with knowledge is required for a person to be designated as competent.

The Essentials
Before we talking about specific rigging approaches, let’s look at a few terms. Initials often seen on stands and rigging equipment are “WLL” (Working Load Limit), “SWL” (Safe Working Load), or “MRL” (Maximum Rated Load). For practical purposes, they mean basically the same thing: the maximum amount of static weight that the item will safely hold continuously, when it is used correctly as intended. The key here is a static load, or a load that does not move. Any movement like loudspeakers swinging in the wind or even the act of raising or lowering a chain motor puts additional stresses on rigging equipment. Safe working load limits should never be exceeded.

Another term to be familiar with is “Safety Factor,” the margin of safety added to an item that takes into account loadings over and above the weight being hoisted and for reductions in capacity due to the extra loads imposed by acceleration and inertia (movement).

Some European countries have mandated 10:1 safety margins, while those in the U.S. are still largely self-enforced. Many production companies in the U.S. have adopted a 7:1 safety factor, with margins for life safety (fall arrest, performer aerial acts, etc.) using a 10:1 ratio.

In short, an item’s safe working load is derived by dividing the breaking strength (the point of failure) by the safety factor. An example would be a shackle with a breaking strength of 7 tons would have a safe working load of 1 ton here in the U.S. (SWL= Breaking Strength (7 tons)/safety factor (7:1).

System design plays a large part in the rating of individual components. Like a chain, a rigging system is only as strong as its weakest link. Pull angles and side loadings de-rate the SWL of many items including eye bolts and shackles.

The splay angle of bridles (two or more legs of wire attached to a ring that spreads the load across a larger area) affects their weight loading. Spansets and wire rope slings will have different capacities based on their positioning (i.e., straight vertical pull versus a basket configuration).

All of these factors need to be taken into account by a qualified person who will design the rigging system.

In addition, a competent person should inspect all rigging equipment and hardware before use and periodically do a major inspection for signs of wear, abuse and general adequacy, as well as perform any manufacturer recommended preventive maintenance.

Follow the rules and don’t cut corners when it comes to rigging. Use only loudspeaker cabinets designed by the manufacturer to fly, and use only hardware approved for that specific model. Follow all manufacturer recommendations concerning their individual products. Never modify any rigging hardware as it may affect the weight loading capacity of the item. Purchase only known quality rigging equipment.

Further, factor in enough time to do rigging correctly and make sure the crew isn’t tired—rushing and fatigue cause accidents. Double check everything before it goes up in the air. Once items leave the ground, they better be rigged properly or gravity will demonstrate why it’s the most powerful force in the universe!

Only finger tighten shackle bolts, never use a tool. If you’re worried that a pin might vibrate out, mouse (secure) the pin in place with twine or wire. Always load a shackle pin to end, never from side to side. When attaching to a beam or other structural component, always pad the beam edges so a sharp edge doesn’t injure the wire rope or sling. Never leave any slack in guy wires.

Plenty Of Methods
Now let’s move on to looking at ways loudspeakers can be deployed and positioned.

Ground Stack. The easiest way to set up a PA is by placing it on the ground or stage. While this method seems to have no dangers associated with it, there are a few lurking. Loudspeakers stacked on top on each other can fall from the stack due to vibrations. Truck straps are commonly employed to keep them from vibrating apart.

Keep in mind, however, that strapped stacks can be top heavy and can topple over, either from vibrations or from a crowd pushing to get closer to the stage.  Using a larger subwoofer that provides a bigger footprint as the base of a stack can add stability to a strapped stack of loudspeakers.

Many line arrays are designed to be ground stacked by inverting the fly bar and using it as a base. The cabinets are connected to each other and the base with their fly hardware, making a stable ground stack with the angles being able to be adjusted for coverage same as when flown. Loudspeakers placed on uneven terrain will be unstable and if used on a grass or dirt surface could shift during rain. A common remedy is to place a stage or plywood platform on the ground and level it. Then the loudspeaker stack will have a solid level surface to rest upon and not sink into mud after a rain storm.

Scaffolding. Before flying PA was common practice, stacking loudspeaker boxes on scaffolding towers was a common approach at larger shows. It’s still a common way of elevating horizontal arrays and delay stacks at festivals and fairs. On uneven ground, screw jack leveling legs should be used to ensure the scaffolding is level and all cross and diagonal bracing needs to be in place before loading the tower. Make sure all decking is securely in place before any speakers placed on the tower and they should be strapped down so they can’t vibrate off.

If used outdoors, the tower needs to be guyed down in case of winds. Signage and banners placed on the scaffolding act like sails and will transmit high wind loading to the system. Using an open weave fabric for the signage will allow some wind the blow through, reducing the wind loading on the tower.

Banners should have a quick release system in place so they can be removed rapidly in case of unexpected high winds. Line arrays can be flown inside scaffolding towers, with the chain motor for the array usually connected to a beam secured across the top of the tower. Make sure the beam is secured to the tower and not just held in place by the weight of the PA.

Stands. Tripod stands are commonly employed for loudspeakers at smaller shows. While very safe, the tripod legs should be extended to their largest footprint when possible so they provide maximum stability for the stand. Make sure the top of the stand is correctly sized for the loudspeaker socket or the cabinet could tilt and its center of gravity will not be directly over the pole.

Also, position the stand where the tripod legs will not be a trip hazard as a fall could cause a person injury, as well as possibly knock over the stand (and loudspeaker).

Use fixed leg tripod stands only on level ground. Saddle style sandbags can be used with tripod stands to add a bit of weight to the bottom for increased stability. The sandbag should straddle the leg, not hang from any bracing. To avoid crew injury, larger loudspeakers should be hoisted onto stands by two people.

Truss Totem. Particularly popular on corporate gigs, truss sections are bolted to bases and used in an upright position for lighting trees, projector and delay speaker stands. Totems can be very top heavy so additional weight, usually sandbags, are placed on the base to help with stability.

When used with a single loudspeaker on the top, make sure that the cabinet’s center of gravity is located directly over the center of the truss. When used with a column-type loudspeaker attached to the side of the truss, make sure that additional weight is placed on the base on the opposite side to offset any leaning tendency caused by the side loading of the loudspeaker. Totems should only be used on level ground. If used outdoors, these systems need to be guyed down.

Pole Mounted On Sub. Pole mounting a loudspeaker on a subwoofer provides a clean and easy setup for many gigs. It allows the top cabinet to be raised to a good operating height while eliminating the tripod base that may become a trip hazard.  Some systems allow for the use of two poles for larger top cabinets or line array-style boxes. To avoid problems, use only the manufacturer’s recommended poles with these systems, as different poles may not be compatible or as stable as the factory units.

Crank Towers. These come in many forms from larger versions of tripods to heavy-duty units that can hold 600-plus pounds. Crank towers are becoming a popular option to fly a smaller array without having to use motors. Make sure that all outriggers and legs are extended and the tower is leveled correctly before raising the load. Factor in the rigging hardware weight and speaker cable weight when figuring out the total weight you will be lifting. Check with the manufacturer before using the lift outdoors and follow their recommendations on using the system outside.

Pair Of Towers & Truss. This is a common setup at medium-sized shows and corporates that utilizes a pair of crank towers and a span of truss that goes across the front of the stage. The truss usually does double duty and supports some front stage wash lights as well as the loudspeakers. It can give the ability to provide a small left, center and right array without requiring any ceiling points.

Care should be taken to not overload the systems, because total working load on horizontal truss is based on an evenly distributed load. Also take into consideration the weight of the loudspeaker and lighting cables when figuring out the total weight of the system. This type of setup is considered a system and needs to be designed by a qualified rigger.

Line Array Tower Truss. A relatively new option for positioning loudspeakers is the line array tower. Made from truss sections and specialty hardware fittings, these systems can be disassembled for transport or storage and easily bolted together on the job site to support a wide variety of speaker arrays.

Smaller towers may use a manual hoist to lift the array, while larger units utilize a powered hoist. Make sure the tower is leveled correctly before use. When deployed outdoors, the tower must be guyed down per the manufacturer’s recommendation. Some tower systems allow placing subwoofers on top of the forward outrigger legs so they can act as additional ballast.

Dead Hang Flown. Dead hang means that an item is connected to a support structure without a motor or lift system. The attachment point can be an exposed ceiling beam, tent pole, section of truss, etc. An engineer should be called in to certify the weight loading ability of any ceiling point before use if it is not known.

Only competent or qualified riggers should dead hang cabinets as each hanging point needs to be individually assessed by a person with knowledge and experience to determine what hardware is required to safely support and position the loudspeakers.

Access to dead hang points may involve a personnel lift. Make sure fall protection systems are used by anybody in the lift and that the systems are inspected before use. A spotter may be required to assist the lift operator as many types of equipment have blind spots.

Air Wall Tracks. Air walls are the moveable walls in large meeting spaces and ballrooms. These tracks are often utilized as ceiling points for video screens, lighting and audio delay speakers when they are not being used by the wall sections. While technically considered a dead hang, air walls pose a few specific challenges to production crews. All air walls are not the same, and each requires specific fittings dedicated for that track style.

There are also a few universal type air wall hangers that will fit a wide variety of common track styles—but they don’t fit all. Be sure to use the correct fitting or hanger for the track because some models may seem to fit into the wrong tracks, but they’re not providing the correct support. Air walls have limited weight bearing capacity per point, so it’s critical not to overload any point on the track. Because of their limited single point capacities, air walls are commonly limited to flying lightweight objects, or used as cable supports.

A building’s engineering department should be able to tell you how much weight and at what intervals you can fly from the track. One important thing to consider is whether the room needs to be reconfigured and the walls moved during or immediately after the event. If the walls need the tracks, nothing else can be installed on them.

Unistrut Tracks. It’s very common for convention centers and newer hotel ballrooms to have Unistrut tracks installed in ceilings and sometimes even walls. Like air walls, they require a specific connector to be safely used. Different sized tracks have different safe working loads. Again, a building’s engineering department should be able to tell you the weight limits and spacing required for utilizing the tracks.

Motors. They can be used singly to hoist individual loudspeakers and small arrays, and in multiples to hoist larger arrays. A motor attached to the rear of an array can be used to help tilt and aim the array. Only competent or qualified persons should perform any rigging with motors.

Before use, check the motor hooks and inspect the chain for any signs of damage. Make sure the chain bag is attached correctly and is in good shape. Before any motor is operated, the rigger needs to make sure that everybody on the deck is aware that something will be moving. Before an array or truss is lifted, one competent person should double check all rigging hardware and fittings.

Construction Lifts. They’re sometimes used to elevate or fly loudspeakers at festivals. Scissor lifts are popular when small horizontal arrays need to be raised, moved or relocated at events like airshows. Larger festivals might utilize construction cranes or large extendable boom forklifts to fly the main PA system.

Make sure the equipment operators are certified on that model unit, and that an operator is monitoring the equipment continuously during use. Weight loading changes depending on the angle of the boom so it’s vital to no overload the equipment during use. Tag lines should be used to keep an array from swinging in the wind. An engineer should determine the maximum wind speeds the systems can operate in, and if the wind reaches that limit, the PA should be lowered immediately to the ground.

Remember, safety is the most important thing. Rigging is dangerous, but using the proper equipment and procedures, skilled people can overcome gravity, one loudspeaker at a time.

In addition to being the owner of Las Vegas-based production company Tech Works, Craig Leerman is also a U.S. Navy trained and certified rigger.

Posted by Keith Clark on 05/01 at 03:36 PM
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Tuesday, April 16, 2013

Eliminating Potential Trouble & Getting The Noise Out Of A System

Replacing myth, misinformation, and mystery with knowledge and clear understanding

“A cable is a source of potential trouble connecting two other sources of potential trouble.”

The humor in this statement may be lost on those who regularly assemble sound systems. But a reality of sound systems is that a signal accumulates noise as it flows through equipment and cables. And once noise contaminates a signal, it’s essentially impossible to remove it without altering or degrading the original signal.

For this reason, no system can be quieter than its noisiest link. Noise and interference must be prevented along the entire signal path.

Delivering a signal from one box to another may seem trivial, but when it comes to noise, the signal interface is usually the danger zone, not the equipment’s internal signal processing. Many - if not most - designers and installers of audio systems think of grounding and interfacing as a black art. How many times have you heard someone say that a cable is “picking up” noise - presumably from the air like a radio receiver?

Even most equipment manufacturers often don’t have a clue what’s really going on. The most basic rules of physics are routinely overlooked, ignored, or forgotten. As a result, myth and misinformation have become epidemic!

It’s time to replace myth, misinformation, and mystery with knowledge and clear understanding.

How Quiet Is Quiet?
Of course, how much noise and interference is tolerable depends on how a system is used. A monitor system in a recording studio obviously needs much more immunity to ground noise and interference than a construction site paging system.

The dynamic range of a system is the ratio, generally measured in dB, of its maximum undistorted output signal to its residual output noise or noise floor - up to 120 dB of dynamic range may be required in high-performance sound systems.

By the way, with video systems, a 50 dB signal-to-noise ratio is a generally accepted threshold beyond which no further improvement in images is perceivable, even by expert viewers.

Of course, a predictable amount of “white” noise is inherent in all electronic devices and must be expected. White noise is statistically random and its power is uniformly spread across the signal frequency range. In an audio system, it is heard as “hiss.” Excess random noise is generally due to improper gain structure, a topic that really isn’t part of our discussion here.

On the other hand, ground noise, usually heard as hum, buzz, clicks or pops in audio signals, is generally much more noticeable and irritating. (And note that 10 dB noise reductions are generally described as “half as loud,” while 2 dB to 3 dB reductions are “just noticeable.”)

Myths About Earth Grounding
As electronics developed, the common return paths of various circuits were also referred to as “ground,” regardless of whether or not they were eventually connected to earth. In addition, a single ground circuit most often serves, either intentionally or accidentally, more than one purpose.

Thus, the very meaning of the term ground has become vague, ambiguous, and often quite fanciful. Some engineers have a strong urge to reduce these unwanted voltage differences by “shorting them out” with massive conductors - the results are most often disappointing.

Other engineers think that system noise can be improved experimentally by simply finding a “better” or “quieter” ground. Many indulge in wishful thinking that noise currents can somehow be skillfully directed to an earth ground, where they will disappear forever!

Here are some common myths about grounding:

Earth grounds are all at zero volts - presumably with respect to each other and to some “mystical absolute” reference point. This leads to whimsical ideas about lots of ground rods making system noises disappear! In fact, the soil resistance between ground rods is much higher (often tens of ohms) than a wire between them.

Impedance - symbolized as “Z,” it’s the apparent AC resistance of a circuit containing capacitance and/or inductance in addition to pure resistance. Wires have zero impedance, and, therefore, can extend a zero-voltage reference to many locations in a system, eliminating voltage differences. In fact, wires are quite limited:

The DC Resistance of a wire applies only at very low frequencies and is directly proportional to its length. For example, the resistance of 10 feet of 12-gauge wire is about 0.015 Ohms.

The inductance of a wire is nearly independent of its diameter (gauge) but is directly proportional to its length and increases at bends or loops.

Figure 1 (click to enlarge)

Our 10 feet of 12-gauge wire has an impedance of 30 Ohms at 1 MHz (AM broadcast band) as shown in the Figure 1.

Substituting a 1/2-inch diameter solid copper rod lowers the impedance only slightly to about 25 Ohms.

A wire resonates (becomes an antenna) when its physical length is a quarter wavelength. For a 10-foot wire, this means it will essentially become an open circuit at about 25 MHz.

Are earth grounds really necessary for low-noise system operation? Think about all the electronics in an airplane!

Under fortuitous conditions, systems may be acceptably quiet in spite of poor techniques. But physics will ultimately rule and noises may appear for no apparent reason! 

Once we understand how grounding systems and interfaces actually work and how noises couple into signals, finding and fixing problems becomes simple and logical.Perhaps the most important aspect of troubleshooting is how you think about the problem.

Without a methodical approach, chasing noise problems can be both frustrating and time-consuming. For example, don’t fall into the trap of thinking something can’t be the problem just because you’ve always done it that way. Remember, things that “can’t go wrong” do!

Troubleshooting Advice
Don’t start by changing things! Because many problems reveal themselves if we just gather enough clues, gather as much information as possible before you change anything.

Ask questions! Did it ever work right? What symptoms tell you it’s not working right? When did it start working badly or stop working? What other symptoms showed up just before, just after, or at the same time?

Be alert to clues from the equipment itself! Operation of the equipment’s controls, along with some simple logic, can provide very valuable clues.

For example, if the noise is unaffected by the setting of a volume control or selector, logic dictates that it must be entering the signal path after that control.

If the noise can be eliminated by turning the volume down or selecting another input, it must be entering the signal path before that control.

Write everything down! Less than perfect memory can waste a lot of time.

Sketch a block diagram of the system! Show all signal interconnecting cables, including digital and RF, and indicate their approximate length. Mark any balanced inputs or outputs. Generally, stereo pairs can be indicated with a single line. Also note any equipment that’s grounded via its 3-prong power plug, and note any other ground connections such as cable TV or DSS dishes.

Work through the system backwards! As a general rule, and unless clues suggest another starting point, always begin at the inputs to the power amplifiers and sequentially test interfaces backward toward the signal sources.

Additional installments by Bill Whitlock are available here.

Bill Whitlock has served as president of Jensen Transformers for more than 20 years and is recognized as one of the foremost technical writers in professional audio.

Posted by Keith Clark on 04/16 at 05:01 PM
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Monday, April 15, 2013

Riedel MediorWorks 1.14 Software Unifies MediorNet & RockNet Network Control, Configuration

Tighter integration between the company's real-time networks

Riedel Communications has announced version 1.14 of its MediorWorks configuration, control, and monitoring software, which has been refined to enable even tighter integration between the company’s real-time MediorNet and RockNet networks.

This latest MediorWorks release will give users convenient access to all software configuration and control tools for both systems within a single application and window.

“For users of both our MediorNet video, audio, data, and communications network and our RockNet digital audio network, the new MediorWorks 1.14 release increases ease of use, facilitates fast last-minute configuration changes, and improves monitoring of the overall network installation,” says Henning Kaltheuner, head of product management for Riedel Communications. “This is just one of several new developments that illustrates the benefits offered by closely integrated Riedel network solutions.”

MediorWorks 1.14 not only consolidates control and configuration for Riedel real-time networks into a single interface, but also enables extended SNMP monitoring — all while streamlining network setup and reducing cabling requirements. The software release also supports new capabilities such as audio de-embedding for Grass Valley camera signals, making it simpler for users to manage a greater range of functions and have greater control over signal flow.

The new MediorWorks 1.14 release will be available in May.

Riedel Communications

Posted by Keith Clark on 04/15 at 04:23 PM
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Friday, April 12, 2013

Cable Anatomy Part 2: Everything You Wanted To Know About Loudspeaker Cable

And how it affects various components

See Part 1, Microphone Cable, here.

What are the main parts of a loudspeaker cable, and what does each one do?

Typically a loudspeaker cable has two stranded copper conductors, covered with insulation, twisted together with fillers and sheathed with an overall jacket.

How big should the conductors be?

The required size (or gauge) of the conductors depends on three factors: (1) the load impedance; (2) the length of cable required; and (3) the amount of power loss that can be tolerated. Each of these involves relationships between voltage (volts), resistance (ohms), current (amperes) and power (watts). These relationships are defined with Ohm’s Law.

The job of a loudspeaker cable is to move a substantial amount of electrical current from the output of a power amplifier to a loudspeaker system. Current flow is measure in amperes. Unlike instrument and microphone cables, which typically carry currents of only a few milliamperes (thousandths of an ampere), the current required to drive a speaker is much higher; for instance, an 8-ohm speaker driven with a 100-watt amplifier will pull about 3-1/2 amperes of current.

By comparison, a 600-ohm input driven by a line-level output only pulls about 2 milliamps. The amplifier’s output voltage, divided by the load impedance (in ohms), determines the amount of current “pulled” by the load. Resistance limits current flow, and decreasing it increases current flow. If the amplifier’s output voltage remains constant, it will deliver twice as much current to an 8-ohm load as it will to a 16-ohm load, and four times as much to a 4-ohm load. Halving the load impedance doubles the load current.

For instance, two 8-ohm loudspeakers in parallel will draw twice the current of one loudspeaker because the parallel connection reduces the load impedance to 4 ohms.

(For simplicity’s sake we are using the terms resistance and impedance interchangeably; in practice, a loudspeaker whose nominal impedance is 8 ohms may have a voice coil DC resistance of about 5 ohms and an AC impedance curve that ranges from 5 ohms to 100 ohms, depending on the frequency, type of enclosure, and the acoustical loading of its environment.)


How does current draw affect the conductor requirements of the loudspeaker cable?

A simple fact to remember: Current needs copper, voltage needs insulation. To make an analogy, if electrons were water, voltage would be the “pressure” in the system, while current would be the amount of water flowing. You have water pressure even with the faucet closed and no water flowing; similarly, you have voltage regardless of whether you have current flowing.

Current flow is literally electrons moving between two points at differing electrical potentials, so the more electrons you need to move, the larger the conductors (our “electron pipe”) must be. In the AWG (American Wire Gauge) system, conductor area doubles with each reduction of three in AWG; a 13 AWG conductor has twice the copper of a 16 AWG conductor, a 10 AWG twice the copper of a 13 AWG, and so on.


But power amp outputs are rated in watts. How are amperes related to watts?

Ohm’s Law says that current (amperes) times voltage (volts) equals power (watts), so if the voltage is unchanged, the power is directly proportional to the current, which is determined by the impedance of the load. (This is why most power amplifiers will deliver approximately double their 8-ohm rated output when the load impedance is reduced to 4 ohms.)

In short, a 4-ohm load should require conductors with twice the copper of an 8-ohm load, assuming the length of the run to the loudspeaker is the same, while a 2-ohm load requires four times the copper of an 8-ohm load.

Explaining this point leads to the following oft-asked question:

How long can a loudspeaker cable be before it affects performance?

The ugly truth: Any length of loudspeaker cable degrades performance and efficiency. Like the effects of shunt capacitance in instrument cables and series inductance in microphone cables, the signal degradation caused by loudspeaker cabling is always present to some degree, and is worsened by increasing the length of the cable.

The most obvious ill effect of loudspeaker cables is the amount of amplifier power wasted.


Why do cables waste power?

Copper is a very good conductor of electricity, but it isn’t perfect. It has a certain amount of resistance, determined primarily on its cross-sectional area (but also by its purity and temperature). This wiring resistance is “seen” by the amplifier output as part of the load; if a cable with a resistance of 1 ohm is connected to an 8-ohm loudspeaker, the load seen by the amplifier is 9 ohms. Since increasing the load impedance decreases current flow, decreasing power delivery, we have lost some of the amplifier’s power capability merely by adding the series resistance of the cable to the load.

Furthermore, since the cable is seen as part of the load, part of the power which is delivered to the load is dissipated in the cable itself as heat. (This is the way electrical space heaters work!) Since Ohm’s Law allows us to calculate the current flow created by a given voltage across a given load impedance, it can also give us the voltage drop across the load, or part of the load, for a given current. This can be conveniently expressed as a percentage of the total power.


How can the power loss be minimized?

There are three ways to decrease the power lost in loudspeaker cabling. First, minimize the resistance of the cabling. Use larger conductors, avoid unnecessary connectors, and make sure that mechanical connections are clean and tight and solder joints are smooth and bright.

Second, minimize the length of the cabling. The resistance of the cable is proportional to its length, so less cable means less resistance to expend those watts. Place the power amplifier as close as practical to the loudspeaker. (Chances are excellent that the signal loss in the line-level connection to the amplifier input will be negligible.) Don’t use a 50-foot cable for a 20-foot run.

Third, maximize the load impedance. As the load impedance increases it becomes a larger percentage of the total load, which proportionately reduces the amount lost by wiring resistance. Avoid “daisy-chaining” loudspeakers, because the parallel connection reduces the total load impedance, thus increasing the percentage lost.

The ideal situation (for reasons beyond mere power loss is to run a separate pair of conductors to each loudspeaker form the amplifier.

Is the actual performance of the amplifier degraded by long loudspeaker cables?

There is a definite impact on the amplifier damping factor caused by cabling resistance/impedance. Damping, the ability of the amplifier to control the movement of the speaker, is especially noticeable in percussive low-frequency program material like kick drum, bass guitar and tympani.

Clean, “tight” bass is a sign of good damping at work. Boomy, mushy bass is the result of poor damping; the loudspeaker is being set into motion but the amplifier can’t stop it fast enough to accurately track the waveform. Ultimately, poor damping can result in actual oscillation and loudspeaker destruction.


Damping factor is expressed as the quotient of load impedance divided by the amplifier’s actual source impedance. Ultra-low source impedances on the order of 40 milliohms (that’s less than one-twentieth of an ohm) are common in modern direct-coupled solid-state amplifiers, so damping factors with an 8-ohm load are generally specified in the range of 100-200.

However, those specifications are taken on a test bench, with a non-inductive dummy load attached directly to the output terminals. In the real world, the loudspeaker sees the cabling resistance as part of the source impedance, increasing it. This lowers the damping factor drastically, even when considering only the DC resistance of the cable. If the reactive components that constitute the AC impedance of the cable are considered, the loss of damping is even greater.


Although tube amplifiers generally fall far short of sold-state types in damping performance, their sound can still be improved by the use of larger speaker cables. Damping even comes into play in the performance of mixing consoles with remote DC power supplies; reducing the length of the cable linking the power supply to the console can noticeably improve the low-frequency performance of the electronics.

What other cable problems affect performance?

The twin gremlins covered in Understanding Microphone Cable, namely series inductance and skin effect, are also factors in speaker cables. Series inductance and the resulting inductive reactance adds to the DC resistance, increasing the AC impedance of the cable. An inductor can be thought of as a resistor whose resistance increases as frequency increases.

Thus, series inductance has a low-pass filter characteristic, progressively attenuating high frequencies. The inductance of a round conductor is largely independent of its diameter or gauge, and is not directly proportional to its length, either.

Skin effect is a phenomenon that causes current flow in a round conductor to be concentrated more to the surface of the conductor at higher frequencies, almost as if it were a hollow tube. This increases the apparent resistance of the conductor at high frequencies, and also brings significant phase shift.

Taken together, these ugly realities introduce various dynamic and time-related forms of signal distortion which are very difficult to quantify with simple sine-wave measurements. When complex waveforms have their harmonic structures altered, the sense of immediacy and realism is reduced. The ear/brain combination is incredibly sensitive to the effects of this type of phase distortion, but generally needs direct, A/B comparisons in real time to recognize them.

How can these problems be addressed?

The number of strange designs for loudspeaker cable is amazing. Among them are coaxial, with two insulated spiral “shields” serving as conductors; quad, using two conductors for “positive” and two for “negative;” zip-cord with ultra-fine “rope lay” conductors and transparent jacket; multi-conductor, allegedly using large conductors for lows, medium conductors for mids, and tiny conductors for highs; 4 AWG welding cable; braided flat cable constructed of many individually insulated conductors; and many others.

Most of these address the inductance question by using multiple conductors and the skin effect problem by keeping them relatively small. Many of these “esoteric” cables are extraordinarily expensive; all of them probably offer some improvement in performance over ordinary twisted-pair type cables, especially in critical monitoring applications and high-quality music systems. In most cases, the cost of such cable and its termination, combined with the extremely fragile construction common to them, severely limits their practical use, especially in portable situations.

In short, they cost too much, they’re too hard to work with, and they just aren’t made for rough treatment. But, sonically, they all bear listening to with an open mind; the differences can be surprisingly apparent.

Is capacitance a problem in loudspeaker cables?

The extremely low impedance nature of speaker circuits makes cable capacitance a very minor factor in overall performance. In the early days of solid state amplifiers, highly capacitive loads (such as large electrostatic speaker systems) caused blown output transistors and other problems, but so did heat, short circuits, highly inductive loads and underdesigned power supplies.

Because of this, the dielectric properties of the insulation used are nowhere near as critical as that used for high-impedance instrument cables. The most important consideration for insulation for loudspeaker cables is probably heat resistance, especially because the physical size constraints imposed by popular connectors like the ubiquitous 1/4-in phone plug severely limit the diameter of the cable.

This requires insulation and jacketing to be thin, but tough, while withstanding the heat required to bring a relatively large amount of copper up to soldering temperature. Polyethylene tends to melt too easily, while thermoset materials like rubber and neoprene are expensive and unpredictable with regard to wall thickness PVC is cheap and can be mixed in a variety of ways to enhance its shrink-resistance and flexibility, making it a good choice for most applications. Some varieties of TPR (thermoplastic rubber) are also finding use.

Why don’t loudspeaker cables require shielding?

Actually, there are a few circumstances that may require the shielding of loudspeaker cables. In areas with extreme strong radio frequency interference (RFI) problems, the loudspeaker cables can act as antennae for unwanted signal reception which can enter the system through the output transistors. When circumstances require that loudspeaker-level and microphone-level signals be in close proximity for long distances, such as cue feeds to recording studios, it is a good idea to use shielded loudspeaker cabling (generally foil-shielded, twisted-pair or twisted-triple cable) as “insurance” against possible crosstalk form the cue system entering the microphone lines.

In large installations, pulling the loudspeaker cabling in metallic conduit provides excellent shielding from both RFI and EMI (electromagnetic interference). But, for the most part, the extremely low impedance and high level of loudspeaker signals minimizes the significance of local interference.

Why can’t I use a shielded instrument cable for hooking an amplifier to a loudspeaker, assuming it has the right plugs?

You can, in desperation, use an instrument cable for hooking up an amplifier to a loudspeaker. However, the small gauge (generally 20 AWG at most) center conductor offers substantial resistance to current flow, and in extreme circumstances could heat up until it melts its insulation and short-circuits to the shield, or melts and goes open-circuit, which can destroy some tube amplifiers.

Long runs of coaxial-type cable will have large amounts of capacitance, possibly enough to upset the protection circuitry of some amplifiers, causing untimely shut-downs. And of course there is enormous power loss and damping degradation because of the high impedance of the cable.


• Ballou, Greg, ed., Handbook for Sound Engineers: The New Audio Cyclopedia, Howard W. Sams and Co., Indianapolis, 1987.
• Cable Shield Performance and Selection Guide, Belden Electronic Wire and Cable, 1983.
• Colloms, Martin, “Crystals: Linear and Large,” Hi-Fi News and Record Review, November 1984.
• Cooke, Nelson M. and Herbert F. R. Adams, Basic Mathematics for Electronics, McGraw-Hill, Inc., New York, 1970.
• Davis, Gary and Ralph Jones, Sound Reinforcement Handbook, Hal Leonard Publishing Corp., Milwaukee, 1970.
• Electronic Wire and Cable Catalog E-100, American Insulated Wire Corp., 1984.
• Fause, Ken, “Shielding, Grounding and Safety,” Recording Engineer/Producer, circa 1980.
• Ford, Hugh, “Audio Cables,” Studio Sound, Novemer 1980.
• Guide to Wire and Cable Construction, American Insulated Wire Corp., 1981.
• Grundy, Albert, “Grounding and Shielding Revisited,” dB, October 1980.
• Jung, Walt and Dick Marsh, “Pooge-2: A Mod Symphony for Your Hafler DH200 or Other Power Amplifiers,” The Audio Amateur, 4/1981.
• Maynard, Harry, “Speaker Cables,” Radio-Electronics, December 1978,
• Miller, Paul, “Audio Cable: The Neglected Component,” dB, December 1978.
• Morgen, Bruce, “Shield The Cable!,” Electronic Procucts, August 15, 1983.
• Morrison, Ralph, Grounding and Shielding Techniques in Instrumentation, John Wiley and Sons, New York, 1977.
• Ott, Henry W., Noise Reduciton in Electronic Systems, John Wiley and Sons, New York, 1976.
• Ruck, Bill, “Current Thoughts on Wire,” The Audio Amateur, 4/82.

Thanks to Pro Co Sound for this article.

Posted by Keith Clark on 04/12 at 11:09 AM
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Riedel Announces New Connect AVB Interfaces For Real-Time Ethernet Intercom Applications

Analog or AES3/EBU digital audio over AVB-capable local area networks

Riedel Communications has extended its line of AVB-capable products with the release of the Connect AVB (Audio Video Bridging) interfaces Connect A8 and Connect C8, which enable the reliable real-time transport of analog or AES3/EBU digital audio over AVB-capable local area networks (LANs) with guaranteed quality of service.

“Until now, issues with latency, reliability, and missing synchronization have prevented the connection of intercom panels over LAN environments in broadcast facilities, studios, and OB units,” says Thomas Riedel, CEO, Riedel Communications. “Solutions in our AVB-enabled product line negate these challenges, offering a real-time communication solution that meets the quality demands of professional intercom users.”

Intercom applications for Riedel’s AVB-capable products include matrix-to-control panel connections via LAN, audio distribution via LAN, matrix-to-matrix trunking connections via LAN, and distribution of digital party lines via LAN. The Connect AVB converts signals between AES and AVB standards, supporting the transport of audio signals from equipment such as the Riedel Performer and Artist digital intercom systems on existing network infrastructures.

The Connect AVB modules can be used alone as a throw-down module or with a Riedel Smart Rack system. Built into a compact 9.5-inch, 1-RU housing, the new Connect AVB C8 offers eight AES connections on BNC. The device supports both bidirectional AES for intercom panels and unidirectional transport for broadcast AES. Riedel’s new Connect AVB A8 provides eight analog inputs and outputs on RJ45 connectors.

Riedel is member of the AVnu Alliance and designs its AVB-capable products according to the guidelines of interoperability certification by the AVnu Alliance.

Riedel Communications

Posted by Keith Clark on 04/12 at 09:40 AM
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